similar to: Asterisk room monitor

Before committing to about 50 of the spa942's, I like to take a last poll from those on the list to identify any negative issues that might be associated with the audio, functionality, early failures, etc, on the spa942. Expecting to deploy these using existing cat5 cabling and both rj45 jacks. Been using three of theme in a short term demo with the customer, but the demo systems has

I'm using a bunch of SPA942's, and I'm trying to provision them mostly by DHCP (and what I can't set that way, I try to provision via HTTP interface into the phone). I changed the domain in my AstLinux config from "astlinux" to redfish-solutions.com, and set that in my sip.conf file as well: context=incoming

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

Hello List, This is what I want to do: When a call comes in I want to ring an extension that happens to be loud speaker. The users can the press *8 to answer the call. Is there a SIP device that I can connect to Asterisk as an extension that can accomplish something like this? -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

[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 -

Hi, Did anyone have any experience with CyberData SIP-enabled VoIP Intercom units please? Are they any good? Can you recommend anything better? Thanks, Finku -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090720/b5d2d785/attachment.html

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.

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

It looks like there's a problem with the location or naming of the Extra SLN16 sounds: --14:11:43-- http://downloads.digium.com/pub/telephony/sounds/releases/asterisk-extra-sounds-fr-SLN16-1.4.9.tar.gz Resolving downloads.digium.com... 76.164.171.232 Connecting to downloads.digium.com|76.164.171.232|:80... connected. HTTP request sent, awaiting response... 301 Moved

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

Greetings, I am writing an AGI script that needs to check on the idle/busy status of a number of SIP peers (mostly SPA9xx phones, with a few Polycoms and Snoms thrown in for fun). Is it possible to call Asterisk functions (e.g. SIPPEER) from AGI scripts? Based on my Googling, I would guess in the negative. I have tried various permutations of Set() and Eval() without success. I have also

I am using a cyberdata "sip paging adapter" and with the Dial(MulticastRTP/basic/IP:port) and with tshark I see the RTP data, my device looks like its accepting the data and I hear a click for my relay on my device so it would seem its accepting the call, however - I hear no audio... Asterisk 11.11.0 is what I am using. What might be wrong here? Thanks, jerry -------------- next part

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

Hello all, I am using Linksys SPA942 in my current installation activity. I see a peculiar behavior: A call is made and the SPA942 uses its speaker. When the far end of a call hangs up , the SPA942 stays off hook, and after a time plays a fast busy. The user then has to press the line presence button to hang up the phone. I think I must be missing some sip.conf parameter. My sip.conf is pretty

Hi All; Any one used Digium IP Phones D40? I need to know if they are stable with good voice quality? Comparing to Polycom 330, which is better? Let us talk frankly although I know that we have to support Digium. Regards Bilal

I came up with this a few days ago, mostly used the wiki examples, didn't have time to post on the wiki yet, maybe one of you guys with a few minutes can throw it up there, really, I forgot my logon. http://www.voip-info.org/wiki/index.php?page=Asterisk+Paging+and+Intercom The agi script didn't work for me, wouldn't call the active hint extensions, even though they were there, no