SIP cause=47 reason= 16 - Q.850 - SIP to SIP

[dsobrinho]

Hey guys,

I have a problems that have been stressed my brain a coulple of days.

When I tried to call using my ipcommunicator based on sip with destination for my mobile mail box, to check my messages, I have been received a disconnect tone. After done some debugs, I could not identify root of the problems. I can see only a message of disconnection with cause 47 (resource unavailable). I think that is codec problem, but I am not sure.

Does anybody knows if this debugs is enough to dectect the problem?

Following the debugs  attached

[dksingh · Accepted solution]

Without the call flow its hard to pinpoint the issue. But based on debugs attached and

cause of 47(no resource) it appears that the codec being negotiated is g729 on one leg

and g729b on other so a need for transcoder arises which perhaps is not available.

There are ways to interop between various flavors of g729 codecs without needing a transcoder

as such but please clarify what is the call flow we are dealing with first.

[dsobrinho]

Hi my friend,

Thank you for your help and sorry for cannot apply all information and description.

I will try summarized here. When I tried to call for mobile voicemail, the call can not completed and received a disconnect.

My ISP is SiP provider and does not work with codec g711ulaw. Only g729 (families) and g711alaw. When SIP gateway from my service provider contacted the Claro (mobile operator gateway) my SIP provided send me back a re-invite with diferent codec that does stablish the negociate with my voice gateway.

I have already tried to do everything and the only solution was remove the voice-class codec from the dial-peer.

voice class codec 1
codec preference 1 g729r8 bytes 160
codec preference 2 g729br8 bytes 160
codec preference 3 g711alaw

!

!

dial-peer voice 99999 voip
description Outgoing dial-peer to GVT
translation-profile outgoing gvt
preference 1
destination-pattern 025..[2-9].......
no modem passthrough
voice-class codec 1            <------------------- Removed
  session protocol sipv2
session target ipv4:10.210.65.16
dtmf-relay rtp-nte
fax-relay ecm disable
no fax-relay sg3-to-g3
fax protocol pass-through g711alaw

After verified in depth I noticed that the codec that has been trying negociate with my gateway was g711alaw. It was not receiving re-invite from my sip ISP. So, the solution was create a new dial-peer only for services like fax machines.

On the Peer 99999 I only remove the voice-class codec 1 and after that all calls made for mobile mailbox (Claro operator) backs to works properly.

[tobin hawkshaw]

just as a side note, although Cisco tells us it works I have never had much luck with voice-class codecs on anything SIP, I would never use it, its much better to hard code them.

[srsivara]

Hey Tobin,

If you are using CUBE for the SIP calls, here's the related enhancement defect -

CSCta89788    Voice Class Codec code commit

Integrated releases - 15.1(01.06)T, 15.1(01.02)PI13b

So, voice class codec is officially supported only from the releases mentioned above.

- Sriram

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