CUCM SIP road model Discussion

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Model of IPv4

(Required) Enter the domain, subdomain, IPv4 address or the IP address of the subnet.

For the use of model Domain Routing, entering a field of IPv4 model domain name that resolves to an IPv4 address. The domain name can contain the following characters: [, -,., 0-9, A - Z, a - z, *, and].

For the use of model of Routing IP address, enter an IPv4 address, the IPv4 model field that follows the format X.X.X.X, where X is a number between 0 and 255.

The address of IP subnet, in notation Classless Inter - Domain Routing (CIDR), x.x.x.x/y; where is the network prefix that denotes the number of bits of the address that will be the network address.

Tip	If the SIP trunk supports IPv6 or IPv4 and IPv6 (in mode dual-stack), configure the IPv6 in addition to IPv4 model model.

For any system to which you can route SIP, audio and/or video, as a VCS calls.

Or if you have an other PBX that record endpoints with URI and accepts a SIP trunks, you can use it too.

Tags: Cisco Support

Similar Questions

CUCM SIP-trunk multiple Ports

I would like to know if it is possible to create a Script of SIP standardization in CUCM which change the Destination SIP port based on the phone which makes the phone.

Current issue:

The ITSP provider asked that I send calls from different regions to the same address IP SBC but a different port.

Example:

Phone (A) makes a call

RP - RL - RG---> SIP Trunk (5060)---> CUBE (5060<-->5001)---> ITSP1

Telephone (B) makes a call

RP - RL - RG---> SIP Trunk (5060)--->---> CUBE ITSP1 (5060<-->5002)

I can do it on the CUBE, but which require several dial-Exchange and SIP profiles, is it possible in the CUCM to change the sip by device-pool port?.

Or is there another I can match and send calls via the same destination with different dial-peer sip ports.?

Thank you.

Zakiab,

You can do this, its all just impossible. Your signage ports are defined not on the endpoints, but on the B2BUA (CUCM or CUBE). The CUBE, you have the most flexibility, because you can change the port of signs based on your dial-peers. However, as you pointed out quite rightly, need you an insurmountable amount of dial-peers to have this for each endpoint. So I suggest tell you your ITSP to change their design or moving to a new.

PEI - SIP - CME - SIP - error CUCM Media is not Acceptable

Hello world

I have a problem with a TRUNK of SIP ITSP, the question is apparently "SIP/2.0 488 not acceptable media.

I tried several things, I Don t know how to solve this problem.

Outgoing calls is already OK, the problem is with incoming calls: of the ITSP to the CUCM.

I have this topology of the ITSP SIP TRUNK:

ITSP - sip sip - CME - CUCM

The CME configuration is:

Dial-peer voice voip 67
Description * SIP trunk ITSP *.
destination-model 591 [67]...
session protocol sipv2
session target ipv4:172.17.0.13
session udp transport
voice-class sip forced early offer
no interaction of dtmf
Codec g711ulaw

!

Dial-peer voice voip 68
Description * SIP trunk CUCM *.
reply-to address. T
session protocol sipv2
session target ipv4:172.16.6.3
voice-class sip forced early offer
Codec g711ulaw

!

SIP - ua
Disable-early-media 180
connection-reuse

!

voip phone service
list of approved IP addresses
IPv4 0.0.0.0
IPv4 0.0.0.0 0.0.0.0
h323 connections allow h323
allow connections h323 to SIP
allow connections sip h323
allow sip to sip connections
no service additional h450.7
no additional service moved temporarily sip
no service additional sip refer
Fax protocol t38 ls-redundancy version 0 0 hs-redundancy 0 help none
SIP
binding control source-interface GigabitEthernet0/1.5
bind media source-interface GigabitEthernet0/1.5
Registrar Server
offer-early forced

!

On the side of CUCM:

End point of media (checked)
Disable the media beginning on 180 (unchecked)
Requires the idle exchange of SDP for call Media Change (checked)
Early support for voice and video calls (checked)
Send send-receive SDP appealed INVITES (checked)

The result of "debug messages ccsip" and "debug dialpeer inout voice" is:

001062: 03:31:03: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
GUEST sip:[email protected]/ * /: 5060; user = phone SIP/2.0
Via: SIP/2.0/UDP 172.17.0.11:5060; branch = z9hG4bKi2mhl410do70mrc4s141.1
Call ID: [email protected]/ * /.
From: "70965999"<>[email protected]/ * /; transport = udp; user = phone >; tag = 2u2uavrx-CC-1013
To: '69200020'<>[email protected]/ * /; transport = udp; user = phone >
CSeq: 1 INVITE
Max-Forwards: 69
Contact:
Allow: INVITE, ACK, OPTIONS, CANCEL, INFO, BYE PRACK, NOTIFY, MESSAGE, UPDATE
P - asserted-Identity has:
Supported: 100rel, histinfo, prerequisite
P-early-Media: support
Content-Length: 362
Content-Type: application/sdp

v = 0
o = HuaweiSoftx3000 1102026905 1102026906 IN IP4 172.17.0.11
s = SipCall
c = IN IP4 172.17.0.11
t = 0 0
m = audio RTP/AVP 8 18 116 10386
a = rtpmap:8 PCMA/8000
a G729/8000 rtpmap:18 =
a rtpmap:116 telephone-event/8000 =
a = ptime:5
a = sendrecv local curr:qos
a = distance zero curr:qos
a = sendrecv local compulsory are: qos
a = sendrecv distance optional with: qos
a = 3gOoBTC

001063: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Number = 69200020, called number = 69200020, Peer Info Type = DIALPEER_INFO_SPEECH
001064: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Associate the rule of = DP_MATCH_DEST; Called number = 69200020
001065: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
No outbound dial-peer does not; Result = NO_MATCH(-1)
001066: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
dialstring = 69200020, saf_enabled = 1 saf_dndb_lookup = 1, dp_result =-1
001067: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
Result = NO_MATCH(-1)
001068: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Number = 70965999, 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
001069: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result = Success (0) after DP_MATCH_ANSWER; Incoming dial-peer = 68
001070: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result = 0
001071: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Number = 70965999, 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
001072: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result = Success (0) after DP_MATCH_ANSWER; Incoming dial-peer = 68
001073: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result = 0
001074: 03:31:03: //-1/CD12136099D5/DPM/dpAssociateIncomingPeerCore:
Number = 70965999, called number = 69200020, 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
001075: 03:31:03: //-1/CD12136099D5/DPM/dpAssociateIncomingPeerCore:
Result = Success (0) after DP_MATCH_ANSWER; Incoming dial-peer = 68
001076: 03:31:03: //-1/CD12136099D5/DPM/dpMatchSafModulePlugin:
dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result = 0
SIP: Attempt to analyze the attribute not supported at the level of the media
001077: 03:31:03: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Envoy:
SIP/2.0 488 Media is not Acceptable
Via: SIP/2.0/UDP 172.17.0.11:5060; branch = z9hG4bKi2mhl410do70mrc4s141.1
From: "70965999"<>[email protected]/ * /; transport = udp; user = phone >; tag = 2u2uavrx-CC-1013
To: '69200020'<>[email protected]/ * /; transport = udp; user = phone >; tag = C139A0-1755
Date: Thu, November 6, 2014 13:01:04 GMT
Call ID: [email protected]/ * /.
CSeq: 1 INVITE
Allow-events: telephone-event
Reason: Q.850; cause = 65
Server: Cisco-SIPGateway/IOS-15.3.3.M2
Content-Length: 0

001078: 03:31:03: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:

SIP GatewayTelf_JD #ACK:[email protected]/ * /: 5060; user = phone SIP/2.0
Via: SIP/2.0/UDP 172.17.0.11:5060; branch = z9hG4bKi2mhl410do70mrc4s141.1
CSeq: 1 ACK
Call ID: [email protected]/ * /.
From: "70965999"<>[email protected]/ * /; transport = udp; user = phone >; tag = 2u2uavrx-CC-1013
To: '69200020'<>[email protected]/ * /; transport = udp; user = phone >; tag = C139A0-1755
Max-Forwards: 69
Content-Length: 0

Little light at the end of the tunnel?

Thanks in advance!

Hello

can you collect debug voice ccapi inout & debugging ccsip GCE message and attach the logs here please?

your provider sends A Law G711 codec in the PROMPT message, but you have configured G711 U right in the dial-peers.

can you try to fix G711 has the right dial-peers and check out them? Also make sure you have TPMS in the MRGL applied to CUCM SIP Trunk.

is it possible, end point of video recorded in CUCM call standalon SX10 via SIP Trunk?

Dear all,

Need help :)

1.

is it possible, end point of video recorded in CUCM call standalon SX10 via SIP Trunk?

my customers will not buy TP - lic :)

Topology:

8900 (jabber) - CUCM-(SIP) - SX10

2.

Customer, have MCU machine.

If not possible. is it possible, meeting point of MCU use?

Topology:

8900 (jabber) - CUCM - MCU - SX10

1. Yes, but it is not that simple and straightforward that the SX10 can be reached by calling is the public IP address and CUCM does not support the IP address numbering is not a SIP address format...

2. Yes, using the MCU would be much simpler, easier and, in my opinion, a better solution to be implemented. Here is an example of how a stand-alone SX10 could dial the IP address of the MCU standard automatic.

You can also assign a SIP URI address to the meeting point which the SX10 can connect to.

/Jens

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

Calls to the CUCM VCS for non-local domains

We have a situation where there is a single VCS control Expressway, CUCM Pub and and Sub 9.1. In the documentation it says to add a SIP road ground is *. * or *. VCs.domain.

Since we have only a single control VCs and highway, clustering is not required and the SIP domain would for example company.com.

CUCM FQDN is set to domain.com.

Now, the domain configured on CUCM being company.com, name will be the *. * model of routing SIP send everything to the VCS with a URI?

Example - Bob and Tom EX - 60 end points are all two CUCM registrants. URI have been set up for their the DN @domain.com

Dials of Bob [email protected] / * /

CUCM will try and send this call to VCS first, even if Tom EX - 60 entered CUCM?

SIP routing model applies to endpoints calling other points of endpoint registered to CUCM?

Hi Daniel,.

The CUCM that command searching table of URI, is similar to DNs (directory numbers) and RPs (model road). Basically the CUCM will attempt to find the URI locally. If it is available, it will send the call to the URI. If it is not found, it will try to find a match in the SIP routing model. Thus, in you case the default model URI (* or *. *). This is done if the URI does not exist locally.

ARO

S.K.

URI external numbering of CUCM

Dear all,

I currently have a configuration of laboratory for communication between Cisco CUCM and the Cisco VCS. Here's what I've been able to achieve in the distance

Call of CUCM to the endpoints registered to the VCS

Calls from end points registered in the VCS to the endpoints registered to the CUCM

What I want to achieve is external URI numbering of the CUCM to external domains through the VCS highway.

If for example I dial [email protected] / * /, it is resolved to [email protected] / * /-address of the VCS (configured on the trunk between the CUCM and VCS). I know this is a default behavior.

I need to know how I can change the CUCM such that it provides the exact address, I'm composing, i.e. [email protected] / * / VCs without any change made

Kind regards

Osho

What version of CUCM and VCS are you currently using?

What is your CUCM and VCS setup/detail as in:
- CUCM: VCS SIP Trunk (trunk and profile security SIP Trunk SIP)
- CUCM: Models of road
- CUCM: Models SIP road
- CUCM: SIP rule standards applied to the trunk
- VCS control: you can join a history of research output

TCS, CUCM, EX90 recording solution

Dear all

I need to implement registration between telepresence [EX90, SX20] video endpoint. There is the solution integrated on TCS and CUCM, The SIP trunk between TC and CUCM with the model number of routing TCS.

Example of

TC - Extension 9000 [trunk Sip with CUCM]

EX90 - Extension 1001 [register with CUCM]

SX20 Extension 0 1002 [register with CUCM]

Whenever EX90 call SX20., way to trigger the recording to video/voice Trade Commissioner?

If I misunderstand the concept., please help give me an opinion.

Reference: http://www.cisco.com/c/en/us/td/docs/telepresence/tcs/6_2/administration...

Best regards

Goal

If the SX20 or EX90 Multisite option key, one of them can host the Conference. Any endpoint, it is saying the SX20 must make the appeal out of the TCS. Suggest you take a look at the sheets and admin guides, they have some information about Multisite.

SIP trunk CUBE with Callcentric - incoming unanswered call

I'm doing some tests with a Sip trunk with a provider called Callcentric.

It is a CUBE scenario. I use a SIP to the CUCM trunk.

I have a Cisco Callmanager and a virtual router 7200 (GNS3) as a gateway (C7200-ADVENTERPRISEK9-M), Version 12.4 (24).

A CPIC connected to Callmanager, I call out to PSTN and it works perfectly.

When I do an incoming call to the PSTN to the softphone destination, she sounds, but when I press the answer button, the call is not connected and the analog phone in RTC can listen back ringtone.

Do you have any idea what it could be?

Some relevant configurations:

voip phone service
allow sip to sip connections
Fax protocol cisco
SIP
interface FastEthernet0/0 source control binding
bind media source interface FastEthernet0/0
Registrar Server

voice class codec 1
g711ulaw codec preference 1

translation of the voice-rule 1
rule 1 / ^ 8 / /0056/
!
voice translation-rule 2
rule 1 5.0 / /17772114zzz/
!
voice translation-rule 3
rule 1 /17772114zzz/ /500/

voice translation-profile IN
definition of 3 called
!
FLIGHT voice translation-profile
definition of call 2
translate 1 called

Dial-peer voice 1 voip
CALLCENTRIC description
entrants IN translation-profile
translation-profile outgoing OUT
destination-model 8.T
codec voice-class 1
session protocol sipv2
session target sip-Server
incoming called-number 17772114zzz
SIP DTMF-relay-notify rtp - nte
!
Dial-peer voice 2 voip
CUCM description
destination-model 500
media stream-autour
codec voice-class 1
session protocol sipv2
session target ipv4:192.168.10.116
incoming called number 8.T
SIP DTMF-relay-notify rtp - nte

SIP - ua
credentials of username, password 7 17772114zzz 094F471B100928 callcentric.com Kingdom
authentication username 17772114zzz password 7 121A0C051B0803 callcentric.com Kingdom
no remote-party-id
Registrar dns:callcentric.com expires 3600
DNS:callcentric.com SIP server
Home-Office

Thank you guys.

Hello

Please configure the sip requires low profile to remove the header and apply this profile in the dial-peer 1 outgoing.

SIP-class voice profiles 1

response header 200 sip requires DELETE

If this does not work under Dial-peers, try to apply globally.

voip phone service

SIP

SIP profiles 1

Suresh

Please note all useful posts

Supported SIP URI characters

Hello

I'm currently setting up interoperability between Cisco control VCS X7.1 and CUCM V7.1.3.

The IPT dial plan consists of site prefixes and area codes that make up the full DN phones, this unique name is also starts with a #.

For example #0044014761

My problem is (I think), that # is a character in charge of SIP URI that is exposed in the deployment guide. I tried to use 35%, which is the ASCII value for #; Oddly enough, when you use that I can solve the alias against the CUCM SIP trunk, but when you attempt to dial for an endpoint, it fails.

I know that my SIP trunk is OK, as I can carry 9.* by it on the break-out CUCM PSTN calls.

Anyone encountered this problem before, or been able to work around it?

Thank you, if

Sent by Cisco Support technique iPhone App

For your reference, it was similar (quite well) discussion there, a month

https://supportforums.cisco.com/thread/2152374?tstart=210.

CUCM 9.1.2-> 10.5 failure (2)

Hi all

I am trying to upgrade our CUCM 9.1.2 (SU2)--> 10.5 (2) (UCSInstall_UCOS_10.5.2.10000 - 5.sgn.iso)

I believe that I made the requirements of upgrade:
- Our equipment is Cisco UCS with old 'Medium' UCS tested baseline for 2500 users. Each node has 2 x vCPUs.
- 9.1.2 (SU2) tp 10.5 (2) is described as a "Direct Upgrade.
- I have / had at least 30 GB of free in the common partition on each node
- ciscocm.Version3 - keys.cop has been installed on all nodes
- I have not installed the file COP refresh_upgrade, I'm upgrading to 9.1.2 (i.e. no pre version 8.5)
- The specification of the virtual machine meets all the conditions for 10.5 (2)
Installation goes for about 1.5-2 hours before it is "Post-installation" tasks where is fails at the point 6 or 6 of "Installation component database.

E installation log shows:

24/01/2015-17:24:02 component_install | File: / opt/Cisco / install/bin/component_install:807, function: exec_progmeter(), / opt/cisco/install/bin/progmeter has not (1) |

24/01/2015-17:24:02 appmanager.sh | Error function, File:/usr/local/bin/base_scripts/appmanager.sh:273, internal: refresh_upgrade(), does not have refresh_upgrade components infrastructure_post |

24/01/2015-17:24:02 post_install | File: / opt/Cisco / install/bin/post_install:961, function: install_applications(), /usr/local/bin/base_scripts/appmanager.sh-r-mise at level failed (1) |

24/01/2015-17:24:02 post_install | Output with result 1 |

24/01/2015-17:24:02 post_install | INSTALL_TYPE = "refresh upgrade |

It seems very similar to the Cisco support forum question: https://supportforums.cisco.com/printpdf/12334191; However, I was able to remove the AUNP numbering plan; I suspect because I have a number of models road, use it and do not want to delete.

I have improved the numbering plan in an attempt to address the issue, but the upgrade still does not work.

Any ideas would be very appreciated.

Thank you

Kent

I encountered this problem on several upgrades and still fear to see a xxNP in use, how I work around this problem is by exporting the bosses of the route, Bulk Administration > export and select from there.

TAC will then need to get Root access to remove the xxNP (all) and then the upgrade will succeed.

Once the upgrade is complete and you have installed the xxNP that you need then to export models of road again from the Server v10.x because you will need to change the header.txt stored in the .tar file. Copy "CCM: master - 10.5.1.11901 - 1.i386' 10.x export item and replace the one exporting 9.x."

You can then import the 9.x CUCM CUCM 10.x road models

Hope this helps,

Richard

Equivalent of VCS to call the Transformation part mask in the CUCM

I'm changing the ID of group calling using a VCS. This can be done easily with a CUCM by creating a road model and using the 'mask of Transformation part of the call. Anyone know how to do this same thing via a VCS?

Hi Rachel,

This message can help - https://supportforums.cisco.com/thread/2128626 because he has a few suggestions.

Thank you

Guy

Profiles of router SIP (normalization of the number)

Hello, all!

I have some difficulty to come with the regex that is appropriate for the scenario. Does anyone have an idea how to write a translation for outgoing calls rule better?

Scenario: Extension 75001 dials 123-456-7890.

Path of the call: 75001--> CUCM-(SIP Trunk)-> CUBE---> SIP Trunk (PSTN)

Requirement: PSTN requires the full 10 digit phone number in the SIP header or it will reject the call.

Solution: Change the SIP header to translate the extension to a full 10-digit number.

Question: Phone number is 555-123-5001. 7 principals must be bare and only the last 4 digits should then be preceded the NPA - NXX.

For example...

75001 dials 123-456-7890. Appeal to CUCM. CUCM forwards calls to the CUBE. CUBE translates by the SIP header "[email protected] / * /' to"[email protected] / * /'.»

I believe that this can be accomplished with SIP diversion header manipulation, but I'm not totally understand the syntax behind it.

request INVITATION sip-head Diversion change "" "<>[email protected]"/ * / > ""

Thanks in advance for your time and help!

You need a header for reference because it is only used for call forwarding. what you need, is party-remote-ID or P - asserted-identity a, whereby your cucm is configured to use...

"change request INVITATION header sip remote-party-ID" ""

My.CallManager.Home

>"

Replace the NPA - XXX with the correct number, you need

Please note all useful posts

"opportunity is a haughty goddess who don't waste no time with those who are not prepared."

SX20 with CUCM

Hello

I have a doubt, if I have a sx20 recorded 11.5 CUCM if I call a video output or CMR is possible without the Expressay (C and E)

Concerning

Leonardo Santana

All calls will be routed by CUCM.

CUCM SIP only, so is the registration of endpoints to her, H323 will be disabled on the endpoint.

SX80 not save Server SIP config after reboot

We have a SX80 that when we go and configure the SIP to the CUCM server and save it until restart us. For example, define us the SIP server to ccm1.domain.edu then if the system is restarted, it 'default' return to ccm1 without domain information. We are on the last TC. Thanks in advance.

With CUCM, SIP server settings to do CUCM and not on the endpoint itself. Whenever endpoint restarts, it downloads the configuration of CUCM and everything you set on the endpoint will be overwritten. If you need to define a different IP of CUCM, you need change the CallManager group under the device pool and apply this device pool to the SX80 in CUCM.

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

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