similar to: Hang up error: Didn't get a frame from channel

Dear All, I just setup the Asterisk with E100P which it's no problem in Dial In but I have problem when outdial. The connection method is like this : E1 PRI <-SIGNAL-1-> MaxLink (PBX) <-SIGNAL-2-> E100P <-> Asterisk <--> SIP \-----> Analog PHone Now when I tried to dial out by SIP X-Lite on Windows, it shows me Connect, Trying,

I have a new asterisk install (1.0.7) - and in case it's relevant I'm not using autoload option in modules.conf. Generally all is working well. However, when I make a call from my softphone and try to leave a message, the message is cutoff after a few seconds (whenever I pause for 1 second between words). Strangely, when I use an analog phone connected to my ATA, I can record as long as

My dial plan seems to work great - in that when I call extensions 1234 it connects to 1234. Strangely, after the call terminates (the other side hangs up first), Asterisk continues in the same context and then matches to extensions _. which causes an invalid extension error! Why does asterisk not leave the context (called internalmenu) after the remote hangup? Instead, it continues to the

I am running asterisk 1.2.12.1 JK, Message: 26 Date: Thu, 24 May 2007 21:40:31 -0700 From: JK <jk@bingoconsulting.com> Subject: [asterisk-users] Urgent: DTMF does not work with rtpmap:101 telephone-event/8000 To: asterisk-users@lists.digium.com Message-ID: <465668BF.6080800@bingoconsulting.com> Content-Type: text/plain; charset="iso-8859-1" Hello asterisk-users list. I

[I'm reposting this to the asterisk-users list, since it seems to be a bit more active.] Hello, I started messing with Asterisk few days ago, so my overall knoledge about it is still fairy superficial. I think I found an issue with MP3Player; it can be reproducted with this extension: exten => 6001,1,Answer exten => 6001,2,Background(blahblah) exten => 6001,3,Ringing exten =>

Hi asterisk-users, We are using Asterisk 1.2.12.1, and are trying to use the Page application. It seems to work but after approx 4-5 seconds the call is hung up. The dialplan code look like this: exten => _*2XX,1,AGI(get-paging-devices.agi,${EXTEN:2}) exten => _*2XX,n,GotoIf($[ "${PAGING_DEVICES}" = "invalid" ]?i,1) exten => _*2XX,n,SIPAddHeader(Call-Info:

Hello, We have finally upgraded to Asterisk 1.4, however we've run into two issues that weren't occurring before the upgrade. Issue #1: We're an outgoing call center and need to record all calls. We use the uniqueid field in the CDR to match with the recording, which we labeled with {UNIQUEID} in MixMonitor. For some reason, the uniqueid is not correct in the CDR. Here is the

Hello asterisk-users list. I have been scratching my head for almost a week. We are trying to set a service with a company (ip=XXX.XXX.XXX.XXX) and dtmf is not working. In our scenario the SP is sending call to our ser server and ser is forwarding the call to asterisk. In the asterisk debug I can see the DTMF keys are coming but ivr does not recognice those keys at all. I can see this in the

Hi Guys, I'm writing an app that receives a call on an incoming channel (A), the caller negotiates through a series of prompts and is transferred to an outgoing channel (B) using the Dial cmd. That part works perfectly! For billing I'd like to be able to charge for the time that the first caller is connected to the callee on channel (B) so I can pass on my own outgoing voip costs. How

Hello ! In order to chase after a problem I implemented the following dialplan to have an answertime of exactly one minute: exten => xxxxxxxxxxx,1,NoOp(Test wait) exten => xxxxxxxxxxx,n,Answer exten => xxxxxxxxxxx,n,NoOp(Current timestamp: ${STRFTIME(${EPOCH},,%C%y%m%d%H%M%S)}) exten => xxxxxxxxxxx,n,Wait(60) exten => xxxxxxxxxxx,n,NoOp(Current timestamp:

FYI, asterisk DOES now support g723, but you have to pay for it: http://store.yahoo.com/asteriskpbx/asteriskg729.html -----Original Message----- From: Dan Fernandez <danfernandez00@hotmail.com> Date: Mon, 5 May 2003 17:33:05 -0300 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Has anyone gotten SIP_CODEC= to work? Basically, since I?d like to use g723 for sip

We want some of our users to use G729, and some others to use ULAW. Our PSTN gateway provider supports both, so that's not a problem, and if I force him (the PSTN gateway) to allow G729 only, the outgoing call will take place with G729. The problem is that I want to have my PSTN provider configured to allow ULAW as a first priority, then G729. I did it like that: [mypstngate] type=friend

Well, I'm writing yet another queue_log analyser program in PHP, and I have noticed the following entry in my queue_log file from today: 1104796626|1104796618.532|queue|NONE|ENTERQUEUE||no 1104796664|1104796618.532|queue|NONE|EXITWITHTIMEOUT|1 So, pretty sure that I didn't make someone wait 30 minutes in my queue. extensions.conf snippet: [remote-oldnum] exten => s,1,Answer exten =>

Hello, Hopefully I'm posting to the correct list, but if not, please shun me ;). I'm running Asterisk 1.4, with Festival 1.4.1. I've got a test extension setup, Festival configured and for some reason, when I dial that extension I get this: [Jan 10 17:16:05] WARNING[9082]: app_festival.c:511 festival_exec: Festival returned ER See the full debug below: [Jan 10 17:16:05]

I'm tying to use park and announce for call park on Asterisk 1.4.2 but I'm having trouble getting it to work properly. This use to work with an older version of Asterisk. A telephone on the PSTN calls an IP phone. The IP phone is assigned extension 3-8396. 3-8396 answers the call and attempts to perform a blind transfer to x700, the parking lot number. The transfer gets to Asterisk,

Skipped content of type multipart/alternative-------------- next part -------------- Nov 21 14:17:47 VERBOSE[32580] logger.c: <-- SIP read from 172.25.103.222:5060: INVITE sip:1715;phone-context=exp_net.ascom@ascom.be:5060;maddr=172.25.96.48;transport=udp;user=phone;x-nt-redirect=redirect-server SIP/2.0 From:

Dear All It is a little hard to give a good small example of my question,so I will show the full data on the bottom and the attachment.Maybe some one could tell me an appropriate way to show it.I'm sorry for the inconvenience. Q:How to generate a 53*53 diagonal matrix by my data Some problems confused me are that: 1.Since it is a diagonal matrix,I have tried to transform col1 and col2 to

Hello, I m using asterisk version 1.6.2.0 beta. I m trying to test load on it, for which i m using WINSIP installed at two computers and facing two problems. Problem 1: I got 100 users registered to asterisk from each winsip and then initiates 100 calls from one winsip other winsip. But the problem is approx of 60 calls get mature and asterisk give error for the remaining like shown below.

Hi, sorry for the long debug output below. I configured Asterisk with AMP to send the whole number including the extensions of the callers to the called party. Whatever I configure in AMP it looks like it is used, In my eyes it is ok, but doesn't seem to work. 033811234451 is the call id i configured, and it seems to use them, but the caller will only see a 0338189040 instead of my

Hi! I'm using asterisk 1.4.17 with twinkle and a custom phone based on iaxclient 2.0.2 and I'm struggling a bit with DTMF and features.conf. While the twinkle client is able to initiate an attended transfer using *2 (as configured in features.conf), the iax client is not. I can see the DTMF messages showing up on the asterisk console, but asterisk does not invoke the features