Hi Guys, <br>
<br>
My Scenario is <br>
<br>
3Gphone -> (3G network provider)->(Softswitch Cisco PGW)->SIP<-Asterisk<-SIP->SIP phone<br>
<br>
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. <br>
<br>
Am I right or the problem is somewhere else? Please take a look of my config and the trace of the asterisk cli. <br>
<br>
<br>
sip.conf <br>
<br>
[general]<br>
<br>
videosupport=yes<br>
disallow=all
; First disallow all codecs<br>
allow=alaw
; Allow codecs in order of preference<br>
allow=h263<br>
allow=h263p<br>
allow=h261<br>
<br>
[32515901]<br>
type=friend<br>
secret=phone1<br>
host=dynamic<br>
;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info<br>
mailbox=1000 ; Mailbox for message waiting indicator<br>
context=sip<br>
videosupport=yes<br>
maxcallbitrate=128<br>
callerid= "test" <32515901><br>
allow=alaw<br>
allow=speex<br>
allow=gsm<br>
allow=h261<br>
allow=h263<br>
allow=h263p<br>
<br>
<br>
Debug <br>
<br>
<--- SIP read from <a href="http://my.domain.com:5060">my.domain.com:5060</a> ---><br>
INVITE <a href="mailto:sip:32515901@172.18.10.100">sip:32515901@172.18.10.100</a>;user=phone SIP/2.0<br>
Via: SIP/2.0/UDP <a href="http://my.domain.com:5060">my.domain.com:5060</a> ;branch=z9hG4bKterm-30-myphone-32515901-17145<br>
From: myphone <<a href="mailto:sip:myphone@my.domain.com">sip:myphone@my.domain.com</a>;user=phone>;tag=1763495500<br>
To: 32515901 <<a href="mailto:sip:32515901@172.18.10.100">sip:32515901@172.18.10.100</a>;user=phone><br>
Call-ID: <a href="mailto:3f7a5361-3057ba40-5f7fc171-25@my.domain.com">3f7a5361-3057ba40-5f7fc171-25@my.domain.com</a><br>
CSeq: 1 INVITE<br>
Supported: timer<br>
Session-Expires: 1800<br>
Min-SE: 1800<br>
Contact: <<a href="http://sip:myphone@my.domain.com:5060">sip:myphone@my.domain.com:5060</a>><br>
Allow: INVITE,ACK,PRACK,SUBSCRIBE,BYE,CANCEL,NOTIFY,INFO,REFER,UPDATE<br>
Max-Forwards: 70<br>
Content-Type: application/sdp<br>
Content-Length: 317<br>
<br>
v=0<br>
c=IN IP4 <a href="http://85.118.195.7">85.118.195.7</a><br>
m=audio 18010 RTP/AVP 125<br>
a=rtpmap:125 G.nX64/8000<br>
a=X-pc-codec: 125 101 100<br>
a=rtpmap:125 G.nX64/8000<br>
a=rtpmap:101 /8000<br>
a=rtpmap:100 /8000<br>
a=X-sqn:0<br>
a=X-cap: 1 audio RTP/AVP 100<br>
a=X-cpar: a=rtpmap:100 X-NSE/8000<br>
a=X-cpar: a=fmtp:100 192-194,200-202<br>
a=X-cap: 2 image udptl t38<br>
<br>
<-------------><br>
--- (14 headers 13 lines) ---<br>
Using INVITE request as basis request - <a href="mailto:3f7a5361-3057ba40-5f7fc171-25@my.domain.com">3f7a5361-3057ba40-5f7fc171-25@my.domain.com</a><br>
Found peer 'test'<br>
Found RTP audio format 125<br>
Peer audio RTP is at port <a href="http://85.118.195.7:18010">85.118.195.7:18010</a><br>
Found description format G.nX64 for ID 125<br>
Found description format G.nX64 for ID 125<br>
Capabilities: us - 0x1c0008 (alaw|h261|h263|h263p), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing)<br>
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)<br>
[Nov 15 23:15:53] NOTICE[22446]: chan_sip.c:4996 process_sdp: No compatible codecs, not accepting this offer!<br>
<br>
<--- Reliably Transmitting (no NAT) to <a href="http://my.domain.com:5060">my.domain.com:5060</a> ---><br>
SIP/2.0 488 Not acceptable here<br>
Via: SIP/2.0/UDP <a href="http://my.domain.com:5060">my.domain.com:5060</a> ;branch=z9hG4bKterm-30-myphone-32515901-17145;received=<a href="http://my.domain.com">my.domain.com</a><br>
From: myphone <<a href="mailto:sip:myphone@my.domain.com">sip:myphone@my.domain.com</a>;user=phone>;tag=1763495500<br>
To: 32515901 <<a href="mailto:sip:32515901@172.18.10.100">sip:32515901@172.18.10.100</a>;user=phone>;tag=as11b984c5<br>
Call-ID: <a href="mailto:3f7a5361-3057ba40-5f7fc171-25@my.domain.com">3f7a5361-3057ba40-5f7fc171-25@my.domain.com</a><br>
CSeq: 1 INVITE<br>
User-Agent: Asterisk PBX<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
Supported: replaces<br>
Content-Length: 0<br>
<br>
Appreciate any help, <br>
<br>
Niko <br>
<br>