SX20 call SIP

Hello

I would like to know, is it possible to call the SX20 of another endpoint with the address ([email protected] / * /) without registering to VCS

Appeal of sense as a standalone device at the other end.

If so, could you guide me.

Best regards

Titebiket Mohamed Yatim

Why do you do this? If its between cisco/tandberg/3 rd party

equipment and its able to use h323, I would have preferred that.

Stand alone sip very likely work properly behind the NAT.

To do this, I highly recommend using a VCS-E

Not to mention to factory default, it would be just to write what you wrote.

Maybe you want to set the call by default sip Protocol and also play

with the DefaultTransport framework for sip, but you can simply dial the ip address.

Please remember useful frequency responses and identify useful or correct answers.

Tags: Cisco Support

Similar Questions

SX20 Autonomous SIP

Hello

I have two SX20 working in h323 and each works as standalone.

But I can't put their work with SIP. Configuration which is correct?

Concerning

Paulo Ferreira

Hi Paulo,

The recommended configuration is to use H323, if you want to use the Sx20 as independent of an endpoint. Given that SIP functionanlity & features require SIP registrar and Proxy Server be available for appeals.

I would recommend that you configure the Sx20 H323 and in Direct mode and you should be able to make the IP to IP calls with no problems.

Hope that answers your question.

Kind regards

Saint André

Call SIP with Tandberg C40

I have a Cisco Tandberg C40 codec and have been making H.323 calls for an organization with no problems. The organization is now making changes and demanding that I connect to them with their new SIP address. When I try to connect, it is said immediately that the call could not be connected. What are the requirements for placing a SIP call with the Tandberg C40?

Assuming your C40 has the version of the TC6.x software;

go to network found under Control Panel Services and enable SIP.

Don't forget of check that SIP is selected as the Protocol to call when calling this company that the default calling Protocol can be set to H.323, SIP and H.320, so if only make you SIP calls, and then upgrade the SIP, but if you do mostly H.323 calls, then put to H.323, and select SIP among the options of appeal when the dial - or even more easily, add this address to the local directory with specified as a SIP call Protocol.

Just turning on SIP will allow you to make outgoing SIP calls, however, if this company wants to call you, then they can call you only using the public IP address of your system - and they could even not able to do according to their deployment, policies, etc.

If you want people to be able to call you using a SIP URI as [email protected] / * /, then you need to register your system with a Registrar SIP CUCM, VCS - C, etc. and also have SRV SIP records in place.

See the guide to the administrator for more information:

http://www.Cisco.com/en/us/docs/Telepresence/endpoint/codec-c-series/TC4/administration_guide/profile-c60_codec-C60-c40_administrator_guide_tc40.PDF

/Jens

Please note the answers and score the questions as "answered" as appropriate.

Dialing SIP calls SIP Trunk (Avaya SM)

Hello. I work with the telco team to send sip of our endpoints VTC voice calls to an Avaya Session Manager.

I think I have all the areas and rules of research on the side of VCS in order to transfer the call to the SM. I am able to confirm with the team of the phone company that the call reached the SM, but they receive the error "cannot determine realm."

On the side of the VCS, I see the output below. Is the problem on the VCS or SM side? I need to create a trust with the SM? THX. - F
- searchRule (2)
- - Name: SIP Audio out rule2
- - Zone (1)
  - - Name: TEST Session Mgr
  - - Type: neighbor
  - - Protocol: SIP
  - - Found: false
  - - Reason: Internal server error (cannot determine the domain)
  - - StartTime: 2015-03-09 13:21:58
  - - Time: 0.05
  - - Access controller (1)
    - - Address: 172.x.x.x:5061
    - - Alias (1)
      - Type: Url
      - Origin: unknown
      - Value: [email protected] / * /
Another option is to see if you get up with 5060/udp or 5060/tcp, which could make it easier to trace. But yes, I prefer the encryption as well.

Often you have set a kind of trust for a specific connection, which could include the field src or dst, ip, port,...

good luck ;-)

the codec sx20 call automatically disconnects automatically after 3 to 5 minut

Dial sx20 to mcu (5300), is automatically disconnect after 3 to 5 minutes.

Please find join snaps for her.

This error message looks like in there is something wrong with your SX20.

You could try it paginate in a previous software image (if there is a loading) by going to the maintenance > System Recovery > software Recovery Swap and clicking on the switch to software: button.

If there is no image software to swap in, you can try to upgrade the software on your endpoint to install a new software image.

Wayne
--
Remember the frequency responses and mark your question as answered as appropriate.

SX20 Quick Set registered but can not make calls

We have several new SX20 units, two of which I installed and registered via SIP with UCM (v8.6.2, required a pack of devices). I can make a call to the SX20 since a regular IP phone (7961 - SCCP), answering the call to unity of telepresence and continue a conversation only one voice. However what composition of the SX20 the other SX20 or any other IP phone it briefly displays "call for binding", then it goes immediately to the "call was disconnected. I tried just the DN 4-digit, as well as the SIP URI of numbering. Here is a part of the log output of command line for the sx20 indicating the call SIP is rejected with a 403 forbidden message.

It is worth noting that before these devices, we have had no recorded via SIP, SCCP phone only. At this point, these devices don't need to dial outside of our network. I had a case open for two weeks and keep TAC is passed between the teams of telepresence and the AAU. I told them we have never had a SIP device registered with our University Complutense of MADRID before and if there is nothing else to be configured, the answer was nothing more for internal calls only. I'm trying to get escalated now, but thought I'd give forums a go. Any help is greatly appreciated, and I apologize if this question is that something is ridiculously simple.

63018.74 DATACTRL I: DataGateCfgReq(ig=7) hdlc = yes

63018.76 i: SipStack (ui = 8, s = 0) SipDialog sendInviteRejToStack (403: Forbidden)

63018.76 msg unknown MC W: Mcfsm::handleOtherMessages() SIP_DialogFreed_Ind of SipStack

63018.77 SipCall I: sip_call_handler::handleSIPMCallRej(8/0/-1): Appeal dismissed (cause: forbidden)

63018.78 I: MC signalLocalReceiveCapsetChange, RemoteParticipant::signalLRCap

63018.78 I: SipCall = appId = 8 makeOutgoingUpdate, stackId =-1, eventCookie = 1, isEmptyCap = 0, holdState = 0

63018.74 DATACTRL I: DataGateCfgReq(ig=7) hdlc = yes

63018.76 i: SipStack (ui = 8, s = 0) SipDialog sendInviteRejToStack (403: Forbidden)

63018.76 msg unknown MC W: Mcfsm::handleOtherMessages() SIP_DialogFreed_Ind of SipStack

63018.77 SipCall I: sip_call_handler::handleSIPMCallRej(8/0/-1): Appeal dismissed (cause: forbidden)

63018.78 I: MC signalLocalReceiveCapsetChange, RemoteParticipant::signalLRCap

63018.78 I: SipCall = appId = 8 makeOutgoingUpdate, stackId =-1, eventCookie = 1, isEmptyCap = 0, holdState = 0

-JW

A look at the logs on your case - looks like there is no number to call in the header AMATURES or Contact, so CUCM rejects the appeal and sends the 403 forbidden. Note the field 'From' below - should be sip:[email protected] / * /. What you configured as your SIP URI in your SX20? Not that familiar with the SX20, but on my EX90 in the laboratory, there is a field of a SIP URI for that I had to enter the DN @ address >. Anything on CUCM that would block the DN to be passed to the SX call to IP phone?

primaryDN = 1070, cannot be called DN to determine the media... sending 403

SIP/2.0 403 forbidden

Via: SIP/2.0/TCP XX. XX. XX. XX:5060; direction = z9hG4bK8664f3a10726dfcdd3f6437593e431b6.1; rport

From: "1070" tag = 123835963de850ed

In: [email protected] / * />; tag = 1701792190

Date: Thu, 14 June 2012 16:56:46 GMT

Call ID:

[email protected] / * /.

CSeq: INVITE 100

Allow-events: presence

WARNING: 399 CM2 "unknown caller DN.

Content-Length: 0 SIP/2.0 403 forbidden
Via: SIP/2.0/TCP XX. XX. XX. XX:5060; direction = z9hG4bK8664f3a10726dfcdd3f6437593e431b6.1; rport
From: "1070" tag = 123835963de850ed
In: [email protected] / * />; tag = 1701792190
Date: Thu, 14 June 2012 16:56:46 GMT
Call ID: [email protected] / * /

CSeq: INVITE 100
Allow-events: presence
WARNING: 399 CM2 "unknown caller DN.
Content-Length: 0

Linksys E4200 and VoipBuster sip call

Hello

Just recently bought a router Linksys E4200 and it was rocking so far! I've updated since a WRT320N. However it seems that the call SIP (using my account Voip Buster) no longer works...

I don't seem to be able to register with the Voip service. Port is 5060 standard (if I remember correctly, I'm not at home now) and I've already enabled on the router the special tick for Voip calling ALG. Router is upgraded to the latest firmware and it's the V1 router.

With all the foregoing, I cannot yet connect to VoipBuster using SIP. It worked like a charm with the WRT320N, so it's something with the router.

Can anyone help? Thanks a bunch!

Try the following steps:

[A] with the help of the Cisco Connect software (if at all, you have installed the router using this software)

1] open the software and go to the option that says "settings of the router.

[2] then click on the option that says 'Advanced settings', which will take you to the router configuration page.

[3] lower the MTU (Maximum Transmission Unit) of 1500 to 1400 or less (usually found on the main page / basic configuration of your router)

4] click administration and disable UPnP.

[B] If you have not installed Cisco connect then you can connect to the router's user interface using its default IP address in the browser which is 192.168.1.1 and enter "admin" as password empty username field. This will take to the web interface of the router and then follow the steps as indicated above to make the changes.

Second, it also depends on the type of VOIP provider, you must open the ports on your router. H.323 uses 1719 and 1720 for signaling, dynamic ports for media. SIP tends to use 5060 for signage, dynamic again for the media.

Follow this link and open the ports on your router to your VOIP: http://www6.nohold.net/Cisco2/ukp.aspx?pid=80&login=1&app=search&vw=1&articleid=17241

SIP calls failing

Hello

I've implemented VCS in both sites, Site A and site B and I did the neighbors between Site A and Site B VCS rules. Search and transformations are configured and implemented and applied.

Calls from Site B to establish a (H323 and SIP) work well, but calls from Site A to Site B work only for H323. Strange things here is localisation tools show the address cannot be resolved and true for this research nevertheless calls SIP of the Site A to Site B do not work when calls is launched of a Cisco endpoint. I already spent a few days and double checked everything but can't break this case, someone knows what's the problem?

Thanks in advance. Mark

Hi Mark,

Since you are able to make SIP calls to internet via MCU and the zones are configured as donotcheckcredential, I think that the problem is with your endpoints rather than VCS. try this:

-check the DNS configuration on your terminals (EX-series, etc.) and make sure that is correct and the connectivity is here

-Make sure that the outgoing setting is set to off

-Make sure the VCS FULL domain name are correct and not poorly spelled and solvable

s ' there is no proxy between VCS and endpoints server, make sure that this field is empty.

-I don't think issues of NTP server for SIP calls, make sure however NTP server is configured and works for endpoints

-one last thing, try saving the end point that is already registered for site A to site B and try to make a call internet SIP, would this work?

Best regards, Ahmad

How cal get Skype click-to-call to install Firefos 10

I have JUST IMPROVED FIREFOX 10. As it happens everytime I update, Firefox says that it cannot install the add-on Skype click-to-call. I used to be able to add tools, but now Skype is not yet listed here; What is called click-to-call SIP's (beta).
Can someone tell me how toget click-to-call from Skype to work once again, or else advise me as to if he charge associated with the SIP functionality and also how ewell it works...
Thank you.

The compatibility module are the responsibility of the developer. All add-on developers are able to update/test their product before any major version of Firefox, due to release a public Alpha/beta of Mozilla's Firefox. This means that all developers of applications update before a major release is made, which often prevents the compatibility between the module and the browser.

Skype has announced that they have updated their "Click-To-Call' add-on for Firefox10
- Article: update of Skype Click-to-Call
Make sure that you have downloaded the version more returned to the Add on Click-to-Call on the Skype website.

SPA8000 GPON sip GPON del oneway speech

I have SPA8000 with 6 SIP PBX lines. Call SIP GPON to speech always way Del GPON. So that SIP SIP, SIP to PSTN, everything ok. This problem is on version 6.1.10 while 5.10.1 is ok. PL inform what establishes the difference between these two versions. All the 6 SIP lines have oneway speech.

This could be block ports by your router firewall. I could recommend to transfer ports UDP 5060-5061 and UDP 16384-16391. And also, you can activate the NAT mapping and NAT keep by DOSB with an account recorded live on

Cisco Ip phone 3905 calls do not

We have 15 Cisco IP phone 3905 on a network with CUCME ver 9.1. Phones record fine and received number EXT. It has tone. But we can't write another post after dialing '1' the tone cut and nothing happens.

This problem has been posted here before. https://supportforums.Cisco.com/discussion/11651601/3905-issue-making-call

Except in my case, I only use 3905 and calls cannot take place. See my sh run and debug errors and debugging ccsip attached massages ccsip. Thanks for your help.

SH run
Building configuration...

Current configuration: 7315 bytes
!
! Last configuration change at 13:56:27 UTC Thursday, April 24, 2014 by nim
version 15.2
horodateurs service debug datetime msec
Log service timestamps datetime msec
no password encryption service
!
hostname NIM_CME
!
boot-start-marker
boot-end-marker
!
!
logging buffered 51200 warnings
!
No aaa new-model
!
IP cef
!
!
!
no record of conflict ip dhcp
DHCP excluded-address 192.168.0.1 IP 192.168.0.10
DHCP excluded-address IP 192.168.80.1 192.168.80.10
DHCP excluded-address IP 192.168.80.250 192.168.80.255
DHCP excluded-address IP 192.168.70.1 192.168.70.10
DHCP excluded-address IP 192.168.70.250 192.168.70.255
DHCP excluded-address IP 192.168.60.1 192.168.60.10
DHCP excluded-address IP 192.168.60.250 192.168.60.255
DHCP excluded-address IP 192.168.50.1 192.168.50.10
DHCP excluded-address IP 192.168.50.250 192.168.50.255
DHCP excluded-address IP 192.168.40.1 192.168.40.10
DHCP excluded-address IP 192.168.40.250 192.168.40.255
DHCP excluded-address IP 192.168.30.1 192.168.30.10
DHCP excluded-address IP 192.168.30.250 192.168.30.255
DHCP excluded-address 192.168.20.1 IP 192.168.20.10
DHCP excluded-address IP 192.168.20.250 192.168.20.255
IP dhcp excluded-address 192.168.0.250 192.168.0.255
!
dhcp PHCBase IP pool
import all
network 192.168.0.0 255.255.255.0
default router 192.168.0.1
option 150 ip 192.168.0.1
Rental 30
!
dhcp YenegoaLAN IP pool
network 192.168.80.0 255.255.255.0
router by default - 192.168.80.1
lease 10
!
dhcp OronLAN IP pool
network 192.168.70.0 255.255.255.0
router by default - 192.168.70.1
lease 10
!
dhcp EketLAN IP pool
network 192.168.60.0 255.255.255.0
router by default - 192.168.60.1
lease 10
!
dhcp CalabarLAN IP pool
network 192.168.50.0 255.255.255.0
router by default - 192.168.50.1
lease 10
!
dhcp BonnyLAN IP pool
network 192.168.40.0 255.255.255.0
router by default - 192.168.40.1
option 150 ip 192.168.40.1
lease 10
!
dhcp OnneLAN IP pool
network 192.168.30.0 255.255.255.0
default router 192.168.30.1
lease 10
!
dhcp PortOfficeLAN IP pool
network 192.168.20.0 255.255.255.0
router by default - 192.168.20.1
lease 10
!
!
!
no ip domain search
"yourdomain.com" of the IP domain name
No ipv6 cef
Authenticated MultiLink bundle-name Panel
!
!
!
!
!
!
Crypto pki trustpoint TP-self-signed-2286552849
enrollment selfsigned
name of the object cn = IOS - Self - signed - certificate - 2286552849
revocation checking no
rsakeypair TP-self-signed-2286552849
!
!
TP-self-signed-2286552849 crypto pki certificate chain
certificate self-signed 01
3082022B 30820194 02020101 300 D 0609 2A 864886 F70D0101 05050030 A0030201
2 060355 04031326 494F532D 53656 C 66 2 AND 536967 6E65642D 43657274 31312F30
69666963 32323836 35353238 6174652D 3439301E 170 3133 30383230 30353236
33395A 17 0D 323030 31303130 30303030 305A 3031 06035504 03132649 312F302D
4F532D53 5369676E 656C662D 43 65727469 66696361 74652 32 32383635 65642D
35323834 3930819F 300 D 0609 2A 864886 01050003, 818, 0030, 81890281 F70D0101
8100B 979 6576 B1DBA804 61398EE7 DFB6E285 AEBE044F 300D381C 1FBC941C 407D
D062F622 47E0A79E 20641E4C CC90F308 8D65DC2C CC475EC3 0A62175E 867366ED
C5B35A90 83090DDF ADDAF4A4 CA49F2C4 7C3421F1 0B4EC5AE D26A0CE9 7DC3CC55
E604A7A2 0AF66F47 66FAF1BA 2A823FD3 EC9AAC89 5FCEDD29 6B2DDCF9 E1C41D9F
010001A 3 53305130 1 130101 FF040530 030101FF 301F0603 0F060355 C9B50203
B 551 2304 18301680 1486, 158 90DD3652 93809798 C3311ABE 9EC6263E 09301D 06
03551D0E DD365293 809798 C6263E09 311ABE9E 3 C 300 D 0609 B 04160414 86, 15890
2A 864886 05050003 81810000 2B614A99 9B090B99 3A7F9085 C29503B3 F70D0101
E92AB95A ABD6EED5 E9226AAD 63E60837 FF913665 96D2ECAB 6F6DA306 42751B 49
8CC3EF9B E13C3B49 B2B978AD ABC1A42E EFA8D5EF FC4C9C6A A1662E2D 0C140E5D
5F0B6752 CAEC8E8A 53EB3353 E27A8575 C18381D7 9342773B CB3BCD65 54C0DF25
D629972D 409A2F6D 2C82C541 611A1F
quit smoking
voice-card 0
!
!
!
voip phone service
allow sip to sip connections
Fax protocol t38 ls-redundancy version 0 0 hs-redundancy 0 help none
SIP
binding control source-interface GigabitEthernet0/0
bind media source-interface GigabitEthernet0/0
Registration Server expires max 1200 min 300
!
!
Global voice registry
FMC of fashion
source-address 192.168.0.1 port 5060
Max - dn 30
Max-pool 25
load of 3905 CP3905.9 - 2-1-0
1 4085251 model numbering plan... extension-length 3
Flash TFTP-path:
text file
create the profile synchronization 0035041805641981
!
Register of voice dn 1
Number 101
name numbers1
label 4085251001
!
Register of voice dn 2
number 102
name telephone2
label 4085251002
!
Register of voice dn 3
number 103
name Phone3
label 4085251003
!
Register of voice dn 4
number 104
name Phone4
label 4085251004
!
vocal range pool 1
Mac ID 7 95. F323. B7B6
type of 3905
Number 1 dn 1
DTMF-relay rtp - nte
!
Register of voice pool 2
Mac ID 7 95. F323. B81D
type of 3905
Number 1 dn 2
DTMF-relay rtp - nte
Cisco password username user2
!
Register of voice pool 3
Mac ID 7 95. F323. BB30
type of 3905
Number 1 dn 3
DTMF-relay rtp - nte
username cisco password user3
!
Register of voice pool 4
Mac ID 7 95. F323. B7B7
type of 3905
Number 1 dn 4
DTMF-relay rtp - nte
Cisco password username user4
!
!
!
!
!
license udi pid CISCO2911/K9 sn FCZ1734609V
HW-module pvdm 0/0
!
!
!
username privilege 15 secret 4 nimout lpMHsjg3v8XIXfjVSuCP0Tf3rTGlWmA/nJHqUqryL7w
username admin privilege 15 secret 4 twCnybukZZA6Z960oKoBqFYi5O74Z5b73d7LIBiSjrY
!
redundancy
!
!
!
!
!
!
the Embedded-Service-Engine0/0 interface
no ip address
Shutdown
!
interface GigabitEthernet0/0
Description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE $ 0/0
the IP 192.168.0.1 255.255.255.0
automatic duplex
automatic speed
!
interface GigabitEthernet0/0.20
encapsulation dot1Q 20
address 192.168.20.1 255.255.255.0
!
interface GigabitEthernet0/0.30
encapsulation dot1Q 30
192.168.30.1 IP address 255.255.255.0
!
interface GigabitEthernet0/0.40
encapsulation dot1Q 40
192.168.40.1 IP address 255.255.255.0
!
interface GigabitEthernet0/0.50
encapsulation dot1Q 50
192.168.50.1 IP address 255.255.255.0
!
interface GigabitEthernet0/0.60
encapsulation dot1Q 60
IP 192.168.60.1 255.255.255.0
!
interface GigabitEthernet0/0.70
encapsulation dot1Q 70
IP 192.168.70.1 255.255.255.0
!
interface GigabitEthernet0/0.80
encapsulation dot1Q 80
192.168.80.1 IP address 255.255.255.0
!
interface GigabitEthernet0/1
no ip address
Shutdown
automatic duplex
automatic speed
!
interface GigabitEthernet0/2
no ip address
Shutdown
automatic duplex
automatic speed
!
IP forward-Protocol ND
!
IP http server
local IP http authentication
IP http secure server
IP http timeout policy slowed down 60 life 86400 request 10000
!
!
!
!
flash TFTP server: APP3905.9 - 2-1 - 0.zz
flash TFTP server: CP3905.9 - 2-1 - 0.loads
!
control plan
!
!
!
!
!
!
!
profile MGCP default
!
!
!
!
!
access controller
Shutdown
!
!
phone service
No auto-reg ephone
MAX conferences 8-6 win
DN-webedit
transfer full-consult system
!
!
!
Line con 0
password @nimout123
local connection
line to 0
line 2
no activation-character
No exec
preferred no transport
transport output pad rlogin lapb - your MOP v120 udptn ssh telnet
StopBits 1
line vty 0 4
privilege level 15
password @nimout123
local connection
transport input telnet ssh
line vty 5
privilege level 15
password @nimout123
local connection
transport input telnet ssh
line vty 6 15
privilege level 15
local connection
transport input telnet ssh
!
Scheduler allocate 20000 1000
Master of NTP
!
end

#debug ccsip error
Trouble shooting call SIP is enabled
NIM_CME #.
NIM_CME #.
* 24 apr 11:08:36.251: //-1/9D9F100280CC/SIP/Error/ccsip_ipip_media_forking_updat
e_preferred_codec:
MF: Not a forked leg SIP...
SIP: (68) attribute mid, instance of level 1 1 not found.
* Apr 24 11:08:36.251: / / 68/9D9F100280CC/SIP/error/sipSPIStreamTypeAndDtmfRelay:

No voice codec and dtmf-relay correspondence
* Apr 24 11:08:36.251: / / 68/9D9F100280CC/SIP/error/sipSPIDoAudioNegotiation:
Failure of the negotiations in media for m-line 1
* Apr 24 11:08:36.251: / / 68/9D9F100280CC/SIP/error/sipSPIDoMediaNegotiation:

No valid fax or audio stream
* Apr 24 11:08:36.251: / / 68/9D9F100280CC/SIP/error/sipSPIHandleInviteMedia:
Failure of the negotiation of media for an incoming call
* Apr 24 11:08:36.251: / / 68/9D9F100280CC/SIP/error/sipSPIContinueNewMsgInvite:
Unacceptable media for INVITE
* 24 apr 11:08:39.303: //-1/9F70C2BB80D0/SIP/Error/ccsip_ipip_media_forking_updat
e_preferred_codec:
MF: Not a forked leg SIP...
SIP: (69) attribute mid, instance of level 1 1 not found.
* Apr 24 11:08:39.303: / / 69/9F70C2BB80D0/SIP/error/sipSPIStreamTypeAndDtmfRelay:

No voice codec and dtmf-relay correspondence
* Apr 24 11:08:39.303: / / 69/9F70C2BB80D0/SIP/error/sipSPIDoAudioNegotiation:
Failure of the negotiations in media for m-line 1
* Apr 24 11:08:39.303: / / 69/9F70C2BB80D0/SIP/error/sipSPIDoMediaNegotiation:

No valid fax or audio stream
* Apr 24 11:08:39.303: / / 69/9F70C2BB80D0/SIP/error/sipSPIHandleInviteMedia:
Failure of the negotiation of media for an incoming call
* Apr 24 11:08:39.303: / / 69/9F70C2BB80D0/SIP/error/sipSPIContinueNewMsgInvite:
Unacceptable media for INVITE
* 24 apr 11:08:42.871: //70/A191CD1180D4/SIP/Error/ccsip_spi_register_incoming_re
gistration:

No entry found in reg number Table for 104
* 24 apr 11:08:47.803: //-1/xxxxxxxxxxxx/SIP/Error/ccsip_reg_delete_from_cc_call_
id_table:
Entry not found for the search key

* 24 apr 11:08:47.803: //-1/xxxxxxxxxxxx/SIP/Error/ccsip_reg_delete_from_mac_tabl
e:
BCR with mac [7c95f323b7b7] has been deleted
* Apr 24 11:08:47.803: POOL - 4 VOICE REGISTER has not been saved. Name: SEP7C95F323B7
B7 IP:192.168.40.11 DeviceType:Phone

* 24 apr 11:08:47.803: //-1/xxxxxxxxxxxx/SIP/Error/sipSPI_ipip_PassthruContentCon
tainerFreeHelper:
ContentQ null - output
* 24 apr 11:08:47.803: //-1/xxxxxxxxxxxx/SIP/Error/ccsip_register_handle_e164_unr
registration:
SIP registry Error: Invalid args in unreg
* 24 apr 11:08:47.803: //-1/xxxxxxxxxxxx/SIP/Error/ccsip_register_handle_e164_unr
registration:
SIP registry Error: Invalid args in unreg
* 24 apr 11:09:20.891: //-1/B83B30D980D5/SIP/Error/ccsip_ipip_media_forking_updat
e_preferred_codec:
MF: Not a forked leg SIP...
SIP: (71) attribute mid, instance of level 1 1 not found.
* Apr 24 11:09:20.895: / / 71/B83B30D980D5/SIP/error/sipSPIStreamTypeAndDtmfRelay:

No voice codec and dtmf-relay correspondence
* Apr 24 11:09:20.895: / / 71/B83B30D980D5/SIP/error/sipSPIDoAudioNegotiation:
Failure of the negotiations in media for m-line 1
* Apr 24 11:09:20.895: / / 71/B83B30D980D5/SIP/error/sipSPIDoMediaNegotiation:

No valid fax or audio stream
* Apr 24 11:09:20.895: / / 71/B83B30D980D5/SIP/error/sipSPIHandleInviteMedia:
Failure of the negotiation of media for an incoming call
* Apr 24 11:09:20.895: / / 71/B83B30D980D5/SIP/error/sipSPIContinueNewMsgInvite:
Unacceptable media for INVITE
* 24 apr 11:09:24.535: //-1/BA67393580D9/SIP/Error/ccsip_ipip_media_forking_updat
e_preferred_codec:
MF: Not a forked leg SIP...
SIP: (72) attribute mid, instance of level 1 1 not found.
* Apr 24 11:09:24.535: / / 72/BA67393580D9/SIP/error/sipSPIStreamTypeAndDtmfRelay:

No voice codec and dtmf-relay correspondence
* Apr 24 11:09:24.535: / / 72/BA67393580D9/SIP/error/sipSPIDoAudioNegotiation:
Failure of the negotiations in media for m-line 1
* Apr 24 11:09:24.535: / / 72/BA67393580D9/SIP/error/sipSPIDoMediaNegotiation:

No valid fax or audio stream
* Apr 24 11:09:24.535: / / 72/BA67393580D9/SIP/error/sipSPIHandleInviteMedia:
Failure of the negotiation of media for an incoming call
* Apr 24 11:09:24.535: / / 72/BA67393580D9/SIP/error/sipSPIContinueNewMsgInvite:
Unacceptable media for INVITE
* 24 apr 11:09:53.307: //-1/CB8CDFE280DD/SIP/Error/ccsip_ipip_media_forking_updat
e_preferred_codec:
MF: Not a forked leg SIP...
SIP: (73) attribute mid, instance of level 1 1 not found.
* Apr 24 11:09:53.307: / / 73/CB8CDFE280DD/SIP/error/sipSPIStreamTypeAndDtmfRelay:

No voice codec and dtmf-relay correspondence
* Apr 24 11:09:53.307: / / 73/CB8CDFE280DD/SIP/error/sipSPIDoAudioNegotiation:
Failure of the negotiations in media for m-line 1
* Apr 24 11:09:53.307: / / 73/CB8CDFE280DD/SIP/error/sipSPIDoMediaNegotiation:

No valid fax or audio stream
* Apr 24 11:09:53.307: / / 73/CB8CDFE280DD/SIP/error/sipSPIHandleInviteMedia:
Failure of the negotiation of media for an incoming call
* Apr 24 11:09:53.307: / / 73/CB8CDFE280DD/SIP/error/sipSPIContinueNewMsgInvite:
Unacceptable media for INVITE

ebug ccsip?
All activate SIP all traces of debugging
the calls allow CCSIP SPI called backtrace
The trace debugging DHCP enable SIP-DHCP
error activate SIP debug trace
Activate SIP events backtrace
function activate SIP debug trace
Info to activate SIP info trace debugging
Activate SIP media backtrace
messages enable CCSIP SPI debug trace
preauthentication activate SIP preauthentication debugging traces
Activate CCSIP SPI States debug trace
definition of translation activate SIP debug trace
transport transport activate SIP, traces of debugging
Verbose Enable verbose mode

Event ccsip NIMASA_CME #debug
Events to call SIP tracing is enabled
NIMASA_CME #.
April 24, 12:23:59.435: / / 167/25A61E588123/SIP/event/Session-Timer/sipSTSLMain: Eve
NT: E_STSL_SESSION_REFRESH_REQ
April 24, 12:23:59.435: / / 167/25A61E588123/SIP/event/Session-Timer/sipSTSLMain: dir
: method 2, p. 102, resp_code:0, container: 3DAA7E00
Apr 24 12:23:59.435: //167/25A61E588123/SIP/Event/Session-Timer/sipSTSLPrintTDCo
ntainer: Peer-Event: E_STSL_LEG_BY_LEG, SE value: 0, SE reminder: no, Min - SE Va
read: 1800, flags: 2000
April 24, 12:23:59.435: / / 167/25A61E588123/SIP/event/Session-Timer/sipSTSLMain: Eve
NT: E_STSL_SESSION_REFRESH_RESP
April 24, 12:23:59.435: / / 167/25A61E588123/SIP/event/Session-Timer/sipSTSLMain: dir
: method 1, p. 102, resp_code:488, container: 3DAA8220
April 24, 12:24:04.583: / / 168/28B8405A8127/SIP/event/Session-Timer/sipSTSLMain: Eve
NT: E_STSL_SESSION_REFRESH_REQ
April 24, 12:24:04.583: / / 168/28B8405A8127/SIP/event/Session-Timer/sipSTSLMain: dir
: method 2, p. 102, resp_code:0, container: 3DAA8E80
Apr 24 12:24:04.583: //168/28B8405A8127/SIP/Event/Session-Timer/sipSTSLPrintTDCo
ntainer: Peer-Event: E_STSL_LEG_BY_LEG, SE value: 0, SE reminder: no, Min - SE Va
read: 1800, flags: 2000
April 24, 12:24:04.583: / / 168/28B8405A8127/SIP/event/Session-Timer/sipSTSLMain: Eve
NT: E_STSL_SESSION_REFRESH_RESP
April 24, 12:24:04.583: / / 168/28B8405A8127/SIP/event/Session-Timer/sipSTSLMain: dir
: method 1, p. 102, resp_code:488, container: 3DAB0BF8
April 24, 12:24:07.155: / / 169/2A401975812B/SIP/event/Session-Timer/sipSTSLMain: Eve
NT: E_STSL_SESSION_REFRESH_REQ
April 24, 12:24:07.155: / / 169/2A401975812B/SIP/event/Session-Timer/sipSTSLMain: dir
: method 2, p. 102, resp_code:0, container: 3DAA7F08
Apr 24 12:24:07.155: //169/2A401975812B/SIP/Event/Session-Timer/sipSTSLPrintTDCo
ntainer: Peer-Event: E_STSL_LEG_BY_LEG, SE value: 0, SE reminder: no, Min - SE Va
read: 1800, flags: 2000
April 24, 12:24:07.155: / / 169/2A401975812B/SIP/event/Session-Timer/sipSTSLMain: Eve
NT: E_STSL_SESSION_REFRESH_RESP
April 24, 12:24:07.155: / / 169/2A401975812B/SIP/event/Session-Timer/sipSTSLMain: dir
: method 1, p. 102, resp_code:488, container: 3DAA8220
April 24, 12:24:11.595: / / 170/2CE63252812F/SIP/event/Session-Timer/sipSTSLMain: Eve
NT: E_STSL_SESSION_REFRESH_REQ
April 24, 12:24:11.595: / / 170/2CE63252812F/SIP/event/Session-Timer/sipSTSLMain: dir
: method 2, p. 102, resp_code:0, container: 3DAB0CA8
Apr 24 12:24:11.595: //170/2CE63252812F/SIP/Event/Session-Timer/sipSTSLPrintTDCo
ntainer: Peer-Event: E_STSL_LEG_BY_LEG, SE value: 0, SE reminder: no, Min - SE Va
read: 1800, flags: 2000
April 24, 12:24:11.595: / / 170/2CE63252812F/SIP/event/Session-Timer/sipSTSLMain: Eve
NT: E_STSL_SESSION_REFRESH_RESP
April 24, 12:24:11.595: / / 170/2CE63252812F/SIP/event/Session-Timer/sipSTSLMain: dir
: method 1, p. 102, resp_code:488, container: 3DAB0BF8
NIMASA_CME #undebug ccsip events
Events to call SIP tracing is disabled
NIMASA_CME #.
Error ccsip NIMASA_CME #debug
Trouble shooting call SIP is enabled
NIMASA_CME #.
Apr 24 12:24:46.175: //-1/4182B07D8133/SIP/Error/ccsip_ipip_media_forking_update
_preferred_codec:
MF: Not a forked leg SIP...
SIP: (171) of the mid, found 1 1 level instance attribute.
April 24, 12:24:46.175: / / 171/4182B07D8133/SIP/error/sipSPIDoAudioNegotiation:
Failure of the negotiations in media for m-line 1
April 24, 12:24:46.175: / / 171/4182B07D8133/SIP/error/sipSPIDoMediaNegotiation:

No valid fax or audio stream
April 24, 12:24:46.175: / / 171/4182B07D8133/SIP/error/sipSPIHandleInviteMedia:
Failure of the negotiation of media for an incoming call
April 24, 12:24:46.175: / / 171/4182B07D8133/SIP/error/sipSPIContinueNewMsgInvite:
Unacceptable media for INVITE
Apr 24 12:24:50.227: //-1/43EC5D8E8137/SIP/Error/ccsip_ipip_media_forking_update
_preferred_codec:
MF: Not a forked leg SIP...
SIP: (172) the mid not found 1 1 level instance attribute.
April 24, 12:24:50.227: / / 172/43EC5D8E8137/SIP/error/sipSPIDoAudioNegotiation:
Failure of the negotiations in media for m-line 1
April 24, 12:24:50.227: / / 172/43EC5D8E8137/SIP/error/sipSPIDoMediaNegotiation:

No valid fax or audio stream
April 24, 12:24:50.227: / / 172/43EC5D8E8137/SIP/error/sipSPIHandleInviteMedia:
Failure of the negotiation of media for an incoming call
April 24, 12:24:50.227: / / 172/43EC5D8E8137/SIP/error/sipSPIContinueNewMsgInvite:
Unacceptable media for INVITE
Apr 24 12:24:52.031: //-1/45003EE4813B/SIP/Error/ccsip_ipip_media_forking_update
_preferred_codec:
MF: Not a forked leg SIP...
SIP: (173) of the mid, found 1 1 level instance attribute.
April 24, 12:24:52.031: / / 173/45003EE4813B/SIP/error/sipSPIDoAudioNegotiation:
Failure of the negotiations in media for m-line 1
April 24, 12:24:52.031: / / 173/45003EE4813B/SIP/error/sipSPIDoMediaNegotiation:

No valid fax or audio stream
April 24, 12:24:52.031: / / 173/45003EE4813B/SIP/error/sipSPIHandleInviteMedia:
Failure of the negotiation of media for an incoming call
April 24, 12:24:52.031: / / 173/45003EE4813B/SIP/error/sipSPIContinueNewMsgInvite:
Unacceptable media for INVITE

CCSIP DEBUG MESSAGES

Apr 24 14:18:11.086: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Envoy:
SIP/2.0 488 Media is not Acceptable
Via: SIP/2.0/UDP 192.168.0.13:5060; rport; branch = z9hG4bKPjmYBG2kj6Ljizpy4D3JhcmoM
f0RhNAekv
From: "telephone1" <> [email protected]/ * / >; tag = 3b114524-60cb-4f40-97ee-21dd2016e031
To: sip:[email protected]/ * /; tag = 138F9F8-451
Date: Thu, April 24, 2014 14:18:11 GMT
Call ID: 4dd4fcb7-662c-4e25-8836-feaeeb979f81
CSeq: INVITE 16958
Allow-events: telephone-event
WARNING: 304 192.168.0.1 "Media type (s) not available.
Reason: Q.850; cause = 65
Server: Cisco-SIPGateway/IOS-15.2.4.M4
Content-Length: 0

Apr 24 14:18:11.102: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP ACK:[email protected]/ * /: SIP-5060/2.0
Via: SIP/2.0/UDP 192.168.0.13:5060; rport; branch = z9hG4bKPjmYBG2kj6Ljizpy4D3JhcmoM
f0RhNAekv
Max-Forwards: 70
From: "telephone1" <> [email protected]/ * / >; tag = 3b114524-60cb-4f40-97ee-21dd2016e031
To: sip:[email protected]/ * /; tag = 138F9F8-451
Call ID: 4dd4fcb7-662c-4e25-8836-feaeeb979f81
CSeq: ACK 16958
Content-Length: 0

Apr 24 14:20:40.786: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
GUEST sip:[email protected]/ * /: SIP-5060/2.0
Via: SIP/2.0/UDP 192.168.0.12:5060; rport; branch = z9hG4bKPjvoVJeyhxpOvfwgDDywmTGiU
keV.m - NL1
Max-Forwards: 70
From: 'Téléphone2' <> [email protected]/ * / >; tag = 700813bc-4b2d-47a5-8ea5-1e8aec850a3c
To: sip:[email protected]/ * /.
"Contact: <> [email protected]/ * /: 5060 >; + sip.instance = '.<>
00-0000-0000-7c95f323b81d > '; + u.sip! DeviceName.CCM.Cisco.com = "SEP7C95F323B81D"; + u
. SIP! model.ccm.cisco.com = "592"
Call ID: dce6e9c3-0dab-4c9f-86a6-56fe383c60e0
CSeq: 1575 INVITE
Allow: PRACK, GUEST, ACK, BYE, CANCEL, update, SUBSCRIBE, NOTIFY, REFER, MESSAG
E, OPTIONS
User-Agent: Cisco-CP3905/9.2.1
Support: replaces,join,sdp-anat,norefersub,extended-refer,X-cisco-callinfo,X-c
ISCO-ServiceUri,X-Cisco-escapecodes,X-Cisco-Service-Control,X-Cisco-monrec,X-CIS
Co-config,X-Cisco-SIS-4.0.0,X-Cisco-xsi-7.0.1
Expires: 180
Accept: application/sdp
Allow-events: kpml, dialog box
Remote-Party-ID: 'Téléphone2'<>[email protected]/ * /: 5060 >; intimacy = off
Content-Type: application/sdp
Content-Length: 292

v = 0
o =-2208994977 2208994977 IN IP4 192.168.0.12
s = FOXPHONE
c = in IP4 in 192.168.0.12
t = 0 0
a = X - nat:0
m = audio RTP/AVP 0 8 18 111 16392
a = rtpmap:0 PCMU/8000
a = rtpmap:8 PCMA/8000
a G729/8000 rtpmap:18 =
a = annex b fmtp:18 = yes
a = sendrecv
a rtpmap:111 telephone-event/8000 =
a = fmtp:111 0-15

Apr 24 14:20:40.790: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Envoy:
SIP/2.0 488 Media is not Acceptable
Via: SIP/2.0/UDP 192.168.0.12:5060; rport; branch = z9hG4bKPjvoVJeyhxpOvfwgDDywmTGiU
keV.m - NL1
From: 'Téléphone2' <> [email protected]/ * / >; tag = 700813bc-4b2d-47a5-8ea5-1e8aec850a3c
To: sip:[email protected]/ * /; tag = 13B42C0-812
Date: Thu, April 24, 2014 14:20:40 GMT
Call ID: dce6e9c3-0dab-4c9f-86a6-56fe383c60e0
CSeq: 1575 INVITE
Allow-events: telephone-event
WARNING: 304 192.168.0.1 "Media type (s) not available.
Reason: Q.850; cause = 65
Server: Cisco-SIPGateway/IOS-15.2.4.M4
Content-Length: 0

Apr 24 14:20:40.806: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP ACK:[email protected]/ * /: SIP-5060/2.0
Via: SIP/2.0/UDP 192.168.0.12:5060; rport; branch = z9hG4bKPjvoVJeyhxpOvfwgDDywmTGiU
keV.m - NL1
Max-Forwards: 70
From: 'Téléphone2' <> [email protected]/ * / >; tag = 700813bc-4b2d-47a5-8ea5-1e8aec850a3c
To: sip:[email protected]/ * /; tag = 13B42C0-812
Call ID: dce6e9c3-0dab-4c9f-86a6-56fe383c60e0
CSeq: 1575 ACK
Content-Length: 0

Apr 24 14:22:16.110: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
GUEST sip:[email protected]/ * /: SIP-5060/2.0
Via: SIP/2.0/UDP 192.168.0.14:5060; rport; branch = z9hG4bKPjT3UaPZ-qfLE2EACrxGcnjqG
noFQuZXi7
Max-Forwards: 70
From: 'Phone3' <> [email protected]/ * / >; tag = 8ce0e16a-Davis-448 b-8d9c-d0b4ad52589f
To: sip:[email protected]/ * /.
"Contact: <> [email protected]/ * /: 5060 >; + sip.instance = '.<>
00-0000-0000-7c95f323bb30 > '; + u.sip! DeviceName.CCM.Cisco.com = "SEP7C95F323BB30"; + u
. SIP! model.ccm.cisco.com = "592"
Call ID: d5ec8dba-bbe3-44ac-9902-0d52b327c514
CSeq: INVITE 16131
Allow: PRACK, GUEST, ACK, BYE, CANCEL, update, SUBSCRIBE, NOTIFY, REFER, MESSAG
E, OPTIONS
User-Agent: Cisco-CP3905/9.2.1
Support: replaces,join,sdp-anat,norefersub,extended-refer,X-cisco-callinfo,X-c
ISCO-ServiceUri,X-Cisco-escapecodes,X-Cisco-Service-Control,X-Cisco-monrec,X-CIS
Co-config,X-Cisco-SIS-4.0.0,X-Cisco-xsi-7.0.1
Expires: 180
Accept: application/sdp
Allow-events: kpml, dialog box
Remote-Party-ID: 'Phone3'<>[email protected]/ * /: 5060 >; intimacy = off
Content-Type: application/sdp
Content-Length: 292

v = 0
o =-2208995063 2208995063 IN IP4 192.168.0.14
s = FOXPHONE
c = IN IP4 192.168.0.14
t = 0 0
a = X - nat:0
m = audio RTP/AVP 0 8 18 111 16388
a = rtpmap:0 PCMU/8000
a = rtpmap:8 PCMA/8000
a G729/8000 rtpmap:18 =
a = annex b fmtp:18 = yes
a = sendrecv
a rtpmap:111 telephone-event/8000 =
a = fmtp:111 0-15

Apr 24 14:22:16.114: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Envoy:
SIP/2.0 488 Media is not Acceptable
Via: SIP/2.0/UDP 192.168.0.14:5060; rport; branch = z9hG4bKPjT3UaPZ-qfLE2EACrxGcnjqG
noFQuZXi7
From: 'Phone3' <> [email protected]/ * / >; tag = 8ce0e16a-Davis-448 b-8d9c-d0b4ad52589f
To: sip:[email protected]/ * /; tag = 13CB71C - 8 9
Date: Thu, April 24, 2014 14:22:16 GMT
Call ID: d5ec8dba-bbe3-44ac-9902-0d52b327c514
CSeq: INVITE 16131
Allow-events: telephone-event
WARNING: 304 192.168.0.1 "Media type (s) not available.
Reason: Q.850; cause = 65
Server: Cisco-SIPGateway/IOS-15.2.4.M4
Content-Length: 0

Apr 24 14:22:16.126: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP ACK:[email protected]/ * /: SIP-5060/2.0
Via: SIP/2.0/UDP 192.168.0.14:5060; rport; branch = z9hG4bKPjT3UaPZ-qfLE2EACrxGcnjqG
noFQuZXi7
Max-Forwards: 70
From: 'Phone3' <> [email protected]/ * / >; tag = 8ce0e16a-Davis-448 b-8d9c-d0b4ad52589f
To: sip:[email protected]/ * /; tag = 13CB71C - 8 9
Call ID: d5ec8dba-bbe3-44ac-9902-0d52b327c514
CSeq: ACK 16131
Content-Length: 0

Apr 24 14:23:17.418: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
GUEST sip:[email protected]/ * /: SIP-5060/2.0
Via: SIP/2.0/UDP 192.168.40.11:5060; rport; branch = z9hG4bKPjfEwqupP3utIfGrv - WEaaI
hjdCy1q-hM
Max-Forwards: 70
From: 'Phone4' <> [email protected]/ * / >; tag = 93e43cc1-7e69-49ef-899 d-c81bd29eae11
To: sip:[email protected]/ * /.
"Contact: <> [email protected]/ * /: 5060 >; + sip.instance = '.<>
000-0000-0000-7c95f323b7b7 > '; + u.sip! DeviceName.CCM.Cisco.com = "SEP7C95F323B7B7"; +
u.SIP! Model.CCM.Cisco.com = "592"
Call ID: c9eb73ff-f89c-4aad-861d-724af1bd1f36
CSeq: INVITE 27309
Allow: PRACK, GUEST, ACK, BYE, CANCEL, update, SUBSCRIBE, NOTIFY, REFER, MESSAG
E, OPTIONS
User-Agent: Cisco-CP3905/9.2.1
Support: replaces,join,sdp-anat,norefersub,extended-refer,X-cisco-callinfo,X-c
ISCO-ServiceUri,X-Cisco-escapecodes,X-Cisco-Service-Control,X-Cisco-monrec,X-CIS
Co-config,X-Cisco-SIS-4.0.0,X-Cisco-xsi-7.0.1
Expires: 180
Accept: application/sdp
Allow-events: kpml, dialog box
Remote-Party-ID: 'Phone4'<>[email protected]/ * /: 5060 >; intimacy = off
Content-Type: application/sdp
Content-Length: 294

v = 0
o =-2208995137 2208995137 IN IP4 192.168.40.11
s = FOXPHONE
c = IN IP4 192.168.40.11
t = 0 0
a = X - nat:0
m = audio RTP/AVP 0 8 18 111 16388
a = rtpmap:0 PCMU/8000
a = rtpmap:8 PCMA/8000
a G729/8000 rtpmap:18 =
a = annex b fmtp:18 = yes
a = sendrecv
a rtpmap:111 telephone-event/8000 =
a = fmtp:111 0-15

Apr 24 14:23:17.422: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Envoy:
SIP/2.0 488 Media is not Acceptable
Via: SIP/2.0/UDP 192.168.40.11:5060; rport; branch = z9hG4bKPjfEwqupP3utIfGrv - WEaaI
hjdCy1q-hM
From: 'Phone4' <> [email protected]/ * / >; tag = 93e43cc1-7e69-49ef-899 d-c81bd29eae11
To: sip:[email protected]/ * /; tag = 13DA698 - 1 25
Date: Thu, April 24, 2014 14:23:17 GMT
Call ID: c9eb73ff-f89c-4aad-861d-724af1bd1f36
CSeq: INVITE 27309
Allow-events: telephone-event
WARNING: 304 192.168.0.1 "Media type (s) not available.
Reason: Q.850; cause = 65
Server: Cisco-SIPGateway/IOS-15.2.4.M4
Content-Length: 0

Apr 24 14:23:17.434: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP ACK:[email protected]/ * /: SIP-5060/2.0
Via: SIP/2.0/UDP 192.168.40.11:5060; rport; branch = z9hG4bKPjfEwqupP3utIfGrv - WEaaI
hjdCy1q-hM
Max-Forwards: 70
From: 'Phone4' <> [email protected]/ * / >; tag = 93e43cc1-7e69-49ef-899 d-c81bd29eae11
To: sip:[email protected]/ * /; tag = 13DA698 - 1 25
Call ID: c9eb73ff-f89c-4aad-861d-724af1bd1f36
CSeq: 27309 ACK
Content-Length: 0

Apr 24 14:23:19.690: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
GUEST sip:[email protected]/ * /: SIP-5060/2.0
Via: SIP/2.0/UDP 192.168.40.11:5060; rport; branch = z9hG4bKPjfW2oLTIRE62UkXCzpFyuAa
gpbtooQCuv
Max-Forwards: 70
From: 'Phone4' <> [email protected]/ * / >; tag = e8ed6eb5-8653-4154-93ab-1755749c84a6
To: sip:[email protected]/ * /.
"Contact: <> [email protected]/ * /: 5060 >; + sip.instance = '.<>
000-0000-0000-7c95f323b7b7 > '; + u.sip! DeviceName.CCM.Cisco.com = "SEP7C95F323B7B7"; +
u.SIP! Model.CCM.Cisco.com = "592"
Call ID: aa729730-fc15-4fc4-8bee-8d3e9840cc8b
CSeq: 10138 INVITE
Allow: PRACK, GUEST, ACK, BYE, CANCEL, update, SUBSCRIBE, NOTIFY, REFER, MESSAG
E, OPTIONS
User-Agent: Cisco-CP3905/9.2.1
Support: replaces,join,sdp-anat,norefersub,extended-refer,X-cisco-callinfo,X-c
ISCO-ServiceUri,X-Cisco-escapecodes,X-Cisco-Service-Control,X-Cisco-monrec,X-CIS
Co-config,X-Cisco-SIS-4.0.0,X-Cisco-xsi-7.0.1
Expires: 180
Accept: application/sdp
Allow-events: kpml, dialog box
Remote-Party-ID: 'Phone4'<>[email protected]/ * /: 5060 >; intimacy = off
Content-Type: application/sdp
Content-Length: 294

v = 0
o =-2208995139 2208995139 IN IP4 192.168.40.11
s = FOXPHONE
c = IN IP4 192.168.40.11
t = 0 0
a = X - nat:0
m = audio RTP/AVP 0 8 18 111 16390
a = rtpmap:0 PCMU/8000
a = rtpmap:8 PCMA/8000
a G729/8000 rtpmap:18 =
a = annex b fmtp:18 = yes
a = sendrecv
a rtpmap:111 telephone-event/8000 =
a = fmtp:111 0-15

Apr 24 14:23:19.694: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Envoy:
SIP/2.0 488 Media is not Acceptable
Via: SIP/2.0/UDP 192.168.40.11:5060; rport; branch = z9hG4bKPjfW2oLTIRE62UkXCzpFyuAa
gpbtooQCuv
From: 'Phone4' <> [email protected]/ * / >; tag = e8ed6eb5-8653-4154-93ab-1755749c84a6
To: sip:[email protected]/ * /; tag = 13DAF78-5 b 0
Date: Thu, April 24, 2014 14:23:19 GMT
Call ID: aa729730-fc15-4fc4-8bee-8d3e9840cc8b
CSeq: 10138 INVITE
Allow-events: telephone-event
WARNING: 304 192.168.0.1 "Media type (s) not available.
Reason: Q.850; cause = 65
Server: Cisco-SIPGateway/IOS-15.2.4.M4
Content-Length: 0

Apr 24 14:23:19.706: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP ACK:[email protected]/ * /: SIP-5060/2.0
Via: SIP/2.0/UDP 192.168.40.11:5060; rport; branch = z9hG4bKPjfW2oLTIRE62UkXCzpFyuAa
gpbtooQCuv
Max-Forwards: 70
From: 'Phone4' <> [email protected]/ * / >; tag = e8ed6eb5-8653-4154-93ab-1755749c84a6
To: sip:[email protected]/ * /; tag = 13DAF78-5 b 0
Call ID: aa729730-fc15-4fc4-8bee-8d3e9840cc8b
CSeq: 10138 ACK
Content-Length: 0

Apr 24 14:23:21.194: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
GUEST sip:[email protected]/ * /: SIP-5060/2.0
Via: SIP/2.0/UDP 192.168.40.11:5060; rport; branch = z9hG4bKPj1I0P8BoX0CIMO3qfwTYEWe
y.fSEokHib
Max-Forwards: 70
From: 'Phone4' <> [email protected]/ * / >; tag = d56894bf-896e-42b7-8b69-5da16d2337bc
To: sip:[email protected]/ * /.
"Contact: <> [email protected]/ * /: 5060 >; + sip.instance = '.<>
000-0000-0000-7c95f323b7b7 > '; + u.sip! DeviceName.CCM.Cisco.com = "SEP7C95F323B7B7"; +
u.SIP! Model.CCM.Cisco.com = "592"
Call ID: e472fb51-35f0-4f2c-a84a-f0efd0cf9f2e
CSeq: INVITE 12021
Allow: PRACK, GUEST, ACK, BYE, CANCEL, update, SUBSCRIBE, NOTIFY, REFER, MESSAG
E, OPTIONS
User-Agent: Cisco-CP3905/9.2.1
Support: replaces,join,sdp-anat,norefersub,extended-refer,X-cisco-callinfo,X-c
ISCO-ServiceUri,X-Cisco-escapecodes,X-Cisco-Service-Control,X-Cisco-monrec,X-CIS
Co-config,X-Cisco-SIS-4.0.0,X-Cisco-xsi-7.0.1
Expires: 180
Accept: application/sdp
Allow-events: kpml, dialog box
Remote-Party-ID: 'Phone4'<>[email protected]/ * /: 5060 >; intimacy = off
Content-Type: application/sdp
Content-Length: 294

v = 0
o =-2208995140 2208995140 IN IP4 192.168.40.11
s = FOXPHONE
c = IN IP4 192.168.40.11
t = 0 0
a = X - nat:0
m = audio RTP/AVP 0 8 18 111 16392
a = rtpmap:0 PCMU/8000
a = rtpmap:8 PCMA/8000
a G729/8000 rtpmap:18 =
a = annex b fmtp:18 = yes
a = sendrecv
a rtpmap:111 telephone-event/8000 =
a = fmtp:111 0-15

Apr 24 14:23:21.198: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Envoy:
SIP/2.0 488 Media is not Acceptable
Via: SIP/2.0/UDP 192.168.40.11:5060; rport; branch = z9hG4bKPj1I0P8BoX0CIMO3qfwTYEWe
y.fSEokHib
From: 'Phone4' <> [email protected]/ * / >; tag = d56894bf-896e-42b7-8b69-5da16d2337bc
To: sip:[email protected]/ * /; tag = 13DB558-1845
Date: Thu, April 24, 2014 14:23:21 GMT
Call ID: e472fb51-35f0-4f2c-a84a-f0efd0cf9f2e
CSeq: INVITE 12021
Allow-events: telephone-event
WARNING: 304 192.168.0.1 "Media type (s) not available.
Reason: Q.850; cause = 65
Server: Cisco-SIPGateway/IOS-15.2.4.M4
Content-Length: 0

Apr 24 14:23:21.210: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP ACK:[email protected]/ * /: SIP-5060/2.0
Via: SIP/2.0/UDP 192.168.40.11:5060; rport; branch = z9hG4bKPj1I0P8BoX0CIMO3qfwTYEWe
y.fSEokHib
Max-Forwards: 70
From: 'Phone4' <> [email protected]/ * / >; tag = d56894bf-896e-42b7-8b69-5da16d2337bc
To: sip:[email protected]/ * /; tag = 13DB558-1845
Call ID: e472fb51-35f0-4f2c-a84a-f0efd0cf9f2e
CSeq: 12021 ACK
Content-Length: 0

Apr 24 14:23:23.262: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
GUEST sip:[email protected]/ * /: SIP-5060/2.0
Via: SIP/2.0/UDP 192.168.40.11:5060; rport; branch = z9hG4bKPjzDEPLi9QFHsCPzfcVR.g72
hEIRSMbivf
Max-Forwards: 70
From: 'Phone4' <> [email protected]/ * / >; tag = 3dffd3d9-8f9a-48e7-b9a0-9a9f4c6c984d
To: sip:[email protected]/ * /.
"Contact: <> [email protected]/ * /: 5060 >; + sip.instance = '.<>
000-0000-0000-7c95f323b7b7 > '; + u.sip! DeviceName.CCM.Cisco.com = "SEP7C95F323B7B7"; +
u.SIP! Model.CCM.Cisco.com = "592"
Call ID: ea647ad6-36d2-414e-83d3-0e04b35e8129
CSeq: INVITE 26904
Allow: PRACK, GUEST, ACK, BYE, CANCEL, update, SUBSCRIBE, NOTIFY, REFER, MESSAG
E, OPTIONS
User-Agent: Cisco-CP3905/9.2.1
Support: replaces,join,sdp-anat,norefersub,extended-refer,X-cisco-callinfo,X-c
ISCO-ServiceUri,X-Cisco-escapecodes,X-Cisco-Service-Control,X-Cisco-monrec,X-CIS
Co-config,X-Cisco-SIS-4.0.0,X-Cisco-xsi-7.0.1
Expires: 180
Accept: application/sdp
Allow-events: kpml, dialog box
Remote-Party-ID: 'Phone4'<>[email protected]/ * /: 5060 >; intimacy = off
Content-Type: application/sdp
Content-Length: 294

v = 0
o =-2208995142 2208995142 IN IP4 192.168.40.11
s = FOXPHONE
c = IN IP4 192.168.40.11
t = 0 0
a = X - nat:0
m = audio RTP/AVP 0 8 18 111 16386
a = rtpmap:0 PCMU/8000
a = rtpmap:8 PCMA/8000
a G729/8000 rtpmap:18 =
a = annex b fmtp:18 = yes
a = sendrecv
a rtpmap:111 telephone-event/8000 =
a = fmtp:111 0-15

Apr 24 14:23:23.266: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Envoy:
SIP/2.0 488 Media is not Acceptable
Via: SIP/2.0/UDP 192.168.40.11:5060; rport; branch = z9hG4bKPjzDEPLi9QFHsCPzfcVR.g72
hEIRSMbivf
From: 'Phone4' <> [email protected]/ * / >; tag = 3dffd3d9-8f9a-48e7-b9a0-9a9f4c6c984d
To: sip:[email protected]/ * /; tag = 13DBD6C-1273
Date: Thu, April 24, 2014 14:23:23 GMT
Call ID: ea647ad6-36d2-414e-83d3-0e04b35e8129
CSeq: INVITE 26904
Allow-events: telephone-event
WARNING: 304 192.168.0.1 "Media type (s) not available.
Reason: Q.850; cause = 65
Server: Cisco-SIPGateway/IOS-15.2.4.M4
Content-Length: 0

Apr 24 14:23:23.278: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP ACK:[email protected]/ * /: SIP-5060/2.0
Via: SIP/2.0/UDP 192.168.40.11:5060; rport; branch = z9hG4bKPjzDEPLi9QFHsCPzfcVR.g72
hEIRSMbivf
Max-Forwards: 70
From: 'Phone4' <> [email protected]/ * / >; tag = 3dffd3d9-8f9a-48e7-b9a0-9a9f4c6c984d
To: sip:[email protected]/ * /; tag = 13DBD6C-1273
Call ID: ea647ad6-36d2-414e-83d3-0e04b35e8129
CSeq: 26904 ACK
Content-Length: 0

Hello

In my view, that the call fails because the phone of 3905 Announces g729annexB codec. could you please try to configure "voice-class codec" or a "codec G711a/G711u' command under voice register pools 1-4 and check the behavior?

Suresh

Please note all useful posts

duplicate calls made using the alias between two codecs C40

We have 2 new codecs C40. When calling between the two using the alias, it seems that 2 calls are made. In a sense, we get a very quick disonnect message upon acceptance of the call. In the other direction, the call comes in, but on disconnect, we get a message saying missed call.

Seems that the search rules can be duplication of the call some how - probably him endpoint becomes a call SIP and H.323 call at the same time. Could you post the call details and history/details of research?

Sent by Cisco Support technique iPhone App

Incoming SIP - SP CUBE is not of translations

Perplexed as to why the incoming calls from SIP service provider do not correspond to the translation in CUBE

I have a number presented on the incoming CUBE SIP trunk and need to get rid of the figures for the last 3 numbers to present to the CUCM. The test voice translation works, but it seems that the incoming number provided by the supplier is not hit or corresponding to the translation rule.

Incoming dial peer config:

Dial-peer voice voip 60
Description incoming PSTN (elite) to the CUBE
translation-profile entering EliteSIP-DDI-numbers-inbound
session protocol sipv2
incoming called number 44239...
codec voice-class 1
DTMF-relay rtp - nte sip-kpml
No vad

Profile and set the configuration of translation

voice translation rule 44239
rule 1 / ^ 442392006.
rule 2 / ^ \+442392006/ / /.
!
!
voice translation-profile EliteSIP-DDI-numbers-inbound
definition of 44239 called

The result of the translation:

Matched with rule 2
Original number: + 442392006339 translated number: 339
Number of origin type: no number translation type: no
Original number plan: no number plan translated: no

BE6000S #test voice translation rule 44239 442392006339
Matched with rule 1
Original number: 442392006339 translated number: 339
Number of origin type: no number translation type: no
Original number plan: no number plan translated: no

The translation of debugging output:

Voice translation of BE6000S #debug
VoIP translation rule debugging is enabled
BE6000S #.
SIP: Attempt to analyze the attribute not supported at the level of the media
SIP: Attempt to analyze the attribute not supported at the level of the media
065139: June 7 23:35:29.157: //-1/5A562434A112/RXRULE/regxrule_stack_pop_RegXruleNumInfo: stack = 0x3F5552E8; Count = 1
065140: June 7 23:35:29.157: //-1/5A562434A112/RXRULE/regxrule_stack_pop_callinfo_internal: infonum = 0x421F2934
065141: June 7 23:35:29.161: //-1/5A562434A112/RXRULE/regxrule_stack_push_RegXruleNumInfo_internal: stack = 0x3F5552E8; Count = 1
065142: June 7 23:35:29.161: //-1/5A562434A112/RXRULE/regxrule_stack_pop_RegXruleNumInfo: stack = 0x3F5552E8; Count = 1
065143: June 7 23:35:29.161: //-1/5A562434A112/RXRULE/regxrule_stack_pop_callinfo_internal: infonum = 0x421F2934
065144: June 7 23:35:29.161: //-1/5A562434A112/RXRULE/regxrule_stack_push_RegXruleNumInfo_internal: stack = 0x3F5552E8; Count = 1
065145: June 7 23:35:29.165: //-1/xxxxxxxxxxxx/RXRULE/sed_subst: no match! number = matchPattern = id; [; ] * replacePattern$ id =
065146: June 7 23:35:32.157: //-1/5A562434A112/RXRULE/regxrule_stack_pop_callinfo_internal: infonum = 0x0
065147: June 7 23:35:32.169: //-1/5A562434A112/RXRULE/regxrule_stack_pop_RegXruleNumInfo: stack = 0x3F5552E8; Count = 1
065148: June 7 23:35:32.169: //-1/5A562434A112/RXRULE/regxrule_stack_pop_callinfo_internal: infonum = 0x421F2934

Debug messages ccsip just to make sure the call come and the DNIS format (btw - which bit of the track to show the DNIS?)

BE6000S #debug ccsip messages
Call SIP tracing messages is enabled
BE6000S #.
065149: June 7 23:38:16.925: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
GUEST sip:[email protected]/ * /: SIP-5060/2.0
Record-Route:
Via: SIP/2.0/UDP 217.68.246.241:5060; branch = z9hG4bKe4be.24390fd700572c75f3247fa6444e9fcc.0
Max-Forwards: 16
To: <> [email protected]/ * /: 5060 >
From: <> [email protected]/ * / >; tag = as6b74b830
Call ID: [email protected]/ * /: 5050
Contact: <> [email protected]/ * /: 5060 >
CSeq: INVITE 102
User-Agent: Elite hosted voice
Date: Tuesday, June 7, 2016 23:38:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X voipnow-did: + 442392006339
X voipnow-extension: 0071 * 001
X voipnow pbx: 3a5b131e3e
X voipnow-infrastructureid: 92f21508
X voipnow-did: + 442392006339
Content-Type: application/sdp
Content-Length: 520

Ideas?

Dear MEP,

I think that if you add + to incoming called number, it should solve the problem as provider sends with +.

Incoming called number + 44239...

Also run dialpeer voip debug to see dial-peers are put in correspondence on incoming direction of ITSP.thanks

DTMF in SIP Trunk problem

Hello

I have a problem in case of detection of the DTMF

We have a SIP of the ITSP Trunk and everything is ok except DTMF.

The sip trunk is between ITSP and router 3945

ITSP <->3945 <->CUCM 10.5

I have try all method such as rtp - nte, h245 alphanumeric and h245-signal, info-sip, sip-kpml,... in line-peer to ITSPs

ITSP say that he send with standard RFC2833 dtmf (it equals to rtp-net, I know), but when I am 'debug ccsip message', the results show the dtmf with rfc2833 sends us

16 August 13:52:28.991: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip: [email protected] / * /: 5060. user = phone SIP/2.0

Via: SIP/2.0/UDP 10.105.40.34:5060; branch = z9hG4bKejhh8jixkobhnb7ykvi87vuj8

Call ID: [email protected]/ * /.

From: sip: [email protected] / * />; tag = c3cx3vcw-CC-40

To: sip: [email protected] / * /; user = phone >

CSeq: 1 INVITE

Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTRY, PRACK, INFO SUBSCRIBE, NOTIFY, updates, MESSAGE, see

Max-Forwards: 69

Supported: 100rel, timer

User-Agent: Huawei SoftX3000 V300R010

Session time-out: 300

Min - SE: 90

Contact: sip: [email protected] / * /: 5060; user = phone >

Content-Length: 374

Content-Type: application/sdp

v = 0

o = HuaweiSoftX3000 11042757 11042757 IN IP4 10.105.40.34

s = call Sip

c = IN IP4 10.105.40.34

t = 0 0

m = audio 27762 RTP / AVP 8 0 18 4 2 98 99 102

a = rtpmap:8 PCMA/8000

a = rtpmap:0 PCMU/8000

a G729/8000 rtpmap:18 =

a = rtpmap:4 G723/8000

a = rtpmap:2 G726-32/8000

a = rtpmap:98 G726-40/8000

a = rtpmap:99 G726-32/8000

a = rtpmap:102 G726-24/8000

a = ptime:20

a = fmtp:18 annex b = No.

It is a message to guest (with sdp) of ITSP

As you can see the line with red color must have a code with number of 101 but rather a code with number of 18

In my "ccsip media debug' output show that the method of negotiation between me and itsp 'incoming voice. '

It's my router config:

voip phone service
No IP trust to authenticate
allow connections h323 to SIP
allow connections sip h323
allow sip to sip connections
SIP
interface FastEthernet0/0/1 source control binding
bind media source interface FastEthernet0/0/1
min - to 300 session expires-300
!

Dial-peer voice 2 voip---> router CUCM and vice versa
translation-profile outgoing toos
destination-model 42584...
session protocol sipv2
session target ipv4:10.20.30.70
Codec g711ulaw
DTMF-relay rtp - nte
!
VoIP voice 10 Dial - peer---> router for ITSP and vice versa
destination-model. T
session protocol sipv2
session target ipv4:10.105.40.34
incoming called-number. T
DTMF-relay rtp - nte
Codec g711ulaw

I have configured cucm with a sip section to my favorite router with active PSG and RFC2833

BUT HE DIDN'T THERE WAS NO DETECTION OF DTMF IN INCOMING CALLS AND OUTGOING

I even test dial-position 10 without any method of dtmf-relay because I want to configure it with the incoming voice method, but it does not work

I change the codec but does not solve the problem

There is an interesting point and that is, if use Elastix 3945 of the router and configure dtmf among elastix and itsp as "inbound" method and method dtmf among elastix and cucm as 'rfc2833' everything is OK (ITSP<--Inbound--> Elastix <--rfc2833-->CUCM)

Please give me a solution to solve the problem between Cisco 3945 and ITSP

Concerning

It would be more accurate to say that the problem has been bypassed by using a transcoding resource. The correct resolution here would be to open a ticket to trouble with the ITSP. I agree that the PROMPT message you show doesn't include advertising RFC2833 in the SDP offer. It is a problem that only the carrier can rectify.

CUCME no calls incoming, outgoing calls okay

Hello everyone,

I'm setting up a CUCME with SIP trunk, I can make calls outside, but I can´t receive everything from the outside, it's my second time as a SIP configuration

I ve use debug command voice dialpeer all to check was happening, but I can´t find the problem.

This is my config:

IP server host sip - A.B.C.D

!

voip phone service

list of approved IP addresses

IPv4 A.B.C.D 255.255.255.252

!

translation of the voice-rule 1

rule 1 / 325277\ (\) / / 1\1 /.

!

voice translation-profile IN

translate 1 called

!

Dial-peer voice 1 voip

Description * incoming SIP trunk call *.

entrants IN translation-profile

session protocol sipv2

session target sip-Server

incoming called-number.

codec voice-class 1

voice-class sip dtmf-relay rtp - nte force

DTMF-relay rtp - nte

No vad

!

ePhone-dn 1

number 100

Description of RECEPTION

!

ePhone 2

address Mac YYYY. BENAMER. CCBC

ePhone-model 1

type 7942

Keep-Conference

button 1:1

NOTE: The IP address are hidden, just for safety

Here is the output from my debug/tests:

voice translation rule 1 32527700 #test

Matched with rule 1

Original number: 32527700 translated number: 100

Number of origin type: no number translation type: no

Original number plan: no number plan translated: no

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

Number = 32527700, called number = 32527700, Peer Info Type = DIALPEER_INFO_SPEECH

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

Associate the rule of = DP_MATCH_DEST; Called number = 32527700

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:

Dial String = 32527700, expanded String = 32527700 number = 32527700T

Timeout = TRUE, incoming = FALSE, Peer Info Type = DIALPEER_INFO_SPEECH

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:

Result =-1

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

No outbound dial-peer does not; Result = NO_MATCH(-1)

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

dialstring = 32527700, saf_enabled = 1 saf_dndb_lookup = 1, dp_result =-1

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:

Result = NO_MATCH(-1)

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

Number = 59513212, called number =, Voice-Interface = 0 x 0.

Timeout = TRUE, Peer Encap Type = ENCAP_VOIP, Type of research peer = PEER_TYPE_VOICE,

Peer Type Info = DIALPEER_INFO_SPEECH

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

Associate the rule of = DP_MATCH_ANSWER; Number = 59513212

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:

Incoming = TRUE, expand = FALSE

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:

Dial String =, expanded string =, number of calls = 59513212T

Timeout = TRUE, incoming = TRUE, Peer Info Type = DIALPEER_INFO_SPEECH

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:

Result =-1

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:[email protected]/ * /.

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

Associate the rule of = DP_MATCH_ORIGINATE; Number = 59513212

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:

Incoming = TRUE, expand = FALSE

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:

Dial String =, expanded string =, number of calls = 59513212T

Timeout = TRUE, incoming = TRUE, Peer Info Type = DIALPEER_INFO_SPEECH

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:

Result =-1

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:[email protected]/ * /.

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

Result = NO_MATCH(-1) after all rules attempt Match

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result =-1

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeer:[email protected]/ * /.

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:

Number = 59513212, called number = 32527700, Voice-Interface = 0 x 0.

Timeout = TRUE, Peer Encap Type = ENCAP_VOIP, Type of research peer = PEER_TYPE_VOICE,

Peer Type Info = DIALPEER_INFO_SPEECH

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:

Associate the rule of = DP_MATCH_VIA_URI; URI = SIP:A.B.C.D:5060

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:

Incoming = TRUE, expand = FALSE

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:

Dial String =, expanded string =, number of calls =

Timeout = TRUE, incoming = TRUE, Peer Info Type = DIALPEER_INFO_SPEECH

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:

Result =-1

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:[email protected]/ * /.

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:

Associate the rule of = DP_MATCH_REQUEST_URI; URI = sip:[email protected]/ * /: 5060; user = phone

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:

Incoming = TRUE, expand = FALSE

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:

Dial String =, expanded string =, number of calls =

Timeout = TRUE, incoming = TRUE, Peer Info Type = DIALPEER_INFO_SPEECH

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:

Result =-1

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:[email protected]/ * /.

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:

Associate the rule of = DP_MATCH_TO_URI; URI = sip:[email protected]/ * /; user = phone

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:

Incoming = TRUE, expand = FALSE

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:

Dial String =, expanded string =, number of calls =

Timeout = TRUE, incoming = TRUE, Peer Info Type = DIALPEER_INFO_SPEECH

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:

Result =-1

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:[email protected]/ * /.

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:

Associate the rule of = DP_MATCH_FROM_URI; URI = sip:[email protected]/ * /; user = phone

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:

Incoming = TRUE, expand = FALSE

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:

Dial String =, expanded string =, number of calls =

Timeout = TRUE, incoming = TRUE, Peer Info Type = DIALPEER_INFO_SPEECH

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:

Result =-1

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:[email protected]/ * /.

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:

Associate the rule of = DP_MATCH_INCOMING_DNIS; Called number = 32527700

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:

Incoming = TRUE, expand = FALSE

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:

Dial String = 32527700, expanded String = 32527700 number =

Timeout = TRUE, incoming = TRUE, Peer Info Type = DIALPEER_INFO_SPEECH

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/MatchNextPeer:

Result = Success (0); Incoming dial-peer = 1 is set in correspondence

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:[email protected]/ * /.

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:

Result = Success (0) after DP_MATCH_INCOMING_DNIS; Incoming dial-peer = 1

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchSafModulePlugin:

dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result = 0

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerSPI:[email protected]/ * /.

Can someone help me?

Thanks in advance!

I looked on the other leg of the SIP messages appeal, here's the fault for the where incoming call is being failed because the session timer is too small, has received from the SBC (provider)

Call ID:

Call ID: [email protected]/ * /.

INVITE RECEIEVED SBC - SIP

* Jan 29 16:53:19: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

GUEST sip: 32527700 @(WAN): 5060; user = phone SIP/2.0

Via: SIP/2.0/UDP (SIP_SERVER): 5060; Branch = z9hG4bK776928550196f0d843ca0b092

Call ID: [email protected]/ * /.

From:; tag = 6e8b9968-CC-25

TO:

CSeq: 1 INVITE

Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTRY, PRACK, INFO SUBSCRIBE, NOTIFY, updates, MESSAGE, see

Max-Forwards: 70

Supported: 100rel, timer

User-Agent: Huawei SoftX3000 V300R601

Session time-out: 300

Min - SE: 90

Contact:

Content-Length: 376

Content-Type: application/sdp

v = 0

o = HuaweiSoftX3000 4507886 4507886 IN IP4 (SIP_SERVER)

s = call Sip

c = IN IP4 (SIP_SERVER)

t = 0 0

m = audio RTP 11554 / AVP 8 0 18 4 2 98 98 98

a = rtpmap:8 PCMA/8000

a = rtpmap:0 PCMU/8000

a G729/8000 rtpmap:18 =

a = rtpmap:4 G723/8000

a = rtpmap:2 G726-32/8000

a = rtpmap:98 G726-40/8000

a = rtpmap:98 G726-32/8000

a = rtpmap:98 G726-24/8000

a = ptime:20

a = fmtp:18 annex b = No.

In response to GUY sends 422

Envoy:

SIP/2.0 422 Session Timer too small

Via: SIP/2.0/UDP (SIP_SERVER): 5060; Branch = z9hG4bK776928550196f0d843ca0b092

From:; tag = 6e8b9968-CC-25

Up to:; tag = 4CD1E84-2094

Date: Wednesday, January 29, 2014 22:53:19 GMT

Call ID: [email protected]/ * /.

CSeq: 1 INVITE

Allow-events: telephone-event

Min - SE: 1800

Server: Cisco-SIPGateway/IOS-15.2.4.M

Content-Length: 0

According to rfc

If the Session time-out interval is too low for a proxy (i.e., lower)

that the value Min - SE that the proxy would argue), the

Proxy denies the request with a 422 response. This response

contains a header field in Min - TO identify the minimum session

meantime, she is ready to support. The UAC will try again, this time

including the header of Min - SE in the query field. The header field

contains the largest header field Min - SE that he observed in all 422

responses received previously. In this way, the minimum timer meets the

constraints of all proxies on the way.

http://www.Cisco.com/en/us/docs/iOS/voice/SIP/configuration/guide/sip_cg-msg_tmr_rspns.html#wp1056968

Response message 422

If the value of the Session header expires is too small, the UAS or proxy refuses the call with a response message 422 Session Timer too small . With 422 response, the proxy or the SAMU message includes a header of Min - SE, indicating the value of minimum session, he can accept. UAC can then try again the appeal with a higher value of session timer.

If a 422-response message is received after a GUEST query, the UAC can again INVITE him.

There is two way to fix this

1) asked the SBC (your SIP provider) the value change and the value of standards send the SIP invite session expires

(2) change the value of the Min - SE on the CME on demand

Run this Global Config on CME

voip phone service

allow sip to sip connection

SIP

90 min - to

BR,

Nadeem

Please note all the useful post.

SX20 call SIP

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