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<DIV>The exosip lib may be not parsing the lines with the extra /1 which would require hacking it's code</DIV>
<DIV>We are getting another sip module soon so we can see if the new one has the same problem.</DIV>
<DIV>Thanks, I will keep my eye on it.</DIV>
<DIV><BR> </DIV>Anthony Minessale II<BR><BR>FreeSWITCH http://www.freeswitch.org/<BR>ClueCon http://www.cluecon.com/<BR><BR>AIM: anthm<BR>MSN:anthony_minessale@hotmail.com<BR>JABBER:anthony.minessale@gmail.com<BR><BR>FreeSWITCH Developer Conference<BR>sip:888@66.250.68.194<BR>pstn:712-432-7800
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<DIV style="FONT-SIZE: 12pt; FONT-FAMILY: times new roman, new york, times, serif">----- Original Message ----<BR>From: Sean P. Robertson <spr@ctdistributing.com><BR>To: freeswitch-users@lists.freeswitch.org<BR>Sent: Friday, July 21, 2006 9:17:45 PM<BR>Subject: [Freeswitch-users] SIP SDP Codec Issue<BR><BR><BR>
<P><FONT size=2>I pointed my Zultys IP PBX at my FreeSwitch installation today and FreeSwitch would not accept calls from it.<BR><BR>In the FreeSwitch log is says it is because it "Got no compatible codecs!" ULAW is enabled in the FreeSwitch config and the Zultys system is offering ULAW, but the description that Zultys uses in its SDP looks a little different than it does when a call works through it.<BR><BR>Working (Directly from a Zultys 4x4 Phone):<BR><BR>m=audio 20000 RTP/AVP 0 8 18 101<BR>a=rtpmap:0 PCMU/8000<BR><BR>Failed (From a Zultys MX250 using the FreeSwitch as trunking proxy):<BR><BR>m=audio 20000 RTP/AVP 0 8 18 101<BR>a=rtpmap:0 PCMU/8000/1<BR><BR><BR>Notice the "/1" at the end of the "a" entry on the failed call. I do not know if this is proper or not, but in the interest of interoperability, I thought that I should report the issue.<BR><BR>Attached are ethereal traces of a successful call and a failed one along with the FreeSwitch console output
from the failed call.<BR><BR>If there is a better place to post things like this, please let me know.<BR><BR>Sean<BR><BR><BR><BR><BR></FONT></P>
<DIV>2006-07-21 21:23:54 [DEBUG] mod_exosip.c:1873 log_event() EVENT [EXOSIP_CALL_INVITE]<BR>2006-07-21 21:23:54 [DEBUG] mod_exosip.c:1985 log_event()<BR><- (3 4) [101 Dialog Establishement] for INVITE to: "888" <sip:888@192.168.0.12><BR>2006-07-21 21:23:54 [NOTICE] switch_channel.c:327 switch_channel_set_name() New Chan Exosip/spr-e2d2 [b98514ca-f61d-48c5-989e-ba73784072e5]<BR>2006-07-21 21:23:54 [DEBUG] mod_exosip.c:1346 exosip_create_call() REMOTE SDP:<BR>v=0<BR>o=MX250-3.0.23-0 1153531317 0 IN IP4 192.168.0.11<BR>s=-<BR>c=IN IP4 192.168.0.11<BR>t=0 0<BR>m=audio 20000 RTP/AVP 0 8 18 101<BR>a=rtpmap:0 PCMU/8000/1<BR>a=rtpmap:8 PCMA/8000/1<BR>a=rtpmap:18 G729/8000/1<BR>a=rtpmap:101 telephone-event/8000<BR>a=fmtp:101 0-15<BR>a=ptime:20<BR>2006-07-21 21:23:54 [WARNING] mod_exosip.c:1353 exosip_create_call() Got no compatible codecs!<BR>2006-07-21 21:23:54 [DEBUG] mod_exosip.c:472 exosip_on_hangup() EXOSIP HANGUP Exosip/spr-e2d2 3/4=0<BR>2006-07-21 21:23:54 [NOTICE]
switch_core.c:2447 switch_core_session_destroy() Close Channel Exosip/spr-e2d2<BR>2006-07-21 21:23:54 [DEBUG] mod_exosip.c:1489 destroy_call_by_event() cannot destroy nonexistant call [3]!<BR>2006-07-21 21:23:54 [DEBUG] mod_exosip.c:1873 log_event() EVENT [EXOSIP_CALL_CLOSED]<BR>2006-07-21 21:23:54 [DEBUG] mod_exosip.c:1985 log_event()<BR><- (3 -1) Call Closed<BR>2006-07-21 21:23:54 [DEBUG] mod_exosip.c:1489 destroy_call_by_event() cannot destroy nonexistant call [3]!<BR>2006-07-21 21:23:59 [DEBUG] mod_exosip.c:1873 log_event() EVENT [EXOSIP_CALL_RELEASED]<BR>2006-07-21 21:23:59 [DEBUG] mod_exosip.c:1985 log_event()<BR><- (3 -1) Call released<BR>2006-07-21 21:23:59 [DEBUG] mod_exosip.c:1489 destroy_call_by_event() cannot destroy nonexistant call [3]!</DIV>
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