SIP trunk from ITSP calling issues

[Risat]

Hi, I testing a new SIP connection from Service Provider, my connection will be ITSP-SIP-VGRouter-SIP-CUCM, The issue I am facing is I'm unable to make call outside to ITSP from my Cisco Phone, however I can receive calls from ITSP. Please find the attached show run and debug from the router. Also please note that I'm unable to give the command 'mode border-element' under 'voice service voip', my IOS on the router is c3845-adventerprisek9_ivs-mz.151-4.M10.bin.

[Mohammed Khan]

can you enable following debugs

- debug ccsip message

- debug voice ccapi inout

and make a sample call.

Regards,

Mohammed Noor

[Risat]

Hi MOhammed,

I have attached the debug logs already, please find the call details below

Calling number - 7067333 (Cisco IP Phone)

Called number - 026589000

[Mohammed Khan]

Aug 17 08:37:24.871: //5784/BDBB6F800002/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.1.1.7:5060;branch=z9hG4bK5032256
From: <sip:7067333@10.1.1.7>;tag=82474C8-2617
To: <sip:026589000@10.1.1.6>;tag=y6sqtpp7
Call-ID: 9504FA90-638C11E6-9750F86F-EF5B01D8@10.1.1.7
CSeq: 101 INVITE
Timestamp: 1471423012
Warning: 399 SoftX3000 "SS010000F00156L00373[0000] Cancel received from network"
Content-Length: 0

Telco issue.Contact your service provider.

[Mohammed al Baqari]

I saw that the call is terminated by your operator.

Received: 
SIP/2.0 487 Request TerminatedVia: SIP/2.0/UDP 10.1.1.7:5060;branch=z9hG4bK5032256From: <sip:7067333@10.1.1.7>;tag=82474C8-2617To: <sip:026589000@10.1.1.6>;tag=y6sqtpp7Call-ID: 9504FA90-638C11E6-9750F86F-EF5B01D8@10.1.1.7CSeq: 101 INVITETimestamp: 1471423012Warning: 399 SoftX3000 "SS010000F00156L00373[0000] Cancel received from network"Content-Length: 0

You are sending G711ulaw/alaw to your operator. Can you confirm if they are happy with it. Also, open a ticket with them to find out why they are rejecting the call.

[Risat]

Thanks for the reply, do i need to enable 'mode border element' for this scenario and why I am not able to put this command even though the IOS supports (c3845-adventerprisek9_ivs-mz.151-4.M10.bin), also please note that I have a transcoder configured and registered with the CUCM, can this help to rectify our issue here ??

[Mohammed al Baqari]

This command is supported in universal packets. You won't be able to put it in your image.

Don't worry about it. Its for licensing purpose but won't impact the functionality

[Risat]

When I configure SIP Xlite on my laptop and connect directly to the ITSP SIP interface, the xlite works fine inbound and outbound without any issue, TELCO technician just test in this way and says that there is no issue from their side, we have to correct this from our side :-(

[Diego Chvz]

Ris,

Enable debug ccsip message and debug voice ccapi inout and make two test, one using the SIP Xlite and the other one using the Cisco IP Phone so we can compare how devices are negotiating.

[Risat · Accepted solution]

Iam connecting xlite directly with the SIP connection from ITSP, which is not touching the VG, so i can't do the debug with xlite, I am attaching another set of debug logs with Cisco phone, inbound from ITSP which is working and also outbound to ITSP which is not working, also attaching the VG-backup, please check.

[Diego Chvz]

Ris,

What I can see from your output is that definitely the session is terminating from the side of your ITSP:

SIP/2.0 487 Request TerminatedVia: SIP/2.0/UDP 10.1.1.7:5060;branch=z9hG4bK7F311C9From: <sip:7067333@10.1.1.7>;tag=D552984-1959To: <sip:026589300@10.1.1.6>;tag=s837krtpCall-ID: 55044E52-645711E6-A60CF86F-EF5B01D8@10.1.1.7CSeq: 101 INVITETimestamp: 1471510093Warning: 399 SoftX3000 "SS010000F00156L00373[0000] Cancel received from network"Content-Length: 0
Aug 18 08:48:45.316: //9454/7E0D8A000002/CCAPI/cc_api_call_disconnected:

In addition we can see that unlike the other call working OK, here there are some t38 media exchanges in the ITSP side.

SIP/2.0 200 OKVia: SIP/2.0/UDP 10.1.1.6:5060;branch=z9hG4bKqnf6lv004gcgj4rmk2t1From: <sip:ping@10.1.1.6>;tag=49ce0a8af6b286d5217997b5e4f3dfdc0002vn1To: sip:ping@27067333.serviceprovider;tag=D55C1DC-7A6Date: Thu, 18 Aug 2016 08:48:52 GMTCall-ID: 3d187f50fa1379a90b6069a3785941890002vn1@10.1.1.6Server: Cisco-SIPGateway/IOS-12.xCSeq: 61807 OPTIONSAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventAccept: application/sdpSupported: 100rel,timer,resource-priority,replaces,sdp-anatContent-Type: application/sdpContent-Length: 450v=0o=CiscoSystemsSIP-GW-UserAgent 2035 6339 IN IP4 192.168.75.220s=SIP Callc=IN IP4 10.1.1.7t=0 0m=audio 0 RTP/AVP 18 0 8 9 4 2 15 3c=IN IP4 10.1.1.7m=image 0 udptl t38c=IN IP4 10.1.1.7a=T38FaxVersion:0a=T38MaxBitRate:9600a=T38FaxFillBitRemoval:0a=T38FaxTranscodingMMR:0a=T38FaxTranscodingJBIG:0a=T38FaxRateManagement:transferredTCFa=T38FaxMaxBuffer:200a=T38FaxMaxDatagram:320a=T38FaxUdpEC:t38UDPRedundancy

Is this a new setup? Is faxing working OK?

To determine what exactly is failing we really need take a look at the logs from the ITSP side since it seems like some capabilities that you are sending are not of the liking of them.

Please test disabling the line:

fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none

from the voice service voip and configure it at dialpeer level (create a new dialpeer for faxing purposes) and leaving your dial-peer 104 as it is.

Enable debug ccapi inout and debug ccsip messages and attach the new debug output. 

[Risat]

This is a new setup, there is no fax here at the moment, so i have removed the fax T38 command now. please find the attached debugs taken after removing the fax command.

[Diego Chvz]

OK, Ris,

The same error but we are narrowing down the possible causes. Discarding media issues, I suspect that you are sending the incorrect user-id portion in identify headers so apply your voice-class sip profile this way:

First turn on the inbound SIP Profile feature as shown below:

voice service voip
 sip
  sip-profiles inbound

Apply your voice-class sip profile 100 to the incoming dial-peer voice 104 as shown below:

dial-peer voice 104 voip
 description ##  CUCM Incoming Calls ##
 voice-class sip profiles 100 inbound

It will look like this:

dial-peer voice 104 voip
 description ##  CUCM Incoming Calls ##
 preference 1
 session protocol sipv2
 session target ipv4:192.168.75.202
 session transport udp
 voice-class sip profiles 100 inbound
 incoming called-number 8T
 voice-class codec 1  
 dtmf-relay rtp-nte
 no vad

Now make a test and please perfom the same procedure for retrieving logs and attach them. If it doesn't work then you will need to engage with your ITSP to ask for the correct user-id portion and alternatively ask for them to verify why they are disconnecting your call.