similar to: Zap Channels Hang

Hello, I am migrating a PABX system based in Asterisk 1.4 to Asterisk 13, with success. I have an application that sends an action Originate to AMI for calling, it's working well, but when i see to Asterisk's CLI, i see 2 calls for just one originate: pftestes40copiabh*CLI> core show channels verbose Channel Context Extension Prio State Application

Does anyone know if the 1-800 iaxtel gateway is down? I've been trying to use it all day today and asterisk says it's ringing: Channel (Context Extension Pri ) State Appl. Data IAX2[iaxtel]/1 ( s 1 ) Ringing AppDial (Outgoing Line) SIP/2201-a253 (home 18888476626 1 ) Ring Dial IAX2/XXX:YYYY@iaxtel.com/18888476626@iaxtel But I

Hi, I am new in asterisk and i've bought a X100p and a TDM400... First of all, how can i verify my config files ? Secondly, when i'm trying to pass a call to the outside, i ve a Notice about appdial.c (l 554) telling me: unable to create channel of type Zap ...and i don't understand... Finally, when i plug my analog phones in RJ45 of my TDM400, there is no tonality ( i'm not

Hi, I compiled and am running the latest CVS but strange things are now happening.. it looks like asterisk is randomly declaring my calls to be fax calls, complaining and then sending the calls into a black hole... if I hangup the calls below (soft hangup) asterisk locks up and I have to kill the process. NOTICE[21526]: File chan_zap.c, Line 3520 (zt_read): Fax detected, but no fax

I am trying to run the make command to install Asterisk, but I get the following error: make ... ... checking for tgetent in -ltermcap ... no checking for tgetent in -ltinfo ... no checking for tgetent in -lcurses ... no checking for tgetent in -lncurses ... no configure: error: termcap support not found I am running Mandrake 9.1 on a Pentium II 200MHz. Could this be a hardware issue? I

I am trying to use SoftHangup in my dialplan, but it's either not working or I'm not using it correctly. when i'm on the console, i see: pbx1*CLI> core show channels Channel Location State Application(Data) SIP/vgw1-000000a2 2156181505 at inbound:1 Up AppDial((Outgoing Line)) SIP/143-0000009f s at macro-SaferSIPDial Up Dial(SIP/99302156181505 at vgw1,, 2 active

When I type: asterisk -rx "core show channels" I usually get Channel Location State Application(Data) SIP/pstn-4444-000003 7807574622 at internal: Up Dial(SIP/77807574622 at pstn-9998 SIP/pstn-9998-000003 (None) Up AppDial((Outgoing Line)) Is there a way to pull information about time the channel started? -- Joseph

When I issue a 'core show channels' command I notice that long usernames (and channel number) are truncated. For example, if the username is FONEMITEL1234567890 for a trunk, then it will show SIP Privilege: Command Channel Location State Application(Data)" IAX2/FONEMITEL123456 1296197222 at entryhome<mailto:1296197222 at entryhome> Ringing

Hi, We are using Vicidial and sometime even when agent disconnects, outgoing call originated by dialer is still active. Since call was initiated by dialer and then bought into meetme conference of agent and we can't corelate this call to any agent channel. When agents are dialing, channels doesn't show calls vicidial2*CLI> show channels Channel Location

Just wondering if there's any significance to the jacks on the TDM cards. They appear to be RJ-45 instead of RJ-11, and I wasn't quite sure if that's something that makes a difference from the user perspective. . . Thx. B.

This just started after upgrading to 11.11.0. After a call is completed (both ends hang up) the call still shows as active. # asterisk -x "core show channels" Channel Location State Application(Data) SIP/thinktel-0000000 (None) Up AppDial((Outgoing Line)) SIP/4164251212-00000 4165555555 at LocalSets Up Dial(SIP/thinktel/4165559999) 2 active

Fry's Electronics has a new Linksys 2 line ATA box for sale for $59.99 retail. They have a version with a router for $89.99. We picked the non-router version up and it seems to be a rebadged Sipura SPA-2000. The box has a Vonage service package inside as well, but it does work with other services. The box also has a "User Guide" meant for end-users that is very well written [no

Hello asteriskers, I wanted to force a hangup of a SIP to SIP call from the Asterisk CLI> prompt, and found references on using the command "soft hangup <SIP/channel>", but as you can see below, the "soft hangup" command does not seem to exist, and there is no mention about it in the UPGRADE*.txt documents. Can anyone shed light on what would replace "soft

Thank you, I needed a starting point to start my post. 1. Asterisk 12.8.1 (IAX2 voice issues) no video issues. Voice issues on IAX2 Trunks, All extensions are SIP. The IAX2 trunks on Asterisk 12.8.1 produces only one error out of : iax2 set debug trunk on [2015-03-10 16:28:42] WARNING[9614][C-0000000b]: chan_iax2.c:1793 compress_subclass: Can't compress subclass 2097217 On the box running

sorry, this problem is fixed by myself. we must need set 'canreinvite=no' each user. --- I'm try to discconect a call with SIP. when caller make a call, 'show channels' result is following. mack*CLI> show channels Channel (Context Extension Pri ) State Appl. Data SIP/mack-1bfc (default 1 ) Ringing AppDial (Outgoing

Hi, I have a case where SIP channels will not be destroyed, resulting in further calls to ChanIsAvail() to fail. The process (I believe) to replicate this is the following: - Make a call to another SIP phone that is an "intercom" call (Auto-Answer) - For whatever reason, the phone happens to go UNREACHABLE during this call - Phone comes back REACHABLE, but channel still exists in

So I've been testing asterisk under LXC for a few days now and am very happy with the results. My test server is a 1.8GHz Celeron with 256KB cache and 512MB RAM and I have 20 containers each running asterisk (and apache/php,sendmail and a few other minor things) More for fun than anything else, I've tried daisy-chaining instances together - so 20 asterisks running on the same host, 0

Im trying to inter-connect my current PBX system and Asterisk. Asterisk has some users from different networks (internet).. I used cisco router using 4 fxs to pbx and SIP to asterisk. Is there any way i can allow the ip address of cisco to connect to my asterisk using SIP? IP Address of cisco is 192.168.0.254 here's a part of my sip.conf [general] port = 5060 bindaddr = 0.0.0.0

I have two calls (626 and 542) coming into the same phone (524). SIP/524-000005b5!smvoice-sip!!1!Up!AppDial!(Outgoing Line)!_2XX!!3!9!SIP/542-000005b4 SIP/542-000005b4!smvoice-sip!_2XX!8!Up!Dial!SIP/524|30|!542!!3!9!SIP/524-000005b5 SIP/524-000005b3!smvoice-sip!!1!Up!AppDial!(Outgoing Line)!_2XX!!3!40!SIP/526-000005b2

Hi, 1) is there a function like "zap destroy channel" to destroy sip channels? Zap/10-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/-081aee08 (pstn-out s 7 ) Ring Dial Zap/g1/0123456789|50|g Zap/8-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/-081aee08 (pstn-out s