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the legs separately as if they were not related to the same call. So the
ingress leg negotiates ulaw, and despite it knowing that the peer also
supports g729 fails the call since it's already decided on ulaw and the
egress leg only accepts g729.

If this is design intent I'm wondering if there is demand enough to justify
a feature request?

Any advice on how I can work around this issue?

Thanks,

-Ryan

On Tue, Aug 2, 2011 at 3:51 PM, Ryan McGuire <rdmcguire01 at gmail.com> wrote:

> Running build 1.8.5.0 (compiled from source) I seem to be having an issue
> with codec negotiation. I have a Grandstream HT503 FXO port connected to a
> pstn line, a Polycom SP501, and a SIP trunk with callwithus.
>
> What I'm essentially looking to accomplish is for ulaw or g729 (preferably
> ulaw) to be used to the Grandstream FXO or any other internal endpoint, and
> for g729 only to be used outbound to my SIP trunk.
>
> Here are the basics of my config, showing the codec list from "sip show
> peer <peer>":
>
> Polycom SP501 (desk phone):
> --------------------------
> disallow=all
> allow=ulaw&g729
>   Codecs       : 0x104 (ulaw|g729)
>   Codec Order  : (ulaw:20,g729:20)
>
> Grandstream HT503 (fxo gateway):
> --------------------------
> disallow=all
> allow=ulaw&g729
>   Codecs       : 0x104 (ulaw|g729)
>   Codec Order  : (ulaw:20,g729:20)
>
> CallWithUs (SIP trunk):
> --------------------------
> disallow=all
> allow=g729
>   Codecs       : 0x100 (g729)
>   Codec Order  : (g729:20)
>
> When I place an outbound call from the Polycom to callwithus, the invite
> from the pcom shows both ulaw and g729 in the SDP:
> INVITE sip:************@192.168.0.1;user=phone SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.201;branch=z9hG4bKc8aa981a8B8FF58D
> From: "Office" <sip:2001 at 192.168.0.1>;tag=4CD2B2A0-B94A2531
> To: <sip:919785013620 at 192.168.0.1;user=phone>
> [...]
> m=audio 2258 RTP/AVP 18 0 8 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
>
> Asterisk sees this:
> [Aug  2 15:00:31] VERBOSE[1918] chan_sip.c: Capabilities: us - 0x104
> (ulaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0
> (nothing)/text=0x0 (nothing), combined - 0x104 (ulaw|g729)
>
> The call goes out the callwithus trunk:
> [Aug  2 15:00:31] VERBOSE[1315] pbx.c:     -- Executing
> [s at macro-dialout-trunk:19] Dial("SIP/2001-00000047",
> "SIP/CallWithUs/**********,300,tTwW") in new stack
>
> And then this, no INVITE goes out to callwithus at all:
> [Aug  2 15:00:31] WARNING[1315] chan_sip.c: No audio format found to offer.
> Cancelling call to **********
> [Aug  2 15:00:31] VERBOSE[1315] app_dial.c:     -- Couldn't call
> SIP/CallWithUs/**********
>
> Similarly, if I set the Grandstream FXO trunk to ulaw only, the call fails
> as well. It seems as if allowing only a single codec is the issue, if I
> change the priorities of all codecs to g729 first and ulaw second, the call
> goes through as g729 successfully.
>
> Smells like a bug to me, but I may be overlooking something in my config.
>
> Thanks,
>
> -Ryan
>

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