similar to: Fun with virtual asterisks ...

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

Hi, i have an asterisk box running with E100P (E1) line as PSTN gw. Sometimes zap channels hang and i couldn't make any PSTN calls but SIP calls are still fine. When this happens I also couldn't restart/reload asterisk from the CLI. I have to kill the asterisk process and run safe_asterisk again. any ideas? asterisk*CLI> show channels Channel (Context

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

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

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

Through the AMI how can I tell if a call is on hold or not? I am using 1.4.X Thanks, jerry

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

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

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

Hello, I want to place an Asterisk-server A in front of 2 other Asterisk-servers (B1 & B2). This first Asterisk-server A needs to send incoming calls to one of the 2 available Asterisk-servers (B1 or B2) behind it. So I want the first Asterisk-server A to accept the call, and based upon some checks in the dialplan send the call through to one of the other Asterisk-servers (B1 or B2)

I recently compiled asterisk 13.2.0 on an RK3288 , I am suspecting problems with the format H264, Asterisk 12.8.1 compiled on the same hardware is behaving very well for the same format H264 Problem of asterisk 12.8.1 is IAX2 trunk bad voice quality. Could someone investigate the problem of Asterisk 13 with video support on H264 ? Thank you. -------------- next part -------------- An

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

Keep in mind this is 1.2... I have a peer in sip.conf: [my-uk900] context = uk900 host = a.b.c.d type = friend Why am I seeing DNS queries like my.example.com and uk900.example.com? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST Newline

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

I am having a problem when a SIP registration fails. I get the following messages in the log: Sep 25 18:51:02 NOTICE[1125329600]: File chan_sip.c, Line 2874 (sip_reg_timeout): Registration for '<user>@fwd.pulver.com@65.39.205.114' timed out, trying again Sep 25 18:51:02 NOTICE[1125329600]: File chan_sip.c, Line 5119 (handle_request): Registration from

I am trying to collect enough information about an problem a client is having with its asterisk 11.17.0 x86_64. This issue was observed before in 1.8.20, and we upgraded to 11.15.0 and then to 11.17.0 with no solution. Background: this client is a telemarketing call-center that generates outgoing calls with close to a hundred agents operating simultaneously in peak hours. The system uses

Hi, I'm having alot of trouble understanding how to use dialplans for outgoing calls on Dahdi. Context : I have 3 TI spans, so 69 voice channels and three D channels (24,48,72). This is on a TE420B from Digium, if it matters. Here are my (apparently simple) questions in no particular order: 1) Dial(DAHDI/5555555555|20) doesn't work. But Dial(DAHDI/42/5555555555|20) does