similar to: Extra Sounds Missing on 1.6.1.6 install

Hello everyone, I am using the Asterisk REST API in order to establish a call to an endpoint and to send over a remote file (HTTP). The issue is that I am experiencing an audio quality issue. I have tried encoding the file differently, but everytime Asterisk is cutting the audio frequencies above 4Khz. The call is established with G.722 and the audio file is mono 16Khz 16 bit sln16 extension.

Hi, I follow the web,http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf - mohstream.sh , to configure music on hold to play using mms but failed. Anyone can play using mms? ango

Hi all, I'm new to this list, so forgive me if I'm not supposed to ask this: I currently own a Linksys SPA941 SIP phone with 5.1.8 firmware. Is there any way to use TLS with this phone<--->asterisk (v 1.6.0.9)? It is said that is supports TLS/SRTP but I don't see any of these options in the configuration file or the admin (advanced) SIP conf panel. Am I missing something? Thnx

I'm using a Asterisk 1.6.1.6 with dahdi. We need to redirect phone calls to a certain number when there is nobody. So I read about call reflection but the call reflection applications on bristuff are not for 1.6.1.6. Are there any other applications or patches that provides call reflection for Asterisk 1.6.1.6?? Greetz TM

(Note: This announcement originally went out with an incorrect version number mentioned for the Extra sounds. It should have went out as Extra Sounds 1.4.9 and has been corrected in this announcement. Thank you for your understanding.) The Asterisk development team is pleased to announce the release of Asterisk Core Sounds version 1.4.15, Extra Sounds 1.4.9, and Freeplay Music On Hold sound

Greetings, I'm having a heck of a time with one way audio on a SPA2012. It's public IP connected directly to cable modem. One line configured. Asterisk is multihomed Public IP outside / Private Inside. Extensions inside network are can't hear audio from phone outside connected via the spa-2012. Outside can here audio from inside the network. Ring works both ways. I've

hi, which file extensios are supported in mixmonitor application? https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Application_MixMonitor can i record to Opus? -- --------------------------------------- Marek Cervenka =======================================

[Apr 26 10:47:01] NOTICE[32151]: chan_sip.c:16223 sip_poke_noanswer: Peer '3516533812' is now UNREACHABLE! Last qualify: 86 [Apr 26 10:47:11] NOTICE[32151]: chan_sip.c:12723 handle_response_peerpoke: Peer '3516533812' is now Reachable. (98ms / 2000ms) [Apr 26 12:08:49] WARNING[32273]: app_dial.c:1242 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -

I started to have a problem today that all my calls through voipjet result in just timing out after my assigned timeout period. I tried multiple of their servers with the same problem. Anyone else having a problem? I am running: Asterisk SVN-branch-1.2-r24381M built by root @ asterisk.hulber.com on a i686 running Linux on 2006-05-03 14:14:07 UTC I can connect with other IAX providers.

I want to use a voip speaker phone as a room monitor. Requirements: A phone that I can set to auto answer in speaker mode. A phone with a good speaker phone. Ability to make the audio one way. I want to monitor the room but not have my voice heard in the room. Yes, the mute button can accomplish this also. I have been using the SPA942's around the house (the speaker is just ok but

1.6.1 svn 190575: CC="cc" CXX="g++" LD="" AR="" RANLIB="" CFLAGS="" make -C menuselect CONFIGURE_SILENT="--silent" menuselect make[1]: Entering directory `/home/asterisk/rpmbuild/BUILD/asterisk-1.6.1/menuselect' gcc -m64 -march=native -mtune=native -floop-interchange -floop-strip-mine -floop-block -c -o

Greetings! I'm hoping someone can help me with what should be the most basic of problems. Essentially, I want to have certain calls on an Asterisk 1.2.25 (Yes I know its old, upgrade, etc... its on my roadmap) install go out a couple of analog lines and all other calls go out a PRI. The analog lines are setup in Zaptel group 1 and the PRI channels are in Zaptel group 0. Here is my relevant

Hi all, If I use a SIP softphone and set to gsm codec clearly I get a 8K sample... if I change that to something like opus I get a much better sounding input... How do I get a "better" than 8K sampled input ? I "desire" to have that input be from a pipe. I have cd quality audio in this pipe - I would like to get that audio into asterisk to then send out to endpoints. How do I

Went from 1.6.0.6 to 1.6.0.8 and resulted in segmentation fault. Reverted to 1.6.0.6 and back to normal. ------------------ Linux asterisk.hulber.com 2.6.18-128.1.1.el5 #1 SMP Mon Jan 26 13:58:24 EST 2009 x86_64 x86_64 x86_64 GNU/Linux Apr 3 11:49:56 asterisk kernel: asterisk[3780]: segfault at 00002ce1ac0537a8 rip 0000003e980715a8 rsp 00007fff5bf00c30 error 4 Apr 3 11:50:00 asterisk

Hi, I am using Asterisk 1.4.25. I have one Broadvoice account. I Integrated this broadvoice account with Asterisk Server. I am Able to Make calls but cannot recieve calls. In Incoming calls, call lands to SIP extension, as I attend the call....It gets hungup......... If i dont transfer this call to extension or I play any file then it works OK. But as I transfer it to SIP Extension it get

I just upgraded from 1.6.0.14 to 1.6.1.6 and now my realtime static configuration for extensions.conf will not load. All other realtime configs work (SIP, IAX2, Voicemail). I cannot find any reference or documentation about the structure of the realtime static database for 1.6.1.x but I have used the same table structure since 1.4.x. CREATE TABLE `ast_config` ( `id` int(11) NOT NULL

Hello all, I have a pretty much standard installation of an Asterisk 1.6.1.6 with no PRI cards of any type (full VoIP). Occasionally (it happens every 2 weeks or so), it just stops running. I was using safe_asterisk but it seems that safe_asterisk did not restart it. I do have the core dump file at /tmp/core.myservername-2009-10-20T18:36:20+0200 but it seems it's from an earlier crash. When

Hi all, I have a new installation with asterisk 1.6.1.6 but I'm unable to receive calls from a SIP trunk: [Oct 2 14:30:09] NOTICE[21554]: chan_sip.c:18523 handle_request_invite: Call from 'user001' to extension 'user001' rejected because extension not found. Are there any changes from 1.6.0 to 1.6.1 (or there is a bug)? Below my simple configuration: sip.conf

Hi, In dev-list, some people reported Asterisk 1.6.2-rc2 would suddenly restart. Here, a platform running 1.6.1.6 is also suddenly restarting (once or twice a day with moderate load (40 users)). I don't have much details to report here at the moment. Has someone met something similar ? Thoughts ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Previously, when I wanted to forward to incoming callerid when I forwarded a call to another number I had to set the callerid on the outgoing call to be that of the incoming number. So today I do this: exten => s,n,Set(CALLERID(name)=${CALLERIDNAME}) because I want the outgoing callerid that I forward to not be the normal callerid of the local extension but I want to forward the incoming