similar to: No caller ID, straight to voicemail

Hi, I am currently trying out the asterisk@home (version 1) release of Asterisk, and I want to configure it as follows: Calls from regular telephony network (PSTN) come in through my VoIP provider over SIP and outgoing calls to the PSTN should be routed through the ViOP provider onto the PSTN network. I thus have no direct PSTN connection, but only a SIP connection. Incomming calls

I have just finished implementing an Asterisk system for my place of business (first one), and after three days of flawless usage, Asterisk seems to have crashed. I wasn't running with '-g', so I don't have a core dump. Here's the sequence of events leading up to the crash: 1. call comes in on our TDM2400P 2. all of our phones (about 26 Polycoms) ring. (it's after

Hello, I have problem to make SIP calls, i have asterisk + PC InterP4 + Digium TDM400P here is the content of the sip.conf: [SIP_PROVIDER] type=peer fromuser=testcomclient username=testcomclient secret=testr host=IP_SIP_PROVIDER ;dtmfmode=rfc2833 context=interne canreinvite=no ;allerid=Beer disallow=all allow=ulaw allow=gsm allow=g723.1 ; Asterisk only

hello, Im tryin to make Calls from MS Netmeeting(h323) to Xlite(SIP) it rings, but as soon as i answered it dissconnects!!!! This is what i get from the Asterisk console: -- Executing Dial("OH323/R27469", "SIP/xlite1|10") in new stack Aug 13 10:19:03 DEBUG[524304]: chan_sip.c:1265 create_addr: Setting NAT on RTP to 0 Aug 13 10:19:03 DEBUG[524304]: chan_sip.c:1500 sip_call:

Hello everybody. I have an customer asterisk 1.0.5 running well since 3 monthes, 2 TDM cards 4 FXO, 4 FXS. Since one week, unable to pass call between Zap and Sip getting the "exception on 15, channel 1" The * box is connected to an eads PBX and it seems that failure started when they make some changes on the PBX. Have someone an idea and what is causisng this failure? Here are the

I am hoping some of you can help me troubleshoot this problem I am having with my home asterisk machine. I have incoming POTS service using a SPA-3000 (extension 119). Calls on that line go to an attendant recording that offers a menu choice: press 1 for Nancy, press 2 for the rest of us. In reality, pressing anything other than 1 sends the call to the rest of us by dialing both extensions 101

Hello, I have setup a test server with asterisk/AMP and have several 7960's connected to it. The asterisk server has a public ip and all the 7960's are behind nat'd routers. When I try to call from extension to extension I get directed straight to voicemail. I do not have any cards installed and instead direct everything to an Ondo server. I have been told it's not an AMP

Hello, I'm using Asterisk@Home. I'm still new to Asterisk, and trying to grasp it all. I'm wanting to do a simple setup of One SIP provider (Broadvoice) and 3 SIP Software Phones. I'm able to call the VoIP provided line fine and get dropped to the digital receptionist (or mailbox). However, when I try to send outbound calls I get "Error 503 Service Unavailable" and

Hi, I am having trouble with dropped calls in Asterisk. I've done a bunch of searching but all I could find was setting busydetect and callprogress to yes in zapata.conf to help combat the problem, but I did this to no avail. The following is the output from /var/log/asterisk/full at the time the call was dropped on me. May 13 08:37:13 DEBUG[5379]: Stopping retransmission on

I'm using EyeBeam from xten, and whenever I call another user, the callee phone rings but my SIP phone immediately hangs up. The other end keeps on ringing but when the callee answers, there is no sounds. I have found the "Didn't get frame from channel" error occurring in each such call. What does this mean? How can I fix it? -Mike- May 15 22:31:10 DEBUG[4792]:

I posted a message yesterday that it looked like I was missing cdr sent to mysql. I've done a little more digging and it now appears that calls I initiate via TAPI (asttapi) are ending (when the phone is hungup) with DIALSTATUS=CANCEL May 17 08:59:56 VERBOSE[14874]: -- Lauching Dial(IAX2/sturtevant/18008648331) on IP/200- May 17 08:59:56 VERBOSE[14874]: -- Called

Hey, I've done some searching for this and never really found a concrete answer. Is there a specific reason or solution why just in the middle of a call Asterisk will drop it and I'll get dial tone again? Anyways, this is the output from /var/log/asterisk/full at the time of disconnection: May 13 08:37:13 DEBUG[5379]: Stopping retransmission on

I'm using Asterisk 1.0.4 with AMP and Broad Voice. I have that with only 5 XTen Lite phones. I'm able to call / etc with internal phones just fine. I can call outside Vonage Numbers, and other BroadVoice Numbers. I have vonage where I live (626) and can call that fine. However, other 626 numbers I get similar errors as below. However, everytime, I try to call cell phones, and or

I cant seem to be able to figure this out. As much as I can tell it is a codec problem. I can dial out to 612@fwd.pulver.com and the "Call Me" test there rings my phone. However when the callee endpoint answers, there is a failure to translate: Outgoing Call for 612 612 is not a local user -- Called 612@fwdpulvercom No path to translate from SIP/fwdpulvercom-dd5a(2) to

Hi Guys, I have a really odd one here. We are dropping calls occasionally... there are no error messages being spat out, but I can see this suspicious behavior in the debug logs; Jun 30 14:58:48 DEBUG[19856] pbx.c: Function result is 'Other' Jun 30 14:58:48 DEBUG[19856] pbx.c: Function result is '(null)' Jun 30 14:58:48 DEBUG[19856] pbx.c: Function result is 's' Jun 30

Hi all, I've been running Asterisk with a TDM400P for about 6months, no problems. 2 in/outgoing analog lines, one analog phone. Recently I was messing with the XTEN client, got to finagling with things, and not knowing what was wrong I upgraded from 1.0.7 to 1.0.9 (both asterisk + zaptel). I was testing various things, and found everything worked except outgoing calls. So I checked

Hi! We have a SIP server with a TE410P card with asterisk version Asterisk CVS-D2005.02.12.14.37.11-04/13/05-16:14:03. Sometime the calls get disconnected with now reason and the users get a busy signal. The log file show this for one of the calls that got disconnected: Aug 31 22:51:53 VERBOSE[3911]: -- Accepting call from '46362302' to '36917474' on channel 0/5, span 1 Aug

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 folks, I'm having trouble configuring Asterisk to play an "invalid extension" message to anyone dialing an undefined extension. First I tried using the 'i' pseudo-extension, but it didn't work at all; searching the wiki I found that page: http://www.voip-info.org/tiki-index.php?page=Asterisk%20i%20extension where it basically says that the 'i'

Hi, I have two local SIP extensions (both bt100). One is on remote location behind another nat (16), but everyithing seems to be setup correctly as it can register and is listed as OK(57ms). However I can only call in one direction between those two. Extensions are defined in same context: exten => 11,1,Macro(oneline,SIP/11) exten => 16,1,Macro(oneline,SIP/16) both using same macro