similar to: Delay before audio starts

Hello, I 'm looking for a way to pass the '302 moved temporarily' received from the SIP device back to the SIP provider. Here is the setup: Some SIP phones are connected to an Asterisk System version 1.8. External connection to the public network is also done via SIP to a VoIP provider. Phone A has a CFW all calls to a phone number in public network (Mobile Phone) incoming call to

This patch offers interface to get the library name which software uses. Function: char *vorbis_version_string(void); PATCH (for libvorbis-1.2.1RC2): diff -crN libvorbis-1.2.1RC2/include/vorbis/codec.h libvorbis-1.2.1RC2_NI/include/vorbis/codec.h *** libvorbis-1.2.1RC2/include/vorbis/codec.h Mon Aug 25 05:57:44 2008 --- libvorbis-1.2.1RC2_NI/include/vorbis/codec.h Sat Sep 13 05:00:22 2008

Hi, I have trouble getting MOH to work after an upgrade from asterisk 1.4 to 1.6.1.4. The call goes on hold, MOH is started, and then stops right away. Here are the files both of type .raw: Tsunami*CLI> moh show files Class: default File: /etc/asterisk/musiconhold/Fr?d?ric Chopin - Polonaises Op. 40-2 File: /etc/asterisk/musiconhold/Fr?d?ric Chopin - Polonaises Op. 40-1 These files

Hi all, I’m trying to rewrite Diversion header when call forwarding is done on the phone. The phone sends "302 Moved Temporarily" response and sets Diversion header to a local number, but before Asterisk sends this call towards TSP provider I need to change Diversion header to a full PSTN number. I am using PJSIP_HEADER in a pre-dial handler (configuration is below). On the same

Hi everyone, I'm having a bit of an issue after upgrading from asterisk ~1.2.24 to 1.6.2.11, with the old version when the user would go to transfer a call, they would press Transfer, then the speed dial button for the extension, optionally introduce the call, and then press Transfer again to complete the transfer. Now, with the new version, when you hit the speed dial button, asterisk

Good evening, I just signed up with Nufone and I am able to receive calls with no problem via my 800 number. Outgoing calls are not going through though. My extensions.conf is as follows: [nufone-out] exten => _91NXXNXXXXXX,1,SetCallerID(mynumber) exten => _91NXXNXXXXXX,2,Dial(IAX2/user:pass@switch-2.nufone.net/${EXTEN:1}) exten => _91NXXNXXXXXX,3,Congestion Whenever I try to

Greetings, I have a very basic equal-weight ring-all queue set up in queues.conf: [sales-queue] ;music = default strategy = ringall periodic-announce-frequency = 20 announce-holdtime = no timeout = 15 maxlen = 0 member => SIP/1xxxxxxxxxx at junction_networks,1 member => SIP/1xxxxxxxxxx at junction_networks,1 member => SIP/dude,1 member => SIP/homie,1 member => SIP/fellow,1 But

I need some extensions logic assistance, I'm trying to dial out one of multiple SIP trunks, in sequence. I need to detect a busy SIP trunk(I only allow 1 call per trunk) and roll over to a second or third depending on that busy status Here's what I've got for a macro thusfar, but it's not working(fails if the 1st trunk is busy) extensions.conf: [globals] trunk_1 => SIP/trunk1

Hello everyone, my name is Miles, I am fairly new to asterisk. I have recently begun to learn asterisk and I have a couple of questions. 1. After installing asterisk using the following instructions; a. sudo mkdir /usr/src/asterisk && cd /usr/src/asterisk b. sudo wget http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1 1.2.1.tar.gz c. sudo tar

I am currently running two different versions of Asterisk 11.0.1 11.2.1 I have noticed the bug occur on both servers. The issue is that when I try to dial a phone number sometimes the call will never go out. I will check the Asterisk server with NGREP and see that the SIP messages are making it to Asterisk but Asterisk isn't responding. I do the following command "netstat -nap |grep

Hi, when I try to install Asterisk 11.2.1 the console return error which it tells: /usr/bin/ld: final link failed: No space left on device and the process exits installation. How can I solve this problem? Tmp folder is empty..... Thanks,Jordi -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have a weird problem with call using my T1 card. I can make calls fine using my analog and IP phones, but when I try to initiate a call using a .call file, I get the following error -- Attempting call on DAHDI/g1/1XXXXXXXXXX for s at test:1 (Retry 1) -- Requested transfer capability: 0x00 - SPEECH -- PROGRESS with cause code 127 received it happens on certain numbers I dial, but if I

Hi, I have the following setup: Ubuntu 12.04.02 LTS (64 bit) Asterisk 11.2.1 Sangoma 4-Port-Card (A104d) with firmware 43 (german e1-ports connected) WANPIPE Release: 3.5.28 DAHDI Version: 2.6.1 Echo Canceller: HWEC libpri version: 1.4.12 I call via sip into the dialplan. Then I do a "Dial(DAHDI/g1/voicenumber,r)". The call is bridged and everything is fine. "dahdi show

Hello guys... I compiled inside of "mock" libdrm 2.67 mesa 11.2.1 ati 7.6.1 mockbuild says: libdrm x86_64 2.4.67-1.20160218gitadd8936.el7.centos local-drivers - OK mesa-libGL-devel x86_64 11.1.2-1.20160210.el7.centos local-mesa - OK installed packaged: xorg-x11-drv-ati.x86_64 7.6.1-1.el7.centos @local-drivers OK libdrm.x86_64 2.4.67-1.20160218gitadd8936.el7.centos OK

I am planning on dabbling with some VOIP providers. I was thinking of Teliax first. My thinking is that the first LD call would go to teliax and the second (etc.) calls would go out to the PSTN. I could then verify bandwidth and quality to decide when to add more trunks and to Internet connections. I have been doing some concept testing with FWD for toll free calls, but I am using 393 as a

via google, I found the reference regarding digit maps for the Polycom phones: http://lists.digium.com/pipermail/asterisk-users/2005-January/082884.html But I don't see any instruction for prepending digits to the number dialed. Does anyone know how to prepend a digit to the number dialed (from the Polycom side, not Asterisk)? I can do this pretty easily on a Sipura. i.e. Say I want to

When I make a regular call from my SIP phone connected to my Asterisk server I have no issues, however when I make a call using Originate : 'Channel'=>"SIP/1XXXXXXXXXX@sip.broadvoice.com", 'Context'=>'mycontext', 'Exten'=>'899', 'Priority'=>1, 'Callerid'=>'whatever')); It creates a screech sound when the

I configured this once now I forgot what I did. Two Broadvoice accounts. Incoming is simple - just use the phone numbers. Outgoing: Dial out on a specific line and/or set up the groups and select the other "line" if the first one is busy? -- James Taylor MetroTel 3505 Summerihll Road Suite 11 Texarkana, Texas 75503 903-793-1956

Hi List, I am having trouble connecting two * boxes together via SIP. It looks like * is authenticating the hostname, not the username. The Sip.conf looks fine on both sides, but I get: Jun 2 14:44:52 WARNING[2407]: chan_sip.c:6829 handle_response: Forbidden - wrong password on authentication for INVITE to '"+1XXXXXXXXXX" <sip:+1XXXXXXXXXX@206.80.70.56>;tag=as562b672b'

Hey all I have been beating on this all weekend long. Any feed back would be appreciated. We stood up a 11.6 system. We tested everything we could think of. We moved over to it and all seemed to be working good than a customer told us that they were not hearing our vociemail greetings. When we call into the system and it drops to voicemail we just get a beep no greeting played. We checked