[Asterisk-video] 3G to SIP problem

[Nikolay Milovanov]

Hi Guys,

My Scenario is

3Gphone -> (3G network provider)->(Softswitch Cisco
PGW)->SIP<-Asterisk<-SIP->SIP phone

I am using Asterisk 1.4 beta 3.  I am calling from 3G to SIP. As I see from
the trace Asterisk is not supporting the clear chanel codec (a=rtpmap:125
G.nX64/8000) used by the PGW.

Am I right or the problem is somewhere else? Please take a look of my config
and the trace of the asterisk cli.


sip.conf

[general]

videosupport=yes
disallow=all                    ; First disallow all codecs
allow=alaw                      ; Allow codecs in order of preference
allow=h263
allow=h263p
allow=h261

[32515901]
type=friend
secret=phone1
host=dynamic
;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
mailbox=1000 ; Mailbox for message waiting indicator
context=sip
videosupport=yes
maxcallbitrate=128
callerid= "test" <32515901>
allow=alaw
allow=speex
allow=gsm
allow=h261
allow=h263
allow=h263p


Debug

<--- SIP read from my.domain.com:5060 --->
INVITE sip:32515901 at 172.18.10.100;user=phone SIP/2.0
Via: SIP/2.0/UDP my.domain.com:5060;branch=z9hG4bKterm-30-myphone-32515901-17145
From: myphone <sip:myphone at my.domain.com;user=phone>;tag=1763495500
To: 32515901 <sip:32515901 at 172.18.10.100;user=phone>
Call-ID: 3f7a5361-3057ba40-5f7fc171-25 at my.domain.com
CSeq: 1 INVITE
Supported: timer
Session-Expires: 1800
Min-SE: 1800
Contact:  <sip:myphone at my.domain.com:5060>
Allow: INVITE,ACK,PRACK,SUBSCRIBE,BYE,CANCEL,NOTIFY,INFO,REFER,UPDATE
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 317

v=0
c=IN IP4 85.118.195.7
m=audio 18010 RTP/AVP 125
a=rtpmap:125 G.nX64/8000
a=X-pc-codec: 125 101 100
a=rtpmap:125 G.nX64/8000
a=rtpmap:101 /8000
a=rtpmap:100 /8000
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 192-194,200-202
a=X-cap: 2 image udptl t38

<------------->
--- (14 headers 13 lines) ---
Using INVITE request as basis request -
3f7a5361-3057ba40-5f7fc171-25 at my.domain.com
Found peer 'test'
Found RTP audio format 125
Peer audio RTP is at port 85.118.195.7:18010
Found description format G.nX64 for ID 125
Found description format G.nX64 for ID 125
Capabilities: us - 0x1c0008 (alaw|h261|h263|h263p), peer - audio=0x0
(nothing)/video=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0
(nothing), combined - 0x0 (nothing)
[Nov 15 23:15:53] NOTICE[22446]: chan_sip.c:4996 process_sdp: No compatible
codecs, not accepting this offer!

<--- Reliably Transmitting (no NAT) to my.domain.com:5060 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP
my.domain.com:5060;branch=z9hG4bKterm-30-myphone-32515901-17145;received=
my.domain.com
From: myphone <sip:myphone at my.domain.com;user=phone>;tag=1763495500
To: 32515901 <sip:32515901 at 172.18.10.100;user=phone>;tag=as11b984c5
Call-ID: 3f7a5361-3057ba40-5f7fc171-25 at my.domain.com
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

Appreciate any help,

Niko
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