similar to: 13 sec. delay what is causing it?

Asterisk stopped answering the calls. I'm just experimenting with asterisk, upon setup there is a [demo] context. I have SPA-3000 with PSTN line: Dial plan 2: S0<:1000@10.0.0.101> my sip.conf localnet = 10.0.0.101 localmask = 255.255.255.0 [3000] type=friend host=dynamic username=3000 secret=my_secret mailbox=3000 context=from_pstn callerid="PSTN GW" <3000>

Dear all I have TDM808B 8 port FXO it is configure perfectly but i got some problem of incomming phone Hangup and callerid display problem i am going to explain you the issue i have install asterisk 1.4 and i have 100 of SIP phone now everything is fine but problem is when i incoming call on FXO and dial sip extention SIP phone is rining but when i disconnect my incoming

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

Hi, I running into the following problem on my Asterisk setup: --snip-- [Sep 3 01:40:59] NOTICE[9170] chan_dahdi.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 3 [Sep 3 01:47:07] ERROR[9170] chan_dahdi.c: No more room in scheduler [Sep 3 01:47:07] ERROR[9170] chan_dahdi.c: Asked to delete sched id -1??? [Sep 3 01:47:07] ERROR[9170] chan_dahdi.c: No more room in scheduler [Sep

On Fri, 18 Mar 2016, Trey Hilyard wrote: > I thought this would be as easy as > exten => _XXXXXXXXXX\;rn=+19136630000,1,Goto(from_pstn,${EXTEN:0:10}) Have you tried the '_!.' pattern? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST

I'm a bit confused about how WaitExten works. I assumed that when it returns 0, the next priority in the extension would be executed, but that doesn't seem to be the case. When I get to WaitExten and enter extension 8, it plays the message, then Waits another 10 seconds and times out. [local] exten => s,1,Wait,1 ; Wait a second, just for fun exten =>

On Fri, Mar 18, 2016 at 10:49 AM Administrator TOOTAI <admin at tootai.net> wrote: > Le 18/03/2016 16:20, Trey Hilyard a ?crit : > > I am trying to set up my Asterisk server so that it will recognize an > > incoming call to the Asterisk's own Location Routing Number (LRN), > > validating the "rn" in the INVITE and then using the Called Number from >

I have posted soo many times in the past but never recieved even a single reply . seem like you people are ignoring me or either way too busy .. never mind this is my last try . How can record a conversation with asterisk ? I tried to use Record() but dint work for me .. here is what i tried . exten => s,1,Wait,1 ; Wait a second, just for fun exten => s,2,Answer

I am running * with a Junghanns quadbri that should allow us to integrate our ISDN house telephone system with VOIP. Preferably I would like to run a setup, so that our internal ISDN phones on an S bus are not aware that * is sitting in between. With the configuration below I run into the following problems: 1. On outbound calls, I get the normal rining call progress tone althought the

I am using a combo of static files and Asterisk Realtime configuration. This section works fine when a static file: --------------------------- [from_pstn] ;Voipgate exten => 4507,1,Goto(from_pstn,s,1) exten => s,1,Macro(dial-ext) exten => s,2,Hangup --------------------------- But, when I drop it in the database and try it in Realtime mode I get this error: ---------------------------

Well I dont want it to operate as a group. I understand that it is *8 on most PBX systems but I would like it to work as dialing *8+extension-thats-ringing to have it go over to my extension. I am having a lot of trouble finding more information from the wiki and google. Thanks! -James -----Original Message----- From: asterisk-users-bounces@lists.digium.com

Dear all I am going to setup Asterisk Call center solution and i have setup my queue and agent i have 2 SNOM ip phone but when i call to queue my agent phone is rining without musicnhold or when both phone is busy then i call to queue its directy hangup without musicnhole means my call not goes in to queue what is the problem my queue.conf [root at pbx asterisk]# cat

I'm testing Teliax tall free number line and I'm experiencing long delay about 1sec. during conversation. When I call myself over FWD the response is normal no delay or cut messages. When I call my number over FWD the is a long delay, welcome message usually cuts off few first words and during conversation my voice arrives after about 1sec. delay. Since, the 800-number is only accessible

It appears that my X100P card is only answering after two rings. Ideally, I'd like it to answer on the first ring. Here is the incoming section of my extensions.conf file: [incoming] exten => s,1,Answer exten => s,2,BackGround(demo-congrats) ; Play a congratulatory message exten => 1234,1,Goto(jgunther,1234,1) exten => 4321,1,Goto(mgunther,4321,1) exten =>

Hi all, I'm researching IP phones for a new office setup. We will need 30 phones. I have read the wiki and the polycom site for the phones, but I still had some questions about real world experience with these phones. -According to the documentation, the 500 series ( and 600, according to a polycom rep ) have built in hubs. Has anybody noticed performance issues in this setup, when

I have a TDM400P with 2 FXO cards and I'm using Asterisk@Home 2.8 I noticed that when I place a call to the analog lines from outside, Asterisk takes a while to actually ring the extension the call is being sen to. I've been doing some tests, calling from my cellphone and here is what I see... - After the first ring on my cell, Asterisk logs to the CLI that is has an incoming call -

i call to my tollfree number buy my CLI send the next error: Aug 14 08:15:22 WARNING[25931]: format_wav.c:169 check_header: Unexpected freqency 22050 Aug 14 08:15:22 WARNING[25931]: file.c:441 ast_filehelper: Unable to open file on /var/lib/asterisk/sounds/procall3.wav Aug 14 08:15:22 WARNING[25931]: file.c:828 ast_streamfile: Unable to open procall3 (format ulaw): No such file or directory Aug

Hello, I ordered the Devel lite kit, and installed it. I am just trying to get the FXO port to work, and am having trouble. To load the card I do the following. modprobe wcfxs modprobe wcfxo ztcfg -vv asterisk -vc My /var/log/asterisk/messages show Nov 3 11:03:39 WARNING[3317]: zt hook failed: Device or resource busy Here is my /etc/zaptel.conf fxoks=1 fxsks=4 loadzone=us defaultzone=us

Dear all I have asterisk with TDM808B FXO port with i call in asterisk and i promt IVR then user dial extention for user then my SIP phone rining but i disconnect or hangup my mobile phone but still SIP phone rining and stop rining after timeout is there any PSTN problme or FXO signalling problme i have configuraed singalling=fxs_ks ----PGP Signature-- Satish

hi community, I'm new to this list & asterisk in general, so let me first say thx to everybody involved in providing such great tools & ressources!! I'm currently trying to implement a simple voicebox-system. for demonstration purposes, I've successfully connected my cellphone via bluetooth using the current chan_cellphone-patch on the current SVN-version of asterisk.