similar to: IAX Trunk issue.

Dale, Sorry for taking so long to answer, I've been traveling. Thanks so much for the suggestion, your solution worked perfectly. I'm not sure why I didn't notice that the IAX trunk was working in the other direction. Once again, thanks for your help. Mitch Date: Mon, 25 Jun 2012 05:44:37 -0500 From: Dale Noll <dnoll at wi.rr.com> Subject: Re: [asterisk-users] IAX Trunk

I'm running 1.2.9.1, and I can't get caller id dialplan matching to work. When calling from 9220370 to 1234, the following does not match. exten => 9220370/1234,1,NoOp(${CALLERIDNUM}) exten => 9220370/1234,2,Answer exten => 9220370/1234,3,Playback(tt-weasels) However, when calling from 9220370 to 1234, this DOES match. exten => 1234,1,NoOp(${CALLERIDNUM}) exten =>

Can anyone tell me what I missed? I'm trying to setup a simple extension (400) that reports the time when it is dialed. I searched the threads and it seems like this should work... Here's what's in my extensions.conf: exten => 400,1,Answer() exten => 400,n,Wait,1 exten => 400,n,SayUnixTime(,EST5EDT,) exten => 400,n,Playback(tt-weasels) [BTW, tt-weasels is hillarious!

Call from 'sip' to extension '+1xxxyyyzzzz' rejected because extension not found in context 'out'. But [out] exten => s,1,NoOp( this is the extension: ${EXTEN}) exten => s,n,Answer() exten => s,n(weasels),PlayBack(weasels-eaten-phonesys) ........ If I set "s" to "_." it works. Shouldn't "s" work here? Is it because the

Hi! I just found out that Asterisk (1.4) does not write CDRs if the incoming call was not forwarded but handled internally without answering the call. E.g.: [from_pstn] exten => 997,1,Answer() exten => 997,2,Playback(tt-weasels) exten => 997,3,Hangup() exten => 999,1,Playback(tt-weasels|noanswer) exten => 999,4,Hangup() For incoming calls to 997 a CDR will be written, but not

I've RTFM and Googled but can't seem to get sip_autoreg to work (or perhaps I'm just completely missing the point of it). (what I'd like to do is avoid having to put explicit entries for every SIP phone into extensions.conf). Asterisk is creating entries in the (virtual) context sip_autoreg: asterisk*CLI> dialplan show sip_autoreg [ Context 'sip_autoreg'

Hello, I'm running asterisk-1.0.6 on a centos3.4 box. I'm still in testing phase and so far everything is running smoothly. I'm now trying to play a soundfile or an mp3file using 'MP3Player', 'Playback' or the 'Background' commands, but don't get any sound. The logfile says: -- Executing BackGround("SIP/joa-9def", "tt-weasels") in

Hi All, I've been having a hard time getting the AbsoluteTimeout function to work. Is this Function working in for SIP? I've search all the messages in the news letters and tried what was suggested and still have not gotten it to work. Below is a portion of my extensions.conf. I've also been running these test on ver 0.5.0 exten => _X.,1,Absolutetimeout(20) exten =>

I have this extension: exten => 1322,1,Answer() exten => 1322,n,Set(CfMC_AMDValue="NotChecked") exten => 1322,n,GotoIf($["${CfMC_DoAMD}" != "Yes"]?NOAMD) exten => 1322,n,AMD() exten => 1322,n,Set(CfMC_AMDValue = ${AMDSTATUS}) exten => 1322,n(NOAMD),Wait(1) exten => 1322,n,UserEvent(E1322-1,${CfMC_ActionID}=${CHANNEL} & ${CfMC_AgentToUse}

On 10/01/2017 10:03 PM, Reindl Harald (mobile) via samba wrote: > > Am 02.10.2017 um 06:35 schrieb ToddAndMargo via samba: >> M$'s patches/updates can be miserable and cause all kinds >> of havoc.  It is a judgment call on when and how to install >> M$'s patches/updates.  It is best to make sure you have a good >> anti-virus updated and running.  Your AV is

I had to return my TDM11B because it put the PSTN line 'off hook' the moment I plugged it in and wouldn't hang it up. The new card seems to work because I can actually make an outgoing call from the FXO port to my cell phone, so I'm pretty happy about that. But Asterisk doesn't recognize incoming calls from the PSTN. If I dial my home phone from my cell phone asterisk

I have made bri-stuff.0.0.2rc19 to work (I think) but I can not achieve any in-dialing nor I can dial out; this is what I have from "pri intense debug span 1" command ---------- *CLI> pri intense debug span 1 Enabled EXTENSIVE debugging on span 1 -- Executing Playback("SIP/201-a862", "tt-weasels") in new stack -- Playing 'tt-weasels' (language

Hello everyone, I'm looking for a way to play a sound file to an already established bridged call. It is meant for one party, but it's ok if both parties would hear it. Ideally, I'd like to be able to trigger this from the Management Interface with something like: Action: Playback File: tt-weasels Channel: Zap/nn However, I haven't seen anything like that being

And let's also spell things properly! Like 'internationalisation' ...'Weasels have got into your phone system' instead of 'gotten into your phone system...' And 'please press the hash key..' instead of 'pound key' There should probably be en_uk, en_us, en_ca, en_za, en_nz, en_oz, en_ie and en_in etc to allow each English-speaking country to localise

I'm using wine 1.1.3 and when i try to play Conquer it loads everything and then freezes on the loading screen. I opened Play.exe with terminal and this is what i weasel at weasels-playground:~$ '/home/weasel/.wine/drive_c/Program Files/Conquer 2.0/play.exe' err:alsa:ALSA_CheckSetVolume Could not find 'PCM Playback Volume' element fixme:mixer:ALSA_MixerInit No master control

I installed Asterisk 1.0.9 in a Freebsd 5.4 ( with no zaptel card); I have 2 zoom x5v and works great ( in extensions 123 and 321 ) but I was trying to test cmd Playback, MusicOnHold, MP3Player but when I call to extension 100 I don't hear the sound ( mp3 or gsm that I put) , I only hear noise If I leave a message in a mailbox the same, all the record is noise --------- extensionns.conf

hi friend need ur help in dial plan, i want to allow exten 2000 to 2005 can make call outside and exten 2006 to 2010 can not make call outside. heres my dial plan. ? sip.conf ? [general] port = 5060 bindaddr = 0.0.0.0 context = others [2000] type=friend context=outside secret=1234 host=dynamic [2001] type=friend context=outside secret=1234 host=dynamic [2002] type=friend context=outside

/var/lib/asterisk/sounds/soundfile.alaw /var/lib/asterisk/sounds/soundfile.wav to go from alaw to mp3, first convert to wav, then use lame <options> /var/lib/asterisk/sounds/soundfile.wav /var/lib/asterisk/sounds/soundfile.mp3 sox looks like it can ogg/vorbis, but mine doesn't list mp3. You might fetch the source for sox and see if it can do mp3; lame is probably just as easy to obtain

I'm trying to setup a primitive announcement-paging system in my house using the line-out from my * box to a cheap amplifier that runs to speakers on our first and second floors from the basement. I have a extension that connects to Console, and console is set to auto-pickup. I'm using alsa drivers. This all works great, except for one thing. I want to play a tone over the console after

Hi I'm using asterisk 1.8.25.0 on CentOS 6. I have compiled it with all the calendar modules: *CLI> module show like calendar Module Description Use Count res_calendar.so Asterisk Calendar integration 4 res_calendar_ews.so Asterisk MS Exchange Web Service Calenda 0 res_calendar_caldav.so