[asterisk-users] Outgoing trunk is restricted to g.729 but need ulaw

[Jack Henery]

Hi,

I am taking over an asterisk system from another person and having an issue
where a sip trunk is restricting the outgoing codecs to just g.729

I am dialing in from a Cisco 7960.  The Invite from the Cisco has the
folowing M line:
m=audio 17022 RTP/AVP 18 0 8 101.
So it is allowing g.729, ulaw and alaw.

Asterisk is tandeming the call out over a SIP trunk

sip.conf tandem trunk config:
   [trunk-out]
   host=192.168.1.6
   type=friend
   disallow=all
   allow=ulaw
   allow=alaw
   allow=gsm
   allow=g729
   context=from-trunk
   nat=no
   qualify=100

But the outgoing Invite has the following m line:
   m=audio 17064 RTP/AVP 18 101.

This system does realtime which I am not really familiar with but the only
stuff that seems relivent is one table called sip_devices with 2 columns
disallowd and allowed.  I think this should only affect the phones though.
For this extension the values are disallowed=all and allowed=g729;ulaw;alaw

I did try to search here and Google but I am not sure what to use for a
search string.

I turned on debug to level 3 :

    -- Executing [s at macro-dialout:36] Dial("SIP/1234-00000039",
"SIP/trunkout/1xxxxxxxxx,60,L(180000:20000)") in new stack
[Mar 19 18:22:56] DEBUG[17418]: chan_sip.c:25057 sip_request_call: Asked to
create a SIP channel with formats: 0x100 (g729)
[Mar 19 18:22:56] DEBUG[17418]: chan_sip.c:7215 sip_alloc: Allocating new
SIP dialog for 4870fab953c16a611b9248584748fe59 at 127.0.0.1:0 - INVITE (No
RTP)
[Mar 19 18:22:56] DEBUG[17418]: rtp_engine.c:344 ast_rtp_instance_new:
Using engine 'asterisk' for RTP instance '0x9ded230'
[Mar 19 18:22:56] DEBUG[17418]: res_rtp_asterisk.c:472 ast_rtp_new:
Allocated port 19718 for RTP instance '0x9ded230'
[Mar 19 18:22:56] DEBUG[17418]: rtp_engine.c:353 ast_rtp_instance_new: RTP
instance '0x9ded230' is setup and ready to go
[Mar 19 18:22:56] DEBUG[17418]: res_rtp_asterisk.c:2370 ast_rtp_prop_set:
Setup RTCP on RTP instance '0x9ded230'
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
[Mar 19 18:22:56] DEBUG[17418]: chan_sip.c:4683 do_setnat: Setting NAT on
RTP to Off
[Mar 19 18:22:57] DEBUG[17418]: acl.c:715 ast_ouraddrfor: For destination
'192.168.1.6', our source address is '192.168.1.25'.
[Mar 19 18:22:57] DEBUG[17418]: chan_sip.c:3243 ast_sip_ouraddrfor: Setting
SIP_TRANSPORT_UDP with address 192.168.1.25:5060
[Mar 19 18:22:57] DEBUG[17418]: chan_sip.c:6557 sip_new: *** Our native
formats are 0x100 (g729)
[Mar 19 18:22:57] DEBUG[17418]: chan_sip.c:6558 sip_new: *** Joint
capabilities are 0x100 (g729)
[Mar 19 18:22:57] DEBUG[17418]: chan_sip.c:6559 sip_new: *** Our
capabilities are 0x10e (gsm|ulaw|alaw|g729)

The receiving asterisk system only does ulaw or alaw.

I am sure it is a mis-configuration somewhere, just not finding it.

Where should I look to enable the other codecs?
What else would help in troubleshooting?

Thank you,
JH
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