search for: 00103

I have recently posted a message regarding hung SIP channels when using the Monitor() command. Well, I was wrong.   The channel hanging wasn't caused by the Monitor command. They also hang when they aren't monitored. The cause seems to be our PSTN gateway provider. When for some reason their PSTN gateway crashes or reboots (OK, this should happen, but anyway...) and RTP/SIP data stops

...different brands of devices. The Peer with IP 10.204.10.12 is an AudioCodes MP-124, and every other IP is a number of Welltech 3504A 4-port FXS devices. asterisk-srv1*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Format 10.204.10.12 161913 2d381f58106 00103/00000 UNKN (d) 10.204.10.12 161913 1e68f3610c9 00103/00000 UNKN (d) 10.204.10.12 463913 28862156821 00102/05918 UNKN (d) 10.204.10.12 468945 25028137781 00102/16213 UNKN (d) 10.204.10.20 305129 57f9ac-acc0 00102/00002 UNKN (d) 10.204.10.15 467040...

I am a fairly new user of Asterisk and I am generally impressed with its features. I have some questions about the SIP channel support: 1. I have noticed that even when there are no active calls, there is a list of active SIP channels. This appears to be a bug. Has anyone seen this? 2. If there are stuck SIP channels, how can one clear them without re-starting Asterisk? 3. One time a

Hello all, When I do 'sip show channels' I have seen a lot of entries where these calls has already been terminated. Some of these channels are bolong to calls being made 2 days ago but still showing from the CLI. They look like 10.223.51.173 0022676583 130b36625fc 00102/00103 unknow(d) Rx: BYE 10.223.51.173 0022676583 5533069e578 00102/00103 unknow(d) Rx: BYE 10.223.51.173 0016513973 234f7bba140 00102/00103 unknow(d) Rx: BYE 10.223.51.173 0027226765 487b770b231 00102/00103 unknow(d) Rx: BYE 10.223.51.173 0016513973 69b59aa2084 00102/00103 u...

...Spectralink phones have entries in 'sip show channels' that do not go away. None of the other phones do this. Is there anyway to remove these entries without restarting Asterisk? Any ideas on what could be done to prevent this? Example output: xxx.xxx.xxx.xxx 541 14dd18886d1 00103/00102 0x0 (nothing) No Rx: BYE xxx.xxx.xxx.xxx 546 1e7c2fd84ab 00103/00102 0x0 (nothing) No (d) Rx: BYE xxx.xxx.xxx.xxx 546 80f99ee6-6c 00103/00104 0x0 (nothing) No Rx: BYE xxx.xxx.xxx.xxx 546 0d9b184254b 00104/00102 0x0 (nothing) No...

...our Cisco 7206 SIP->PRI gateway.) asterisk*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Format 65.72.107.2 8327549222 1758081f67e 00102/00000 ulaw 10.0.0.48 3035 0008a3d2-05 00101/00102 ulaw 10.0.0.48 3035 0feb1c11386 00103/00101 g729 65.72.107.2 5126800422 28D20837-69 00103/00101 g729 4 active SIP channel(s) 5126800422 is still on hold while 3035 talks with 8327549222. 3035 now presses 'Join' on her phone and this happens: asterisk*CLI> sip show channels Peer User/ANR Call ID...

Hello, In a couple of Asterisks, after type "sip show channels" we have a lot of these: IP_PEER dst_number something 00102/00103 unkn No (d) Rx: BYE IP_PEER dst_number2 something2 00102/00103 unkn No (d) Rx: BYE We are using ASterisk 1.2.x When I say "a lot" I mean more than 180, more than 230, etc. Is it normal? How we can remove it? Thank you very much, -- Carles Pina i Estany GPG id: 0x8CBDAE64 h...

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

I put canreinvite=yes in my sip, for a sipura 3000 and a xlite, however, if I call the traffic still go throw the asterisk. How come? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060617/8f4449fa/attachment.htm

...Cisco phone to an external phone, all the Cisco-user hears is a ticking crackle and after about a minute the phone disconnects. A 'sip show channels' reveals the following: Peer User/ANR Call ID Seq (Tx/Rx) Format [VoIP-provider] [ext. number dialed] 5b1fe97c04d 00103/00000 g729 [IP of Cisco phone] [ID of Cisco] 0002b9a7-4b 00102/00102 ulaw 2 active SIP channel(s) Here g729 pops up, even though I have configured [VoIP-provider] to only allow/use ulaw/alaw. asterisk -vvv shows: -- Executing Dial("SIP/[ID of Cisco]-4663", "S...

...s In this case, when Asterisk user hang up and remote phone still not hang up, I do show like this Show channel verbose 0 active channels 0 active calls Sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message A.B.C.D 112233445566 14448d41170 00103/00104 unkn No (d) Rx: BYE CLI> sip show channel 14448d41170ac3a66a41602575476d5f@W.X.Y.Z * SIP Call Direction: Outgoing Call-ID: 14448d41170ac3a66a41602575476d5f@W.X.Y.Z Our Codec Capability: 256 Non-Codec Capability: 1 Their Codec Capability:...

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.

...00 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 SIP channel(s) Is there a problem with iaxtel? Any ideas? Asterisk CVS-HEAD-07/06/04-01:33:49 built by root@grover on a i686 running Linux -- PhoneBoy

Can someone tell me how to clear hung SIP channels in asterisk without restarting? Currently I have 62 channels and only show 10 in use.. this is some of the sip show channels output.. xxx.xxx.xxx.xxx 00xxx24xxx 04240xxxxxx 00103/00001 UNKN (d) ?How can I remove these? from * without rebooting? ? .o-------------------------------------------------------o. Brian Fertig Network Engineer Planet Telecom, Inc. Tampa, FL Office 813.864.3161x107 Office 813.864.3164 Direct 813.817.9961 Cellular 813.881.9762 Fax Web: www.planet-...

...lished at SIP level, but there's no RTP traffic between the machines. If I make a "sip show channels at the Asterisk console, I see: server*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message 192.168.1.101 phone_A 10095d01445 00103/00000 ulaw No Tx: ACK 192.168.1.107 phone_B 182E175F-F6 00102/00002 ulaw No Tx: ACK 2 active SIP channels (ULAW?!?!?, not even ALAW!!!) As far as I understand, since in this case the communication can not be established directly between A and B (i.e. bypassing Asterisk as the medi...

...Up Bridged Call(SIP/5303-089f1558 SIP/5303-089f1558 s@macro-dial:10 Up Dial(SIP/5301||) 2 active channels 1 active call asterisk1*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message 192.168.5.203 line0 2eeb3516264 00103/00000 ulaw No Tx: ACK 192.168.5.203 5303 28948-0xca0 00102/00001 ulaw No Tx: ACK 2 active SIP channels asterisk1*CLI> show hints asterisk1*CLI> -= Registered Asterisk Dial Plan Hints =- 5303 : SIP/5303 State:InUse Watchers 1 5301...

...assing it to SIP/02YYYYYYYY-e487 -- SIP/sales-7d0b answered SIP/02YYYYYYYY-e487 -- Attempting native bridge of SIP/0282058347-e487 and SIP/sales-7d0b asterisk*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Format 202.177.222.24 02XXXXXXXX 01f672b7696 00103/00000 g729 202.177.222.24 02YYYYYYYY 447542a4000 00101/31350 g729 4 active SIP channel(s) (I changed the numbers to XXXXXXXX and YYYYYYYY in the debug output as well) Thanks in advance, Paul _________________________________________________________________ New year, new job – there'...

...Content-Length: 0 --- (10 headers 0 lines)--- Destroying call '0a72198652f1d9677bbb1c19350ec6f9@192.168.1.108' sip no debug SIP Debugging Disabled *CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Format Last Msg 147.135.0.128 1509XXX57X 480fa3c14d1 00103/00103 ulaw Tx: ACK 1 active SIP channel(s) *CLI> soft hangup SIP/sip.broadvoice.com-5604 Requested Hangup on channel 'SIP/sip.broadvoice.com-5604' << Hangup on console >> *CLI> hangup *CLI> I'd post my sip.conf, but it's pretty much configured as it is i...

...249' Sep 25 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 2...

...7d",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>