I'm not sure if anybody has determined the cause/fix for this problem, but I am getting the same problem.
I turned on syslog debugging and there were some interesting results: 1. phone call answered on line2: Apr 18 11:14:13 192.168.1.21 [0:0]AUD ALLOC CALL (port=16428) Apr 18 11:14:14 192.168.1.21 [0:0]RTP Rx Up Apr 18 11:14:14 192.168.1.21 [1:0]AUD ALLOC CALL (port=16430) Apr 18 11:14:14 192.168.1.21 [1:0]RTP Rx Up Apr 18 11:14:17 192.168.1.21 [1]Off Hook Apr 18 11:14:17 192.168.1.21 [1]CID interrupted Apr 18 11:14:17 192.168.1.21 [1:0]RTP Rx 1st PKT @16430(2) Apr 18 11:14:17 192.168.1.21 [0:0]AUD Rel Call Apr 18 11:14:17 192.168.1.21 CC:Ended Apr 18 11:14:17 192.168.1.21 CC:Connected Apr 18 11:14:18 192.168.1.21 [1:0]ENC INIT 0 Apr 18 11:14:18 192.168.1.21 [1:0]RTP Tx Up (pt=0->c0a80114:15840) Apr 18 11:14:18 192.168.1.21 [1:0]RTCP Tx Up Apr 18 11:14:18 192.168.1.21 [1:0]DEC INIT 0 Apr 18 11:14:22 192.168.1.21 [1]On Hook Apr 18 11:14:22 192.168.1.21 [1:0]AUD Rel Call Apr 18 11:14:22 192.168.1.21 DLG Terminated Apr 18 11:14:22 192.168.1.21 Sess Terminated Apr 18 11:14:49 192.168.1.21 DLG Terminated Apr 18 11:14:49 192.168.1.21 Sess Terminated Apr 18 11:14:54 192.168.1.21 CC:Clean Up 2. Phone call answered on line1 (no outgoing voice) Apr 18 11:01:16 192.168.1.21 [0:0]AUD ALLOC CALL (port=16424) Apr 18 11:01:16 192.168.1.21 [0:0]RTP Rx Up Apr 18 11:01:16 192.168.1.21 [1:0]AUD ALLOC CALL (port=16426) Apr 18 11:01:16 192.168.1.21 [1:0]RTP Rx Up Apr 18 11:01:18 192.168.1.21 [0]Off Hook Apr 18 11:01:18 192.168.1.21 [0]CID interrupted Apr 18 11:01:18 192.168.1.21 Codec 135 not defined in DPT Apr 18 11:01:18 192.168.1.21 Codec 135 not defined in DPT Apr 18 11:01:18 192.168.1.21 CC:Connected Apr 18 11:01:18 192.168.1.21 No Common TxCodec Apr 18 11:01:18 192.168.1.21 [0:0]RTP Rx 1st PKT @16424(2) Apr 18 11:01:19 192.168.1.21 [1:0]AUD Rel Call Apr 18 11:01:19 192.168.1.21 CC:Ended Apr 18 11:01:19 192.168.1.21 [0:0]DEC INIT 0 Apr 18 11:01:21 192.168.1.21 [0]On Hook Apr 18 11:01:21 192.168.1.21 [0:0]AUD Rel Call Apr 18 11:01:21 192.168.1.21 DLG Terminated Apr 18 11:01:21 192.168.1.21 Sess Terminated Apr 18 11:01:50 192.168.1.21 DLG Terminated Apr 18 11:01:50 192.168.1.21 Sess Terminated Apr 18 11:01:53 192.168.1.21 CC:Clean Up Seems like for some reason, line1 is not agreeing on the Tx codec..i tried playing around with the codecs configured in both the sipura configs and asterisk, but could not find anything that worked.. Below is the interesting flash-hook fix: Apr 18 11:17:29 192.168.1.21 [0:0]AUD ALLOC CALL (port=16432) Apr 18 11:17:29 192.168.1.21 [0:0]RTP Rx Up Apr 18 11:17:29 192.168.1.21 [1:0]AUD ALLOC CALL (port=16434) Apr 18 11:17:29 192.168.1.21 [1:0]RTP Rx Up Apr 18 11:17:32 192.168.1.21 [0]Off Hook Apr 18 11:17:32 192.168.1.21 [0]CID interrupted Apr 18 11:17:32 192.168.1.21 Codec 135 not defined in DPT Apr 18 11:17:32 192.168.1.21 Codec 135 not defined in DPT Apr 18 11:17:32 192.168.1.21 [0:0]RTP Rx 1st PKT @16432(2) Apr 18 11:17:32 192.168.1.21 CC:Connected Apr 18 11:17:32 192.168.1.21 No Common TxCodec Apr 18 11:17:32 192.168.1.21 [1:0]AUD Rel Call Apr 18 11:17:32 192.168.1.21 CC:Ended Apr 18 11:17:32 192.168.1.21 [0:0]DEC INIT 0 Apr 18 11:17:35 192.168.1.21 [0]Hook Flash Apr 18 11:17:35 192.168.1.21 [0:0]RTP Tx Dn Apr 18 11:17:35 192.168.1.21 [0:0]RTP Rx Dn Apr 18 11:17:36 192.168.1.21 CC:Hold Apr 18 11:17:36 192.168.1.21 [0:0]RTP Tx Dn Apr 18 11:17:36 192.168.1.21 CC:Remote Resume Apr 18 11:17:36 192.168.1.21 [0:0]RTCP Tx Up Apr 18 11:17:38 192.168.1.21 CC:Connected Apr 18 11:17:38 192.168.1.21 [0:0]RTP Tx Dn Apr 18 11:17:39 192.168.1.21 [0:0]ENC INIT 0 Apr 18 11:17:39 192.168.1.21 [0:0]RTP Tx Up (pt=0->c0a80114:10040) Apr 18 11:17:39 192.168.1.21 [0:0]RTCP Tx Up Apr 18 11:17:39 192.168.1.21 CC:Remote Resume Apr 18 11:17:39 192.168.1.21 [0:0]RTCP Tx Up Apr 18 11:17:39 192.168.1.21 [0:0]RTP Rx Up Apr 18 11:17:39 192.168.1.21 [0:0]RTP Rx 1st PKT @16432(2) Apr 18 11:17:39 192.168.1.21 [0:0]DEC INIT 0 Apr 18 11:17:38 192.168.1.21 [0]Hook Flash Apr 18 11:17:46 192.168.1.21 [0]On Hook Apr 18 11:17:47 192.168.1.21 [0:0]AUD Rel Call Apr 18 11:17:56 192.168.1.21 [1]RegOK. NextReg in 180 Apr 18 11:17:56 192.168.1.21 [0]RegOK. NextReg in 180 Apr 18 11:18:04 192.168.1.21 DLG Terminated Apr 18 11:18:04 192.168.1.21 Sess Terminated Apr 18 11:18:10 192.168.1.21 DLG Terminated Apr 18 11:18:10 192.168.1.21 Sess Terminated Apr 18 11:18:18 192.168.1.21 CC:Clean Up I tried doing some sniffs with ethereal and sip debug, but neither seemed to sow any info like this (of course, I don't know the sip debug stuff that well) w My feeling is that this is a Sipura problem. I've upgraded to the firmare 2.02, but still no difference. Besides that, my configs for line1 and line2 in the sipura configuration are exactly the same, except for SIP port (5060 and 5061 respectively) and the extention numbers (2201 and 2202 respectively) My asterisk config for the sip lines are: [2202] type=friend host=dynamic context=home secret=XXXx callerid="SPA2" <2202> mailbox=2202 dtmfmode=rfc2833 canreinvite=no nat=0 [2201] type=friend host=dynamic context=home secret=Xxxx callerid="SPA1" <2201> mailbox=2202 dtmfmode=rfc2833 canreinvite=no nat=0 I'm also sending a copy of this to the sipura tech support.. Hope this either helps others to possibly find a fix, or if anyone _does_ have a fix, please let me know! I'm pretty close to just returning it, as to me, this simply does not work. Thanks! -Mark ----- Original Message ----- From: Matt McIntyre To: [EMAIL PROTECTED] Sent: Tuesday, March 23, 2004 6:59 PM Subject: RE: [Asterisk-Users] Sipura line 1 outgoing voice problem? I am experiencing this same problem and was wondering if anyone has come to a resolution. I have contacted Sipura but have not heard any response yet and am having trouble determining for sure whether the problem resides with Asterisk or the Sipura. As I have noticed that there are many users on the list who use the Sipura unit without this problem (and even a fellow with one unit that worked and one that did) I think the Sipura must be suspect. __________________ Back in January I started having a problem with my Sipura (and there was at least one other on the list with the same problem) that if I answer an incoming call (via X100P) on line 1 of my Sipura, the caller cannot hear any voice from the internal extension. If the internal user puts the external user on hold (via flash hook) and returns, both directions of audio are fine. Line 2 never has had this problem. For the meantime, I switched the internal phones so that my wife's favorite phone is line 2 and I told her to not pick up with line 1. Not a very permanent solution :) _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
