FMIS 4.5 SIP support?

I see in the product comparison matrix that FMIS 4.5 allows SIP ports, but I'm not find any documentation or API related configuration

Is there a documentation available? Flash Media Gateway has been deployed in FMIS? We can manage incoming calls from SIP with FMIS now?

Thanks Nikhil.

In fact, I found the answer I needed. FMG has not been incorporated into FMS, it just comes with it. So, documentation isn't in the docs of FMS... installation of docs FMG with FMG.

Tags: Adobe Media Server

Similar Questions

Received an email from - find my line < support@fmi-apps.me >.

Hello

My iPhone got stolen and I had a series of email to find my iPhone asking me to connect to iCloud in the last hours persuade me to connect from 2 email accounts, click on the link and try to see the place, but I found a little email off the coast and decided to I opened a session on iCloud and found my phone lit up briefly. Is one of these e-mails are legitimate and if there will be an email to iCloud - what should be?

2 emails I got:

Find my APPLE FindMyIPhone <[email protected]> and <[email protected]> line

Thank you very much for your help in advance,

Fern

None of them are Apple.

EtherChannel does not support on 400 7600 - SIP

Hello

I'm a very big dilemma. I have 1 Gbps connection to my ISP. I'm under BGP to establish connectivity with the provider PE.

Just to inform you, I have redundant connections on the connection of 2 - Cisco 3750 (main & secondary link) next to service provider and there are 4 routers for 2 services (primary and secondary each for 2 services) now use on services - 1A applications increases and reaches 900 ~ 950 MB/s, is not on the impact of the services-2 I pushed the 2 services via secondary link traffic. I have attached ppt for some knowledge

Until a few weeks back the use was about 750 in the direction of entry and 700 Mbps in the exit direction of the WAN port on the service routers 1, but now the use of entry began to reach 900 ~ 950 MB/s in the direction of entry.

I have 400 7600 - SIP (void crack-5XGE-SPA) on the routers [services for Qos ] are related to the switches. I spoke with the service provider, and they said that they would consider more and while I thought to do the first aggregation of ports-2 between the provider of Service & 3750 switch device and an another 2-port aggregation between 3750 Switch & services-1 (Cisco7606) router.

There is no problem to create aggregation Service Provider betn ports device & switch 3750. It does not go between 3750 & services-1 switch (Cisco7606) router according to Cisco aggregation port document being not supported on 7600-SIP 400.

It takes time for the service provider to provide 10 port of concert. I WS-X 6548-GE-TX module Services-(Cisco7606) 1 router. So, I thought WS-X 6548-GE-TX router connection ports on the router with switch and create port channel between Them.But I wonder how would function QOS can be used.

So I need a Council if WS - X 6548-GE-TX ports support QOS with Etherchannel.

Your advice will be appreciated

R.

Valerie

The 3750 is a layer 3 switch. Could you change the design to allow links to layer 3 through him?

What get a 10Gbe for the 3750 and your 7600 module (s)?

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.

How to disable the boot into single user mode SIP?

Hi all

My MacBook Pro (bought in May 2011, out of warranty at this stage) is not able to start. I think that the graphics card is dead, I can boot into single user mode and run the Apple Hardware Test (no problem found). I found this acticle (https://people.cam.cornell.edu/~zc227/extras/early2011mbp_graphics.html), but I can't follow the steps to start the Mac to avoid the AMD card, such as the SIP is activated (cannot move road file extensions).

I can't boot into recovery mode, due to the inability to start the graphical mode. Any other way to disable SIP?

Thanks in advance.

Ana

In the developers page: security configuration is stored in NVRAM, rather than in the file system itself. As a result, this configuration applies to all Mac OS X installations across the entire machine and persists through installations of OS X than Protection of the integrity of support system.

To enable or disable the Protection of the integrity of the system, you must start on recovery OS and run the csrutil(1) command from the Terminal.

Integrity https://support.apple.com/en-us/HT204899 Protection System

This command can be run without root privileges and will tell you whether or not the SIP is enabled.

to find out whether the SIP is enabled or disabled, run

status of csrutil

to disable SIP:

copy & paste the following and then press ENTER:

csrutil disable

To activate the SIP

csrutil enable

After activation or deactivation of the Protection of the integrity of the system on a machine, a reboot is required.

There are some options here:

NetBoot and NetInstall NetRestore requirements in OS X El Capitan - Apple Support

Cannot turn on FileVault 2: "some disc formats do not support the recovery partition" version of the OS - 10.11.4

Hello

I went to turn on FileVault encryption for the first time, and when I click on restart (by FileVault guests) I got a error message that says "FileVault can not be enabled for the drive. Some disc formats do not support the required encryption recovery partition. To use encryption, reinstall this version of Mac OS x on a reformatted drive. »

I found an article from apple support for this problem with El Capitan who recommended first aid running in disk utility. I tried this but then received the same error message.

a little history:

-Mac OS 10.11.4 installed running on the brand/model ADATA SP 550 SSD

-This SSD is a 250 GB drive partitioned into two, with one of these partitions being my OS

-J' ran this BONE of a Bay external thunderbolt

-ssd is formatted as HFS journaling +.

I am really confused about this error message and couldn't find much online about it. I use a fairly recent OS install. I installed the operating system clean and made all my preferences and install so that I can encrypt and then back up this pristine OS environment in time machine so that I could always go back to it on the line in case where things are bunk too upward. I'd have to install yet another OS to clean and do all my long software installs and system preferences settings... SIP. I don't know if that makes a difference, but this version of the OS that I have been cloned by superduper to this SSD partition.

What is the brand of SSD that I? Might that matter? I appreciate all the comments. I'm really scratching my head...

Thank you very much

To enable FileVault, you need a recovery partition, which you don't have. You can create one by reinstalling OS X.

If you do not already have a current backup, please backup all data, and then reinstall the OS. * you don't need to erase the boot volume, and you won't need the backup unless something is wrong. If the system has been upgraded an earlier version of Mac OS X, you may need to the Apple ID and password to use.

There are ways to back up a computer that is not fully functional. Ask if you need advice.

If you installed the runtime Java distributed by Apple, and still need, you will need to reinstall it. It goes the same for Xcode. All other data will be preserved.

Related support article refers to OS X 10.11 ("El Capitan"), but the procedure is the same for OS X 10.7 ("Lion") and later versions.

[SPA3102] SIP recording every hour with the 401 error and directly 12 OK

Location: INET-ADSL modem in bridge mode-SPA3102.
Problem: not really, everything seems to work, can dial in and out.
But...
Because I'm curious, I have a logserver of installation and checked what happened every hour of registration.

What I see in the syslog hourly the 3102 made re - enroll by the SIP provider, but first I think 401 Unauthorized een 12 error and measured, I see an OK message.

Seems weird to me.

The same thing happens when I compose, firstly a 401 that OK.

Can someone explain why the first attempt gives an error and how to avoid this?

Some details of syslog:

I replaced a text in the syslog:

x.x.x.x is real My outside IP address.
yyyyyyyyy = my local phone number including the area code, such as 31201234567, 31 = NL, 20 = codeZone for Amsterdam
username = username by my SIP provider

message 1:
REGISTER SIP:SIP.poivy.com SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060; direction = z9hG4bK-2283ef7b
From: + 31yyyyyyyyy ; tag = c85fff819484d288o0
To: + 31yyyyyyyyy
Call ID: [email protected]

CSeq: 6104 REGISTRY
Max-Forwards: 70
Authorization: Digest username = "username", realm = "sip.poivy.com", nonce = "1663445546", uri = "sip:sip.poivy.com", algorithm = MD5 response = "c8c5b94c384559bb490b59b72be1c674"
Contact: + 31yyyyyyyyy ; expires = 3600
User-Agent: Linksys/SPA3102-3.3.6(GW)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFICATION OPTIONS, see
Support: x-sipura

Message 2:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP x.x.x.x:5060; direction = z9hG4bK-2283ef7b
From: + 31yyyyyyyyy ; tag = c85fff819484d288o0
To: + 31yyyyyyyyy
Contact: sip:x.x.x.x:5060
Call ID: [email protected]
CSeq: 6104 REGISTRY
Server: (very nice Sip Registrar/Proxy Server)
Allow: ACK, BYE, CANCEL, INVITE, REGISTER, OPTIONS, INFO, MESSAGE
WWW-Authenticate: Digest realm = "sip.poivy.com", nonce = "1667015687", algorithm = MD5
Content-Length: 0

Message 3:
REGISTER SIP:SIP.poivy.com SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060; direction = z9hG4bK-4d7052c
From: + 31yyyyyyyyy ; tag = c85fff819484d288o0
To: + 31yyyyyyyyy
Call ID: [email protected]

CSeq: 6105 REGISTRY
Max-Forwards: 70
Authorization: Digest username = "username", realm = "sip.poivy.com", nonce = "1667015687", uri = "sip:sip.poivy.com", algorithm = MD5 response = "46f2176652c0ad8d27f8d3ad1cf72c24"
Contact: + 31yyyyyyyyy ; expires = 3600
User-Agent: Linksys/SPA3102-3.3.6(GW)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFICATION OPTIONS, see
Support: x-sipura

Message 4:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP x.x.x.x:5060; direction = z9hG4bK-4d7052c
From: + 31yyyyyyyyy ; tag = c85fff819484d288o0
To: + 31yyyyyyyyy
Contact: + 31yyyyyyyyy ; expires = 3600
Call ID: [email protected]

CSeq: 6105 REGISTRY
Server: (very nice Sip Registrar/Proxy Server)
Allow: ACK, BYE, CANCEL, INVITE, REGISTER, OPTIONS, INFO, MESSAGE
Content-Length: 0

@hw: thank you for your tip and your right on the spot!

Never, ever, thought of this way of "logging" in a system and use delberatley an error response. With your tip, I thought that allows to read the RFC and found RFC 3665. This RFC describes the Protocol SIP basic call flow. And there he was, almost at the beginning of the real story on page 5! The protocol uses a command register which gives a message of 401 error back just to be sure to avoid security problems "man in the middle" (if I understand correctly). He present a challenge and you the answer to a totalizer new order including your answer on the challenge. Which will lead to an OK return the message.

To resume: nothing weird, it is as expected. My curiosity is frankly satisfied. Another day with what I've learned something. Thank you.

Needed support replacing WRT610N

I have more product Linksys here and tell everyone to use Linksys for years I have many items in cabnet do not use.

Well I got a $$$ WRT610N and totally had problems and it all boils down internal the software of linksys. "I run all these on my home 2xDMA WRT610N 2200" Linksys ", WGA11B"Linksys"old XBOX, XBOX 360, Wii, 2xLaptops, 3xDesktops a webserver.

The main problem is the Linksys SIP ACK block and my USB VOIP MagicJack linksys phone system completely dislike.

Cisco set the correct programming to linksys team crew and keep the product linksys with a good reputation and I will stop using DD - WRT Firmware on your system. As I think it is quite sad that the pirates free make the firmware that works for my support while manufacturing fail to support standard products or option in the GUI of linksys.

MBCMDR

I am at a total loss and love hearing entered Linksys Rep.

I've been a customer of Linksys with multi products and long term. Why did your Firmware not updated or fixed quickly? Why I ask this, because now I'm using dd - wrt firmware because linksys doesn't have support while DD - WRT offer support free firmware for a product they manufacture. They have a general forum like Linksys and you specify your problem. If you have any type of problem, even minor indicate you this in the Ant there they replcate it and write a fix. I know especially they relies to support extra, optiondans GUI to make things don't not necessary but some soft elements.

I don't have the problem of the WRT610N do it yourself resets on own for no reason even after he did with Linksys FW_WRT610N_1.00.02.10_US_20090121_code.bin, I have more problem ACK SIP with my VOIP. ETC.

Now under DD - WRT V24 - 11886_mega - WRT610N.bin Yes, they jump release numbers, but at least they offer firmware for my system and multi other routers.

SIP message with 488 here is not Acceptable

Hello, I'm having this problem. I have read the documentation and I look in this forum for similar problems.
Seems is a Codec problem. Service provider asks a g711alaw. However, I already change the codec, transcoder configured which is register with handler calls and everything seems fine. But outgoing calls always give this error. Can someone help me?

Here's the part of my configuration:

Send-call voice alert

convert-discpi-to-prog voice calls

voice, send rtp-received

voip phone service

list of approved IP addresses

IPv4 10.20.1.0 255.255.255.0

h323 connections allow h323

allow connections h323 to SIP

allow connections sip h323

allow sip to sip connections

service additional h450.12

Fax protocol t38 ls-redundancy version 0 0 hs-redundancy 0 help none

H323

No keepalive timeout h225

SIP

midcall-signalling passthru

SCCP local GigabitEthernet0/1

SCCP ccm 10.20.1.5 identifier 2 priority 2 version 7.0

SCCP ccm 10.20.1.6 identifier 1 priority 1 version 7.0

SCCP

!

SCCP ccm Group 1

associate the ccm 1 priority 1

the associated profile 1 registry COMPcode

the associated profile 2 registry COMP01-PSG

!

transcode dspfarm profile 1

Codec g711ulaw

Codec g711alaw

Codec g729ar8

Codec g729abr8

Codec g729r8

Codec g729br8

maximum sessions 14

associate the PCRS application

!

dspfarm profile 2 PSG

Codec g711ulaw

maximum sessions 120 software

associate the PCRS application

!

Dial-peer voice voip 50

Description # calls of CUCM to VG #.

incoming called number 9.T

DTMF-relay h245 alphanumeric

Codec g711alaw

No vad

!

Dial-peer voice 11 voip

Description * outgoing SIP Trunk call *.

outgoing SIP CALLS OUT translation-profile

preference 1

destination-model 9 t

Setup progress_ind allow 3

progress_ind enable progress 8

progress_ind connect enable 8

redirect ip2ip

session protocol sipv2

session target ipv4:88.XXX. XX.XXX

DTMF-relay rtp - nte cisco-rtp

Codec g711alaw

No vad

The Sip Trunk on the call manager uses a MRGL both of PSG and XCODE.

SCCP Admin State: to the TOP
Gateway local Interface: GigabitEthernet0/1
IPv4 address: 10.20.1.11
Port number: 2000
IP precedence: 5
List of Codec hidden user: no
Call Manager: 10.20.1.5, Port number: 2000
Priority: 2, Version: 7.0, identifier: 2
Call Manager: 10.20.1.6, Port number: 2000
Priority: 1, Version: 7.0, identifier: 1

Transcoding of Oper status: ACTIVE - reason Code: NO

Here are the two him debugs:

* 3 Dec 12:08:32.859: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
GUEST sip:[email protected]/ * /: SIP-5060/2.0
Via: SIP/2.0/TCP 10.20.1.6:5060; branch = z9hG4bK148c5e3b74ba77
From: <> [email protected]/ * */>;tag=1130165~6e3f6e1b-b59e-4604-be6a-d6295dc18cf6-45522220
To: <> [email protected]/ * / >
Date: Monday, December 3, 2012 12:02:40 GMT
Call ID: [email protected] / * /
Supported: timer, resource-priority, replaces
Min - SE: 1800
User-Agent: Cisco - CUCM8.6
Allow: PROMPT, OPTIONS, INFO, BYE, ACK, CANCEL, PRACK, UPDATE, VIEW, SUBSCRIBE, NOTIFY
CSeq: INVITE 101
Expires: 180
Allow-events: presence, kpml
Support: X-cisco-srtp-relief
Support: geolocation
Call-Info: ; method = "NOTIFY; Telephone-event = event; duration = 500 "
Cisco-Guid: 1422849408-0000065536-0000391810-0100733962
Session time-out: 1800
P has asserted-Identity: <> [email protected]/ * / >
Remote-Party-ID: <> [email protected]/ * / >; left = call; screen = yes; intimacy = off
Contact: <> [email protected]/ * /: 5060; transport = tcp >
Max-Forwards: 70
Content-Length: 0

* 3 Dec 12:08:32.863: //-1/54CEF5800005/CCAPI/cc_api_display_ie_subfields:
cc_api_call_setup_ind_common:
Cisco-username = 02036666666
-ccCallInfo IE subfields-
Cisco-ani = 02036666666
Cisco-anitype = 0
Cisco-aniplan = 0
Cisco-anipi = 0
Cisco-anisi = 1
dest = 907718005555
Cisco-desttype = 0
Cisco-destplan = 0
Cisco-ISDS = FFFFFFFF
Cisco-rdn =
Cisco-rdntype = 0
Cisco-rdnplan = 0
Cisco-rdnpi =-1
Cisco-rdnsi =-1
Cisco-redirectreason = - 1 fwd_final_type = 0
final_redirectNumber =
hunt_group_timeout = 0

* 3 Dec 12:08:32.863: //-1/54CEF5800005/CCAPI/cc_api_call_setup_ind_common:
Interface = 0 x 31196458, call Info)
Number = 02036666666, (Calling Name =) (TON = unknown, NPI = unknown, screening = User, spent, presentation = allowed).
Called number = 907718005555 (TON = unknown, NPI = unknown).
The appeal translated = FALSE, Subscriber Type Str = unknown, FinalDestinationFlag = TRUE,
Incoming dial-peer = 50, progress Indication = NULL (0), Calling THE Present = TRUE,
Road Trkgrp Label source, label Trkgrp road target = CLID Transparent = FALSE), call Id = 21355
* 3 Dec 12:08:32.863: //-1/54CEF5800005/CCAPI/ccCheckClipClir:
In: Component number = 02036666666 (TON = unknown, NPI = unknown, screening = User, spent, presentation = allowed)
* 3 Dec 12:08:32.863: //-1/54CEF5800005/CCAPI/ccCheckClipClir:
Departures: Component number = 02036666666 (TONE = unknown, NPI = unknown = User, spent, screening presentation allowed =)
* 3 Dec 12:08:32.863: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

* 3 Dec 12:08:32.863: cc_get_feature_vsa success of malloc
* 3 Dec 12:08:32.863: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

* 12:08:32.863 3 dec: cc_get_feature_vsa number is 1
* 3 Dec 12:08:32.863: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

* 12:08:32.863 3 dec: FEATURE_VSA attributes are: feature_name:0, feature_time:832856160, feature_id:21306
* 12:08:32.863 3 dec: / / 21355, 54CEF5800005, CCAPI, cc_api_call_setup_ind_common:
Set up the event sent;
Call Info (number = 02036666666 (TON = unknown, NPI = unknown, screening = User, spent, presentation = allowed),)
Called number = 907718005555 (TON = unknown, NPI = unknown))
* 12:08:32.863 3 dec: / / 21355, 54CEF5800005, CCAPI, cc_process_call_setup_ind:
Event = 0x2AFCEB88
* 3 Dec 12:08:32.863: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
Try again with the demoted called number 907718005555
* 12:08:32.863 3 dec: / / 21355, 54CEF5800005, CCAPI, ccCallSetContext:
Context = 0x32D52B8C
* 12:08:32.863 3 dec: / / 21355, 54CEF5800005, CCAPI, cc_process_call_setup_ind:
> Handed CCAPI cid 21355 with tag 50 app '_ManagedAppProcess_Default '.
* 12:08:32.867 3 dec: / / 21355, 54CEF5800005, CCAPI, ccCallProceeding:
Progress Indication = NULL (0)
* 12:08:32.867 3 dec: / / 21355, 54CEF5800005, CCAPI, ccCallSetupRequest:
Destination =, Calling IE date = TRUE, Mode = 0.
Leaving Dial-peer = 11, Params = 0x2B62716C, Indication of progress = FROM SIDE IS NO ISDN (3)
* 12:08:32.867 3 dec: / / 21355, 54CEF5800005, CCAPI, ccCheckClipClir:
In: Component number = 02036666666 (TON = unknown, NPI = unknown, screening = User, spent, presentation = allowed)
* 12:08:32.867 3 dec: / / 21355, 54CEF5800005, CCAPI, ccCheckClipClir:
Departures: Component number = 02036666666 (TONE = unknown, NPI = unknown = User, spent, screening presentation allowed =)
* 12:08:32.867 3 dec: / / 21355, 54CEF5800005, CCAPI, ccCallSetupRequest:
The destination model = 9 t., called number = 07718009863, band numbers = FALSE
* 12:08:32.867 3 dec: / / 21355, 54CEF5800005, CCAPI, ccCallSetupRequest:
Number = 02036666666 (TON = unknown, NPI = unknown, screening = User, spent, presentation = allowed).
Called number = 07718009863 (TON = unknown, NPI = unknown).
Redirect = number, display of information is
Account number 02036666666, Destination = final = TRUE flag,.
GUID = 54CEF580-0001-0000-0005-FA820601140A, outbound Dial-peer = 11
* 12:08:32.867 3 dec: / / 21355, 54CEF5800005, CCAPI, cc_api_display_ie_subfields:
ccCallSetupRequest:
Cisco-username = 02036666666
-ccCallInfo IE subfields-
Cisco-ani = 02036666666
Cisco-anitype = 0
Cisco-aniplan = 0
Cisco-anipi = 0
Cisco-anisi = 1
dest = 07718009863
Cisco-desttype = 0
Cisco-destplan = 0
Cisco-ISDS = FFFFFFFF
Cisco-rdn =
Cisco-rdntype = 0
Cisco-rdnplan = 0
Cisco-rdnpi =-1
Cisco-rdnsi =-1
Cisco-redirectreason = - 1 fwd_final_type = 0
final_redirectNumber =
hunt_group_timeout = 0

* 12:08:32.867 3 dec: / / 21355, 54CEF5800005, CCAPI, ccIFCallSetupRequestPrivate:
Interface = 0 x 31196458, Type of Interface = 3, = Destination, the Mode = 0x0,
Call Params (number = 02036666666, (Calling Name =) (= unknown, NPI = unknown, screening = User, TON spent, presentation = allowed),)
Called number = 07718009863 (TON = unknown, NPI = unknown), the appeal translated = FALSE,
Subscriber Type Str = unknown, FinalDestinationFlag = TRUE, outgoing Dial-peer = 11, Call On County = FALSE,
Trkgrp road Label source, label of road target Trkgrp =, tg_label_flag = 0, Application Call Id =)
* 3 Dec 12:08:32.867: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

* 3 Dec 12:08:32.867: cc_get_feature_vsa success of malloc
* 3 Dec 12:08:32.867: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

* 12:08:32.867 3 dec: cc_get_feature_vsa number is 2
* 3 Dec 12:08:32.867: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

* 12:08:32.867 3 dec: FEATURE_VSA attributes are: feature_name:0, feature_time:832857952, feature_id:21307
* 12:08:32.867 3 dec: / / 21356, 54CEF5800005, CCAPI, ccIFCallSetupRequestPrivate:
Application for facility call SPI's success; Interface type = 3, FlowMode = 1
* 12:08:32.867 3 dec: / / 21356, 54CEF5800005, CCAPI, ccCallSetContext:
Context = 0x2B62711C
* 12:08:32.867 3 dec: / / 21355, 54CEF5800005, CCAPI, ccSaveDialpeerTag:
Outbound Dial-peer = 11
* 12:08:32.867 3 dec: / / 21356, 54CEF5800005, CCAPI, cc_api_call_proceeding:
Interface = 0 x 31196458, Indication = NULL (0) progress
* 12:08:32.871 3 dec: / / 21356/54CEF5800005/SIP/Msg/ccsipDisplayMsg:
Envoy:
GUEST sip:[email protected] / * /.XXX:5060 SIP/2.0
Via: SIP/2.0/UDP 194.168.146.148:5060; branch = z9hG4bK4F1C77
Remote-Party-ID: <> [email protected]/ * / >; left = call; screen = yes; intimacy = off
From: <> [email protected]/ * / >; tag = E448A58-A24
To: <> [email protected] / * /.XXX >
Date: Monday, December 3, 2012 12:08:32 GMT
Call ID: [email protected] / * /
Supported: timer, resource-priority, replaces, sdp-anat
Min - SE: 1800
Cisco-Guid: 1422849408-0000065536-0000391810-0100733962
User-Agent: Cisco-SIPGateway/IOS - 12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, VIEW, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: INVITE 101
Time stamp: 1354536512
Contact: <> [email protected]/ * /: 5060 >
Expires: 180
Allow-events: telephone-event
Max-Forwards: 69
Session time-out: 1800
Content-Length: 0

* 12:08:32.871 3 dec: / / 21355/54CEF5800005/SIP/Msg/ccsipDisplayMsg:
Envoy:
SIP/2.0 100 trying
Via: SIP/2.0/TCP 10.20.1.6:5060; branch = z9hG4bK148c5e3b74ba77
From: <> [email protected]/ * */>;tag=1130165~6e3f6e1b-b59e-4604-be6a-d6295dc18cf6-45522220
To: <> [email protected]/ * / >
Date: Monday, December 3, 2012 12:08:32 GMT
Call ID: [email protected] / * /
CSeq: INVITE 101
Allow-events: telephone-event
Server: Cisco-SIPGateway/IOS - 12.x
Content-Length: 0

* 12:08:32.899 3 dec: / / 21356/54CEF5800005/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 trying
Via: SIP/2.0/UDP 194.168.146.148:5060; received = 194.168.146.148; branch = z9hG4bK4F1C77
From: <> [email protected]/ * / >; tag = E488A58-A24
To: <> [email protected] / * /.XXX >
Call ID: [email protected] / * /
CSeq: INVITE 101
Time stamp: 1354536512
Content-Length: 0

* 12:08:32.995 3 dec: / / 21356/54CEF5800005/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 488 here is not Acceptable
Via: SIP/2.0/UDP 194.168.146.148:5060; received = 194.168.146.148; branch = z9hG4bK4F1C77
To: <> [email protected] / * /.XXX >; tag = 3563525093-836070
From: <> [email protected]/ * / >; tag = E488A58-A24
Call ID: [email protected] / * /
CSeq: INVITE 101
Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Contact: <> [email protected] / * /.XXX:5060 >
Reason: Q.850; cause = 65
Content-Length: 0

* 12:08:32.995 3 dec: / / 21356, 54CEF5800005, CCAPI, cc_api_call_disconnected:
Value = 127, Interface = 0 x 31196458, Call Id = 21356
* 12:08:32.995 3 dec: / / 21356, 54CEF5800005, CCAPI, cc_api_call_disconnected:
Call the entry (Responsed = TRUE, Cause value = 127, Retry Count = 0)
* 12:08:32.995 3 dec: / / 21355, 54CEF5800005, CCAPI, ccCallReleaseResources:
free xcoding reserved resource.
* 12:08:32.995 3 dec: / / 21356, 54CEF5800005, CCAPI, ccCallSetAAA_Accounting:
= 0, Id = 21356 Call Accounting
* 12:08:32.995 3 dec: / / 21356, 54CEF5800005, CCAPI, ccCallDisconnect:
Value = 127, Tag = 0x0, entry calls (previous disconnection Cause = 0, remove the Cause = 127)
* 12:08:32.995 3 dec: / / 21356, 54CEF5800005, CCAPI, ccCallDisconnect:
Value = 127, entered calls (Responsed = TRUE, Cause value = 127)
* 12:08:32.999 3 dec: / / 21356, 54CEF5800005, CCAPI, cc_api_call_disconnect_done:
Available = 0, Interface = 0 x 31196458, Tag = 0 x 0, Call Id 21356 =.
Call the entry (disconnect Cause = 127, class voice Cause Code = 0, retry count = 0)
* 12:08:32.999 3 dec: / / 21356, 54CEF5800005, CCAPI, cc_api_call_disconnect_done:
Call interrupted event sent
* 3 Dec 12:08:32.999: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

* 3 Dec 12:08:32.999: cc_free_feature_vsa release A 31, 46758
* 3 Dec 12:08:32.999: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

* 3 Dec 12:08:32.999: free vsacount is 1
* 12:08:32.999 3 dec: / / 21355, 54CEF5800005, CCAPI, ccCallDisconnect:
Value = 127, Tag = 0x0, entry calls (previous disconnection Cause = 0, remove the Cause = 0)
* 12:08:32.999 3 dec: / / 21355, 54CEF5800005, CCAPI, ccCallDisconnect:
Value = 127, entered calls (Responsed = TRUE, Cause value = 127)
* 12:08:32.999 3 dec: / / 21355/54CEF5800005/SIP/Msg/ccsipDisplayMsg:
Envoy:
SIP/2.0 500 Internal Server Error
Via: SIP/2.0/TCP 10.20.1.6:5060; branch = z9hG4bK148c5e3b74ba77
From: <> [email protected]/ * */>;tag=1130165~6e3f6e1b-b59e-4604-be6a-d6295dc18cf6-45522220
To: <> [email protected]/ * / >; tag = E488ADC-CEC
Date: Monday, December 3, 2012 12:08:32 GMT
Call ID: [email protected] / * /
CSeq: INVITE 101
Allow-events: telephone-event
Server: Cisco-SIPGateway/IOS - 12.x
Reason: Q.850; cause = 127
Content-Length: 0

* 3 Dec 12:08:33.035: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP ACK:[email protected]/ * /: SIP-5060/2.0
Via: SIP/2.0/TCP 10.20.1.6:5060; branch = z9hG4bK148c5e3b74ba77
From: <> [email protected]/ * */>;tag=1130165~6e3f6e1b-b59e-4604-be6a-d6295dc18cf6-45522220
To: <> [email protected]/ * / >; tag = E488ADC-CEC
Date: Monday, December 3, 2012 12:02:40 GMT
Call ID: [email protected] / * /
Max-Forwards: 70
CSeq: 101 ACK
Allow-events: presence, kpml
Content-Length: 0

* 12:08:33.039 3 dec: / / 21355, 54CEF5800005, CCAPI, cc_api_call_disconnect_done:
Available = 0, Interface = 0 x 31196458, Tag = 0 x 0, Call Id = 21355.
Call the entry (disconnect Cause = 127, class voice Cause Code = 0, retry count = 0)
* 12:08:33.039 3 dec: / / 21355, 54CEF5800005, CCAPI, cc_api_call_disconnect_done:
Call interrupted event sent
* 3 Dec 12:08:33.039: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

* 3 Dec 12:08:33.039: cc_free_feature_vsa release A 31, 46058
* 3 Dec 12:08:33.039: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

* 3 Dec 12:08:33.039: free vsacount is 0
* 3 Dec 12:08:33.039: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Envoy:
SIP ACK:[email protected] / * /.XXX:5060 SIP/2.0
Via: SIP/2.0/UDP 194.168.146.148:5060; branch = z9hG4bK4F1C77
From: <> [email protected]/ * / >; tag = E488A58-A24
To: <> [email protected] / * /.XXX >; tag = 3563525093-836070
Date: Monday, December 3, 2012 12:08:32 GMT
Call ID: [email protected] / * /
Max-Forwards: 70
CSeq: 101 ACK
Allow-events: telephone-event
Content-Length: 0

Thank you very much

Chris

Hi Chris,

It seems that your CUBE config is correct. The CUCM sends a PROMPT to provide at the beginning now. This means that you have enabled the PSG on the SIP trunk. The codec sent in the INVITATION is G711ulaw. In order to change this, go to the SIP trunk and search "PSG preferred from Codec" and replace it by default of G711ulaw, G711alaw. If all goes well, which does the trick for you.

HTH.

Kind regards

Stefano.

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.

What is SIP/2.0 481 call Leg/Transaction does not means?

Hello

When call cme of sip trunk and hang up the phone call calling debug ccsip show:

Jun 2 08:52:40.541: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor: invite dialog box control
Jun 2 08:52:40.541: //-1/xxxxxxxxxxxx/SIP/Error/sipSPISipIncomingMsg: invalid (STATE_IDLE) method: CANCEL
Jun 2 08:52:40.545: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPISendErrorRespNoCCB: send error no. RCC response to the transport layer
Jun 2 08:52:40.545: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPITransportSendMessage: msg = 0x88D771A0, addr = 172.19.1.78, port = 5060, sentBy_port = 5060, is_req = 0, transport = 1, switch = 0, call = 0 x 00000000
Jun 2 08:52:40.545: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerGetConnectionIfExists: support connection for addr = 172.19.1.78 not found
CME(config-SIP-UA) #.
Jun 2 08:52:40.545: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: display send msg = 0x88D771A0, addr = 172.19.1.78, port = 5060, connId = 0 for UDP
Jun 2 08:52:40.553: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Envoy:
SIP/2.0 481 call Leg/Transaction does not exist
Via: SIP/2.0/UDP 77.72.169.134:5060; branch = z9hG4bK49b6543463d947caa1e4b81bf0267d24; received = 172.19.1.78
From: <> [email protected] / * /: 5060 >; tag = 4e0113ac4c050c86452e9
To: [email protected] / * / add: 5060 >
Call ID: 4c4d940ab840442cb123a51a893cf5f5
CSeq: 3 CANCEL
Content-Length: 0

Hello

This means that the bridge is not no matter what record in the appeal. It was not any call or that it has removed the information.

Looking at the extract, this particular msg was sent in response to cancel event. Without seeing the debug, I'm assuming that GW has previously received cancel and delete the call and send response. But on the side of appellant sent another msg so cancel gw gives above msg

Hope this helps

Thank you

-abu

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.

Writing own SIP Stack on BlackBerry

Hello

I am a newbie to BlackBerry and I want to develop a VoIP on BlackBerry client.

I searched the forum and found that there is no support for SIP API.

So the first step is to write my own SIP stack. I have several queries in my mind:
1. Can I develop my own stack SIP on BlackBerry?
2. What support can I get from the platform?
3. I also analyzed that BlackBerry does not support codecs G.729/G.711 or it allows to install new ones. How do I get these codecs?
4. Is there a RTP API or I need to write my own RTP stack?
5. Are there restrictions on platform?
6. What that all obstacles will face during this evolution?
Anyone can guide me please on them.

Thank you very much

mansriv

I'd be happy it and perhaps as it's possible, but it would take a lot of work, not a society that backup I would not take it.
the blackberry platform is not very accessible to a low level. I don't know where to start without JSR180.

RIM and Skype collaborate on a client, perhaps is that it shows a change in the policy of rims and they will support SIP?

http://about.Skype.com/2009/03/skype_coming_to_blackberry_sma.html

No support for URI Scheme Expressway C

Nice day!!

A question that I have the next deployment

Cluster Version 11 - Expressway CUCM C - Expressway E - Webex

I realized the Collaboration meeting room guide and other deployments work but in this crackling SIP does not work.

Areas - transformations - research rules - is fine but the Expressway C restore the call with the following message " 416 Expressway of the URI scheme does not support C ', I try to reconfigure everything except always show the same error." "

Error.PNG

No idea for the error? Thanks for your time!

Most of these rules is written incorrectly and a lot will not or what you want them to do. Have you tested the rules when you created to see if they would give you the required result with the related task to "Check if a pattern match or transform has the expected result" at the bottom of the web search page of rules?

Your first snapshot DNS rule for CMR Cloud Services there nothing in parentheses in the pattern string, so the result of the replacement string will be invalid.

The other, you're having a string of models with only two things in brackets and then by creating a result with three elements that will not work.

You probably can replace both your rules with a single rule:
```
Pattern string: (.+)@(.+)\.webex\.comPattern behaviour: ReplaceReplace string: \1@\2.webex.com
```
Wayne - don't forget the frequency responses and mark your question as answered as appropriate.

FMIS 4.5 SIP support?

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Thank you very much for your help in advance,

Error.PNG

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