similar to: Park() ignores 'r' option which should disable music on hold in favour of ringing tone

Hi everyone, I hope someone can help me with a problem I'm having with Cisco 7940 phones on the SIP 8.12 image. When I place a call from one of the handsets, the call proceeds as normal for 20 seconds and is then terminated by Asterisk (1.4.26.2): [Oct 3 10:08:55] WARNING[1650]: chan_sip.c:1981 retrans_pkt: Maximum retries exceeded on transmission 00215553-

Dear, I have asterisk 1.6.2.13 in a small hardware PBX (512 GB RAM and Celeron), and last days when I call from one extension to another of the same PBX after I dial the number the rings sound after 20 seconds. In the CLI log, when I debug the AGI, I see always goes good until dialparties.agi, and after that there are 20 seconds without any log, and so the ring sound. I've read

Hello: We have a TDM2433E Digium Card (12 FXS, 12 FXO) and Asterisk 1.8.7.0 running. Everything seems to be ok but call transfers. This is the issue: *A, B, C and D are in FXS ports*. 1) A calls B. B anwers. 2) B tries to transfer the call to C dialing *2 (code for attended transfer). 3) A hears MOH. B dials number C. 4) Asterisk says the dialed number is incorrect or non existing. We tried

Hello, For those customers with only analog lines, who ask for fax2email and email2fax, whats the most reliable solution available and tested with Asterisk? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120529/3e28b56e/attachment.htm>

Hi All; If I set a context other than the default context, then I do not see a generation for a configuration in the extensions_additional.conf for this context, but always the generation for the configuration is for the default context (from-internal). Normally, I have to put some Phones in a context and another Phones in a context, and give each context a privilages, but if I do this, then I

Hey Guys! I am getting this wired error when i am calling IAX trunk. Everything works! but i want to get rid on these RED WARNING messages.. what is wrong here ? I have func_aes.so module loaded. also i remove and test but still same error. -Satish == Using SIP RTP CoS mark 5 -- Executing [7623 at from-sip:1] Macro("SIP/7527-0000000d", "orasebcamdial,7623") in

Hi, I've been meaning to fix my non-working paging feature here for a while, and I've just spent the last 5 hours looking at many, many web pages that all say the same thing. I am using Asterisk 1.6.2.18 and Polycom phones, both older (501 with "latest" legacy 3.1.7 firmware) and newer (335 and 650 with latest 3.3.1f). I have changed the correct values in sip.cfg like

I'm trying to decide if I need to open an issue for this or if it's just a misconfiguration issue of some sort. Here's the situation - yesterday morning, I downloaded asterisk 1.8.19.1 and installed it on a fresh CentOS 5.8 installation and got a shell of a basic asterisk install setup (minimum required configuration files, etc, with no dialplan or sip peers setup yet). In the

Hi All; When the call coming via the E1 dahdi and I handle the call (as first step) by exten => 5631040,1,Goto(OrangeCMG,s,1) it is not working and the call will be disconnected instead of queued. But, when I handle the call (as first step) by playing any sound file and then send for the queue, then it is working fine, WHY? exten => 5631040,1,Playback(WelcomeMessage) exten =>

someone have been installed Asterisk (Trixbox) on VirtualBox which is installed on a Linux host (Ubuntu server 10.04 specifically). I want to know if it is convenient or not, and the reaseons if i should on shouldn't do it. Thanks in advance.! -- Esteban L. Cacavelos de Amoriza Cel: 0981 220 429 -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I'm getting one way audio when calling over the SIP trunk i.e. end device B (remote end of SIP trunk) can hear device A (softphone registered with Asterisk) but device A can't hear device B. Even though I configured same NAT configurations on other servers and they are working good. The NAT configuration is listed below; localnet=130.0.0.0/130.0.0.0 externhost=12.131.12.13

Hi All; Again, the Cisco IP Phones 7942G and using Skinny: I upgraded the firmware to version 8.5 (skinny) and I am using skinny channel (chan_skinny) and the skinny.conf file. The phones are registering, but when we use them to place a call, we only hear tooooooooooo in the handset and we do not hear voice (even when we dial the digits, we only hear toooooooo .. but it dials and destination

NEWBIE alert: i'm a linux person, not an asterisk person so i'm certainly capable of handling any linux-flavoured solution you can suggest. here's a note i got from a local company i know (some proper names removed): ===== start ===== Now and again our ISP goes down and when it does give us a hicup, the Asterisk system shuts down (not very forgiving). When it shuts down our phone

Hi, The call waiting tone is very annoying (you hear nothing while it plays the beep). I need callwaiting because of the queues (the phone has to ring as soon as you hangup) but I want to remove the beep on my dahdi channels, how can I do? Thanks, Niccol? -- http://www.linuxsystems.it

Hi All; Using the FreePBX, after I added the extension from the GUI, I discover that it is automatically added in the extensions_additional.conf in the context [ext-local] and [from-did-direct-ivr] How I can change these context name? I need to determine this. How? Regards Bilal

Hi ALL, still having trouble getting Ast 12 to run. I got it compiled and built but now when I try to run, I'm getting a missing library error that seems to be in error (see below). The .so file DOES exist with correct permissions. [root at Asterisk12 ~]# asterisk -rvvv asterisk: error while loading shared libraries: libasteriskssl.so.1: cannot open shared object file: No such file or

Hello everyone, How can I let Digium Wildcard TDM800P work successfully with DAHDI? Because the Centos recognizes the card but I can't get the analog card working with DAHDI. Thanks in advance, Gilberto -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130309/d82ddd47/attachment.htm>

I have an Asterisk 1.4 box which is sometimes getting the message below. Here is the weird part, the CNG is coming from ANOTHER ASTERISK SERVER. 209.220.119.19 is an Asterisk 11 box. [Oct 8 11:59:27] NOTICE[20798]: rtp.c:849 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 209.220.119.19

I need to disable/enable a peer after hours automatically, and am thinking about doing so via the AMI. Is there a command to enable/disable (or perhaps delete/add) a peer via the AMI? I could create code to modify sip.conf and force a reload, but that seems like the wrong approach... -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I have an internal extension, e.g. 1005 which is being called from an external/public number like 123456789. Now when it comes to the spoken voicemail information it says something like "number 1000 not available", however it should say "number 123456789 not available". How can I configure this? I already googled and I guess this is really easy, but I just couldn't