search for: 0000ms

...Peer Username ID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format (None) 111.111.111.111 xxxxxx 00003/00020 00001/00003 00070ms 0001ms 0071ms g729 (None) 111.111.111.111 xxxxxx 00004/00005 00004/00005 00070ms 0000ms 0070ms g729 (None) 111.111.111.111 zzzzzz 00005/00004 00002/00004 00000ms 0007ms 0071ms g729 (None) 111.111.111.111 xxxxxx 00007/00042 00001/00004 00000ms 0001ms 0071ms g729 IAX2/yyyyyy-14 222.222.222.222 yyyyyy 00014/...

...configured the X-Lite and SIPPS to use GSM codec. Whe I call FWD, I get this info on the channels when the call has not been stablished yet: sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter Format 192.246.69.223 613 1770bf3430d 00102/00000 00000ms 0000ms 2 150.187.xxx.yyy ildefonso C72ACD25-1A 00101/11482 00000ms 0000ms 2 2 active SIP channel(s) -- SIP/fwd-161b answered SIP/ildefonso-d2fc -- Attempting native bridge of SIP/ildefonso-d2fc and SIP/fwd-161b When it gets stablished, I get: sip show channels Peer...

...144 register_verify: No registration for peer 'j' (from 203.99.57.80) advcomm6*CLI> iax2 show channels Channel Peer Username ID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format (None) 203.99.57.80 (None) 00004/15232 00001/00001 00000ms -0001ms 0000ms unknow (None) 203.99.57.80 jaffaradvc 00005/15233 00004/00004 00000ms -0001ms 0000ms unknow (None) 203.99.57.80 (None) 00006/18423 00001/00001 00000ms -0001ms 0000ms unknow 3 active IAX channels Mar 28 03:26:28 DEBUG[2855]: chan_...

...idged Call SIP/1234-faf1 SIP/1234-faf1 (company1 5001 1 ) Up Dial SIP/22225001|20|r 2 active channel(s) sip*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter Format 192.168.1.101 22225001 257684717aa 00104/00000 00000ms 0000ms ULAW 210.17.211.5 1234 003094c2-fd 00104/00102 00000ms 0000ms ULAW 2 active SIP channel(s) Thanks. Ben -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040415/16246be2/attachment....

...alled 526146:secret@iax2.fwdnet.net/612 When the call is trying to connect:- splat*CLI> iax2 show channels Channel Peer Username ID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format (None) 65.39.205.121 (None) 00002/00000 00001/00000 00000ms 0000ms 0000ms UNKN IAX2/65.39.205.121:4 65.39.205.121 596146 00003/00000 00002/00000 00000ms 0000ms 0000ms UNKN 2 active IAX channel(s) And: splat*CLI> iax2 show registry Host Username Perceived Refresh State 65.39.205.121:4569 596146...

..._verify: No registration for peer 'j' (from 203.99.57.80) advcomm6*CLI> iax2 show channels Channel Peer Username ID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format (None) 203.99.57.80 (None) 00004/15232 00001/00001 00000ms -0001ms 0000ms unknow (None) 203.99.57.80 jaffaradvc 00005/15233 00004/00004 00000ms -0001ms 0000ms unknow (None) 203.99.57.80 (None) 00006/18423 00001/00001 00000ms -0001ms 0000ms unknow 3 active IAX channels Mar 28 03:26:28 DEBUG[...

...ent on both of my asterisk servers. Sometimes they disappear for a few seconds and then come back. It always has the same Call ID. voip1*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter Format 192.168.0.102 (None) df92fb1b-8a 00101/03059 00000ms 0000ms UNKN

...IP channel disappear after a call is over. They are always listed as active, even hours later. Here are is the result of "show sip channels": > Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter Format > 172.28.54.160 m4 478c64565be 00104/00000 00000ms 0000ms 0 > 172.28.54.160 m4 4d330ced01e 00104/00001 00000ms 0000ms 0 > 172.28.54.160 m4 0bb7855f15b 00104/00001 00000ms 0000ms 0 > 172.28.54.160 m4 3db8538b4b4 00104/00001 00000ms 0000ms 0 > 4 active SIP channel(s) but none of these ca...

..._verify: No registration for peer 'j' (from 203.99.57.80) advcomm6*CLI> iax2 show channels Channel Peer Username ID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format (None) 203.99.57.80 (None) 00004/15232 00001/00001 00000ms -0001ms 0000ms unknow (None) 203.99.57.80 jaffaradvc 00005/15233 00004/00004 00000ms -0001ms 0000ms unknow (None) 203.99.57.80 (None) 00006/18423 00001/00001 00000ms -0001ms 0000ms unknow 3 active IAX channels Mar 28 03:26:28 DEBUG[...

...e, the SIP side is choosing the wrong codec (though I've tried putting allow=ulaw first and it still didn't help) grover*CLI> iax2 show channels Peer Username ID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format 69.73.19.178 (None) 00003/00000 00001/00000 00000ms 0000ms 0000ms UNKN 69.73.19.178 phoneboy 00005/00102 00019/00017 00099ms 0000ms 0010ms UNKN 2 active IAX channel(s) grover*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Format 10.0.0.250 53 2b890d18-49 00101/00103 ILBC 1 active SI...

Has anybody out there tried to use two different iconnecthere accounts with Asterisk? What I want to do is use a second account if the first is busy. I have tried the following: exten=>_91NXXNXXXXXX,1,StripMSD,1 exten=>_1NXXNXXXXXX,2,Dial,SIP/BYEXTENSION@iconnect ;iconnect is the first account exten=>_1NXXNXXXXXX,3,Dial,SIP/BYEXTENSION@iconnect2 ;iconnect2 is the second account But that

...o 'g729' for this call because of ${SIP_CODEC) variable -- Attempting native bridge of SIP/2016-b119 and SIP/mypstngate-caed *CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter Format 192.168.0.100 0041911234 1f7d34e3642 00102/00000 00000ms 0000ms ULAW 192.168.0.2 2016 4977-4F41-7 00101/00003 00000ms 0000ms G729A 2 active SIP channel(s) [... after hangup ...] == Spawn extension (auth-out, 0911234567, 2) exited non-zero on 'SIP/2016-b119' -- Executing Hangup("SIP/2016-b119", "")...

...even though g.729 is also an 'allowed' codec (w/5 licenses installed). During an average call 'iax2 show channels' provides: Peer Username ID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format 10.0.40.140 astpbx-woo 00002/00002 00005/00006 00040ms 0036ms 0000ms GSM Kris Boutilier Information Systems Coordinator Sunshine Coast Regional District

...have this scenario (CLI output of command "iax2 show channels") IP-AM-PBX*CLI> iax2 show channels Channel Peer Username ID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format (None) 10.229.47.113 REMOTE_SER 06818/14174 00002/00002 00000ms -0001ms 0000ms unknow 1 active IAX channel IP-AM-PBX*CLI> Here I can't issue "soft hangup" command because I haven't a channel to specify ("None" is not a choice :) ). Now the question is: is there a way to drop this (zombie) channel off and release frozen resourc...

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...xited non-zero on 'SIP/202-351f' While the BudgeTone is 'ringing', Asterisk appears to think it has a live connection: *CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter Format 192.168.42.56 202 26e8973e-e3 00101/06079 00000ms 0000ms ULAW 1 active SIP channel(s) Has anyone else tried MOH from a Grandstream BudgeTone extension? It would mean a lot just to hear that it's my local config and not some weird BT problem. If someone could post a working BudgeTone SIP definition, that might help too. Thanks! -Steve

...5 18:51:02 NOTICE[1125329600]: File chan_sip.c, Line 4571 (handle_response): Failed to authenticate on REGISTER to '<sip:<user>@fwd.pulver.com>;tag=as75fc26a2' That continues until asterisk*CLI>sip show channels 65.39.205.114 <usr>@fwd. 3ff9e23356a 00103/00000 00000ms 0000ms UNKN 65.39.205.114 <usr>@fwd. 3ff9e23356a 00102/00000 00000ms 0000ms UNKN 504 active SIP channel(s) tail -4 /var/log/asterisk/messages Sep 25 18:31:00 WARNING[1125329600]: File rtp.c, Line 708 (ast_rtp_new): Unable to allocate socket: Too many open files Sep 25 18:31:00 WARN...

...auth=md5 secret=mypassword disallow=all allow=ulaw qualify=5000 linux-home*CLI> iax2 show channels Channel Peer Username ID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format IAX2/teliax@teliax/7 66.234.228.170 teliax 00007/00309 00001/00000 00000ms 0000ms 0000ms unknow linux-home*CLI> iax2 debug IAX2 Debugging Enabled linux-home*CLI> Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 00017ms SCall: 00005 DCall: 00000 [63.218.7.228:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno:...

...62.254.245.18",response="466777cb9c995d5c862fc5310d33477d",nonce="09f50874",algorithm=md5 10 headers, 0 lines *CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter Format 62.254.245.14 3046 171697661ab 00102/00000 00000ms 0000ms 4 192.168.101.186 2001 000ab714-51 00101/00103 00000ms 0000ms 4 2 active SIP channel(s) *CLI>

This is kind of a repost of one part of a previous question I have had. Peer Username Call ID Seq (Tx/Rx) Lag Jitter Format 213.137.73.178 xxxxxxxxxx 3705df0a5f7 00103/00000 00000ms 0000ms 4 1 active SIP channel(s) I see that there is 0ms Jitter set. How can I set a Jitter buffer for use with sip channels? I can't seem to find any documentation about this. Any help is always appreciated.