Summary:	ASTERISK-06636: RealTime and codecs order
Reporter:	alein (alein)	Labels:
Date Opened:	2006-03-28 06:16:35.000-0600	Date Closed:	2011-06-07 14:07:46
Priority:	Minor	Regression?	No
Status:	Closed/Complete	Components:	Channels/chan_agent
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:
Description:	I have Asterisk and gateway from PSTN to SIP. I dial to FXO, my call is transmitted to Asterisk... When Asterisk reads sipusers and sippeers via ODBC, such dialogue occurs: <-- SIP read from 1.2.3.1:5060: INVITE sip:51@1.2.3.4 SIP/2.0 From: "100"<sip:100@1.2.3.4>;tag=c38aaa27-13c4-44292742-cb57-3ee9 To: <sip:51@1.2.3.4> Call-ID: 6886dc-c38aaa27-13c4-44292742-cb52-9b3@1.2.3.4 CSeq: 2 INVITE Via: SIP/2.0/UDP 1.2.3.1:5060;branch=z9hG4bK-44292742-cc3d-3e9c Max-Forwards: 70 Supported: replaces User-Agent: FXO_GW Contact: <sip:100@1.2.3.1:5060> Proxy-Authorization: Digest username="100", realm="a.b.com", nonce="04792561", uri="sip:51@1.2.3.4", response="3d1e3904b1ccfd86f3803f2f8a08187e", algorithm=MD5 Content-Type: application/sdp Content-Length: 370 v=0 o=FXO_GW 12367 0 IN IP4 1.2.3.1 s=FXO_GW Session i=Audio Session c=IN IP4 1.2.3.1 t=0 0 m=audio 16384 RTP/AVP 0 0 4 0 0 0 8 96 a=fmtp:96 0-11 a=rtpmap:0 PCMU/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:4 G723/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:96 telephone-event/8000 --- (13 headers 16 lines)--- Using INVITE request as basis request - 6886dc-c38aaa27-13c4-44292742-cb52-9b3@1.2.3.4 Sending to 1.2.3.1 : 5060 (non-NAT) Found user '100' Found RTP audio format 0 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 0 Found RTP audio format 0 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 96 Peer audio RTP is at port 1.2.3.1:16384 Found description format PCMU Found description format PCMU Found description format G723 Found description format PCMU Found description format PCMU Found description format PCMU Found description format PCMA Found description format telephone-event Capabilities: us - 0x50f (g723|gsm|ulaw|alaw|g729|ilbc), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xd (g723|ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 51 in service1 (domain 1.2.3.4) list_route: hop: <sip:100@1.2.3.1:5060> Transmitting (no NAT) to 1.2.3.1:5060: SIP/2.0 100 Trying v: SIP/2.0/UDP 1.2.3.1:5060;branch=z9hG4bK-44292742-cc3d-3e9c;received=1.2.3.1 f: "100"<sip:100@1.2.3.4>;tag=c38aaa27-13c4-44292742-cb57-3ee9 t: <sip:51@1.2.3.4> i: 6886dc-c38aaa27-13c4-44292742-cb52-9b3@1.2.3.4 CSeq: 2 INVITE User-Agent: SoftSwitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY m: <sip:51@1.2.3.4> l: 0 ... We're at 1.2.3.4 port 18352 Adding codec 0x1 (g723) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 1.2.3.1:5060: SIP/2.0 200 OK v: SIP/2.0/UDP 1.2.3.1:5060;branch=z9hG4bK-44292742-cc3d-3e9c;received=1.2.3.1 f: "100"<sip:100@1.2.3.4>;tag=c38aaa27-13c4-44292742-cb57-3ee9 t: <sip:51@1.2.3.4>;tag=as416141ca i: 6886dc-c38aaa27-13c4-44292742-cb52-9b3@1.2.3.4 CSeq: 2 INVITE User-Agent: SoftSwitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY m: <sip:51@1.2.3.4> c: application/sdp l: 265 v=0 o=root 8819 8819 IN IP4 1.2.3.4 s=session c=IN IP4 1.2.3.4 t=0 0 m=audio 18352 RTP/AVP 4 0 8 96 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - --- Mar 28 15:15:39 WARNING[8819]: channel.c:2333 set_format: Unable to find a codec translation path from g723 to gsm Mar 28 15:15:39 WARNING[8819]: file.c:821 ast_streamfile: Unable to open auth-thankyou (format g723): No such file or directory Mar 28 15:15:39 WARNING[8819]: app_playback.c:132 playback_exec: ast_streamfile failed on SIP/100-cd23 for auth-thankyou ... Mar 28 15:15:40 WARNING[8819]: chan_sip.c:2542 sip_write: Asked to transmit frame type 1, while native formats is 4 (read/write = 4/4) Mar 28 15:15:40 WARNING[8819]: chan_sip.c:2542 sip_write: Asked to transmit frame type 1, while native formats is 4 (read/write = 4/4) Mar 28 15:15:40 WARNING[8819]: chan_sip.c:2542 sip_write: Asked to transmit frame type 1, while native formats is 4 (read/write = 4/4) ... When Asterisk uses sip.conf only (not ODBC), it keeps order of codecs from the sip.conf (disallow=all, allow=...): ulaw, alaw, g723.1, g729, and the error doesn't occure.
Comments:	By: Olle Johansson (oej) 2006-03-29 18:54:06.000-0600 Please tell us the settings in sip.conf and the settings you have in a realtime table By: alein (alein) 2006-03-30 00:20:14.000-0600 Mea culpa... Delete this report, please. By: Olle Johansson (oej) 2006-03-30 12:25:06.000-0600 Ok, good thing that is was solved :-)