SIP Trunk Messages SIP/2.0 403 Forbidden

cristian.munoz · ‎03-04-2020

Hi All

I trying to call from a SIP phone to PSTN and the debug ccsip (in the CME Branch Office) show the message :

Received:SIP/2.0 403 Forbidden

The scenario is:

SIP Phone -->CME Branch Office ---SIP Trunk-->CME HQ--SIP Trunk-->PSTN

- The calls to SIP Phones of CME HQ works

- The calls from SIP Phones of CME HQ to PSTN works

I have doubt with the dial-peer to PSTN in the CME Branch Office, point to IP of CME HQ:

CME Branch Office

dial-peer voice 51 voip
description *** OUTGOING LOCAL SIP TRUNK PSTN ***
preference 1
destination-pattern 90[2-8]........
session protocol sipv2
session target ipv4:10.97.1.2
voice-class codec 2 ----> G711ulaw and G711alaw
dtmf-relay rtp-nte
no vad
supplementary-service sip handle-replaces

CME HQ

dial-peer voice 51 voip
description *** OUTGOING LOCAL SIP TRUNK PSTN***
translation-profile outgoing prefixout_SIP_TRUNK --->This only delete the number 90
preference 1
destination-pattern 90[2-8]........
session protocol sipv2
session target ipv4:10.232.178.97
voice-class codec 2 ----> G711ulaw and G711alaw
dtmf-relay rtp-nte
no vad
supplementary-service sip handle-replaces

Please, see the attach. the debug ccsip message

TIA

Cristian

TONY SMITH · ‎03-05-2020

At a guess the carrier may be unhappy with your calling number which is presented just as "From: <sip:371@10.97.9.2>". Many carriers need to see a valid number, matching the incoming number range. Others need to see trunk ID or pilot number somewhere in another header. So it's really just a guess without knowing anything about your carrier. Have you asked them?

While awaiting their response, compare your branch office Invite from one from a working call at head office, assuming the carrier is the same of course.

cristian.munoz · ‎03-05-2020

Dear Tony

I foced the id number for the calls from Branch Office to HQ with the prefix supported for PSTN. And now I compared the debug of call from Branch Office to PSTN fail versus call from HQ to PSTN success.

I see the call fail because only offer a=rtpmap:0 PCMU/8000 instead ta call success offer a=rtpmap:0 PCMU/8000 and
a=rtpmap:8 PCMA/8000

Please see attach of the debugs of CME HQ

TIA
Cristian

TONY SMITH · ‎03-06-2020

I agree on the codec differences. Your working call offers both G711u and G711a, and the carrier chooses G711a. Your non-working call offers only G711u, so there is a chance that the carrier will not accept this.

However I'm not convinced that's the only problem, for a codec mismatch I'd expect to see 488 "No Acceptable Media" or similar. So I suggest fix the codec issue first in any case, but also continue to chase the carrier for their response.

We are only guessing until we know exactly what the carrier needs from you.

cristian.munoz · ‎03-06-2020

Dar Tony

The carrier need G711alaw. My question is how to the CME Branch Office to offer both codecs (G771alaw and G711ulaw) instead of offering G711ulaw only.

Here an example of PSTN Call answered:

Mar 6 09:50:09.520: //121451/B0948DD38059/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3F8A71F0
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 263
Called Number : 226928000
Source IP Address (Sig ): 192.168.100.200
Destn SIP Req Addr:Port : 10.232.178.97:5060
Destn SIP Resp Addr:Port : 10.232.178.97:5060
Destination Name : 10.232.178.97

Mar 6 09:50:09.520: //121451/B0948DD38059/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g711alaw
Negotiated Codec Bytes : 160
Nego. Codec payload : 8 (tx), 8 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 192.168.100.200
Source IP Port (Media): 17220
Destn IP Address (Media): 10.232.178.97
Destn IP Port (Media): 32940
Orig Destn IP Address:Port (Media): [ - ]:0

TIA
Cristian

TONY SMITH · ‎03-06-2020

@cristian.munoz wrote:
Dar Tony
The carrier need G711alaw. My question is how to the CME Branch Office to offer both codecs (G771alaw and G711ulaw) instead of offering G711ulaw only.

Without seeing the configurations this is also guesswork, but at a guess your CME that offers both may use a codec class, the CME offering only G711u may have that hard coded on the dial peer. If you wanted to share your configuration(s) we could be more specific.

I still don't think codec mismatch is the most likely cause of the 403 errors.

What does your carrier say?

Roger Kallberg · ‎03-07-2020

@TONY SMITH wrote:
@cristian.munoz wrote:
Dar Tony
The carrier need G711alaw. My question is how to the CME Branch Office to offer both codecs (G771alaw and G711ulaw) instead of offering G711ulaw only.

Without seeing the configurations this is also guesswork, but at a guess your CME that offers both may use a codec class, the CME offering only G711u may have that hard coded on the dial peer. If you wanted to share your configuration(s) we could be more specific.
I still don't think codec mismatch is the most likely cause of the 403 errors.
What does your carrier say?

I would agree with @TONY SMITH on that the 403 error is likely not caused by the codec. What does your telco say about this? Again as Tony previous suggested the likely cause would be the calling number sent for the call. I would recommend you to check this for a working call and compare it with a failed.