similar to: Asterisk@Home .6 Problems with outbound calls using Broadvoice

Hi people, I have a problem with some extensions. The calls are ended after 31/35 seconds, also, it depends on the number which I call. This is the log, but I've not been able to find something wrong... Any ideas? [Jun 11 15:50:46] DEBUG[26071] app_macro.c: Executed application: ExecIf [Jun 11 15:50:46] VERBOSE[26071] logger.c: -- Executing [s at macro-dialout-trunk:16]

Hello Guys, I'm somewhat of a newbie and am desperately seeking for some help... I've managed to get asterisk up and running on my server, and signed up with a broadvoice account... I'm having no problem dialing and communicating between extensions, but whenever anyone tries to call my broadvoice account, they are greeted by no ring or anything, but rather simply a direct to

HELP! Ok, so I have the following SIP.CONF: [general] context=default port=5060 bindaddr=10.1.1.200 externip = XX.XXX.XX.XX localnet=10.0.0.0/255.0.0.0 disallow=all allow=ulaw allow=g729 allow=g726 allow=alaw register => ##########@sip.broadvoice.com:XXXXXXXXX:##########@sip.broadvoice.com/1234 [sip.broadvoice.com] type=peer host=sip.broadvoice.com fromdomain=sip.broadvoice.com

Hello, My client customers complaining that their calls suddenly get hung-up, I am just investigating if the problem from my side, I had a log of a hang-up case, Does it help to know if there is a problem that can be resolved from my side? elastix*CLI> -- Hungup 'IAX2/99999-6813' == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on

Hi All, I have searched the various post and not able to find the solution. I am using Asterisk 1.4.21.2 for outgoing calls. Earlier i used ZAP trunk and it works fine. Now i need to move to SIP trunk and configured the same. When i try from softphone i got error as "Call rejected" and in the asterisk i got error as

Hi this message give me when I calling a number than actually not busy: "Dial failed due to trunk reporting BUSY - giving up" max channel is unlimited and sometimes it dial number ok but most of the time it gives me this error. Please inform me how can solve this problem. thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi,all when i send a call to other server use SIP trunk, i got this below, <--- SIP read from 222.46.18.52:5060 ---> SIP/2.0 403 Forbidden what's rong with is? > Asterisk 1.4.21.2, Copyright (C) 1999 - 2008 Digium, Inc. and others. > Created by Mark Spencer <markster at digium.com> > Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for

i have a prablom here, i want to send a call from A to B use sip trunk , the call can sended B,but not work to PSTN. the message from B server. help pls,what's rong? > > <--- SIP read from 192.168.0.176:5060 ---> > INVITE sip:15921256331 at 192.168.0.151 <sip%3A15921256331 at 192.168.0.151>SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK51a51b96;rport

I just had a user report that they called out to someone on a cell phone this morning, and after a short conversation, the call was dropped/lost. The person on the cell phone says that this is very rare, and would not suspect the dropped/lost call to be on their side. I have looked at the asterisk/full log as thoroughly as I can, and have pasted the lines which seem relevant to that call below.

I am trying to record calls on demand both inbound and outbound calls. I can record outbound calls just fine but not inbound calls or calls from an internally between extensions. I am using the latest asterisk 1.8.x certified version. On an outbound call I see: == Using SIP RTP CoS mark 5 -- Called SIP/ BVTrunk /7190000000 -- SIP/BVTrunk-00000163 is making progress passing it to

For a new project, I have written a dialplan and it is pretty straight forward: The [dialout] context dials out a number, and h extension in this context writes the CDR. But what is happening is that if the callee hangs up first, all values in the CDR are fine, but if the caller hangs up first, the 'dst' column is always 'h'. I put a NoOp right in the begining of this macro to

I finally got my incoming and outgoing to work on Broadvoice with a configuration file that is no where close to the one given by them. Here it Is (sip.conf). For others who have a working config could u please share so that I could compare. Thank You [broadvoice] type=friend username=[number] fromuser=[number] secret=[password] host=sip.broadvoice.com fromdomain=sip.broadvoice.com

Hi, I have done my best and tired of searching the net about the problem. If anybody could help would be a great favour. Description of Problem ------------------------ I am trying to install two Netpci cards(Traverse Technology Netjet ISDN-s) on Trixbox 2 and aim is to use in Asterisk as dailin and dialout. I compliled the driver as directed in the manufacture manual. After installation dmesg

I'd be grateful if someone could point me in the right direction. I have a Broadvoice trunk attached to Asterisk which I use for frequent calls to the UK using the following in extensions.conf exten => _0[1-68].,1,Ringing exten => _0[1-68].,2,Dial(SIP/BV/01144${EXTEN:1}) exten => _0[1-68].,3,Hangup The caller hears immediate ringing, though it seems that Broadvoice

Is anyone else experiencing 404 errors on outbound dial with Broadvoice? I've followed the instructions posted by Broadvoice to configure sip.conf, and inbound calling works fine. Every time I try to dial out, I get a 404 "Not Found" error. Here are the relevant sections from my configs. sip.conf: context=broadvoice-in register =>

I have tried everything to get BV working outbound. All worked fine until the BV change last week. I called BV and they changed me to sip gen with a new password. I stripped my Asterisk server to one phone on Zap/1 until I get this working. The same BV account works fine with a SPA-3000 so I don't suspect a firewall problem. Symptoms: Asterisk registers with BV Ok Incoming calls work

I'm still not getting my outbound to work. I've seen two patches relevant to broadvoice for chan_sip.c which apparently have already been added to CVS. I'm dropping all outgoing calls after ~30 secs. Asterisk doesn't seem to know they're gone though. I called my cell w/ broadvoice and turned on sip debug AFTer the call had physically dropped: *CLI> sip show registry

Hello, I have a TDM400 and currently have 2 of the ZAP Trunks configured on it. Zap/1-1 and Zap/2-1. I am Running Asterisk@home Version 2.4 with AMP version 1.10.010 In my Outbound Routing I have the Trunk Sequence set up so that 0 is Zap/1-1 and 1 is ZAP/2-1 What I see is that when Trunk Sequence 0 is full, it does not open Trunk Sequence 1. I have found that this is true even if I

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, all. The asterisk setup is working fine, receiving calls via broadvoice "initially". ? When call comes in via broadvoice number, asterisk picks it up and routes correctly, as long as the call came in within ~2 min from the previous one. In other words, as long as a call comes in within ~2 min since the previous one, asterisk will answer the call. However, if the call comes in