similar to: Sangoma Question

Is there a way to override sip peers defined in users.conf with respect to their context and hints? Every extension I have defined in users.conf always gets an @default for the hint priority. Below are asterisk outputs and users.conf entries. In peer 1203 I've set a subscribecontext, which is completely ignored. Thanks for any help. nurscarepbx*CLI> core show version Asterisk 1.4.22

Hello, We are having issues with a NEW Sangoma A108D: -- Executing [691918892 at pbx1:1] Dial("SIP/xtravoip200-009d24b0", "DAHDI/g0/691918892|30|m") in new stack [Oct 1 10:04:43] WARNING[14171]: channel.c:3170 ast_request: No translator path exists for channel type DAHDI (native 76) to 256 [Oct 1 10:04:43] WARNING[14171]: app_dial.c:1237 dial_exec_full: Unable to create

Just FYI: IP address 89.248.168.176 has been trying to use the recently release SIP vulnerability in Asterisk to make outbound calls via our box. They are running a bank account callback scam. Jeremy Mann Director of IT Texas Health Management Group Direct Line: 817-310-4956 Main Line: 817-310-4999 Helpdesk: 817-310-4999 x3 Fax: 817-310-4990 Email: jmann at txhmg.com

I'm running some mysql queries on the standard sql logging of calls, and am interested if anyone has any they'd like to share to get good statistics. I'm interested in # of calls per day, based on DST. Number of Calls per day based on SRC, avg duration of calls, etc.. Thanks. Jeremy Mann Director of IT Texas Health Management Group Direct Line: 817-310-4956 Main Line: 817-310-4999

Looking for two things: 1. Anyone that has dialplan logic for an executive assistant. My owners want their extensions to ring on her phone, and be very obvious to her which extension is ringing. They also want her to have presense. She's got Polycom IP 650 with sidecar, they have IP 550 phones. Thusfar I've got her registering to 4 extensions. Each extension is labeled with an

Is the IP 650 sidecar compatible with asterisk? If I pair it with the IP 650 phone, can I have more than 6 "lines" registered w/ the server? Jeremy Mann Director of IT Texas Health Management Group Direct Line: 817-310-4956 Main Line: 817-310-4999 Fax: 817-310-4990 Email: jmann at txhmg.com ________________________________ This e-mail, facsimile, or letter and any files or

Does anyone have a recommendation for a headset that plugs into the Mic/Line-out port on a PC? Ideally something like the Plantronics SupraPlus. I'd prefer Monaural instead of stereo, and cheap in price but not in quality. Thanks for any suggestions... Jeremy Mann Director of IT Texas Health Management Group Direct Line: 817-310-4956 Main Line: 817-310-4999 Fax: 817-310-4990 Email: jmann at

Using the default features.conf setup, if I include parkedcalls in my dialplan, and a call gets parked, and times out, where does the call go? Does it go to a timeout extension in parked calls, or does it go to a timeout extension in the original context? (Using an AEL based dialplan similar to below). -- context internal { ... ... t { jump 600 at

Hi, Does anybody have an idea as to why dahdi_test results drop to unacceptable levels after doing a wanrouter stop/start using a Sangoma card? See below that it drops from 99.99% to 98.55%: [root at bin]# dahdi_test Opened pseudo dahdi interface, measuring accuracy... 99.999512% 99.992874% --- Results after 2 passes --- Best: 100.000 -- Worst: 99.993 -- Average: 99.996193, Difference:

I'm getting the following error over and over on the console: pbx_dundi.c:2975 dundi_rexmit: Max retries exceeded to host Any idea how to troubleshoot this? My network latency is roughly 40-50ms between all hosts in my dundi cloud. Jeremy Mann Director of IT Texas Health Management Group Direct Line: 817-310-4956 Main Line: 817-310-4999 Fax: 817-310-4990 Email: jmann at txhmg.com

Hey Guy, I have ubuntu 10.04 64bit and compiled dahdi / wanpipe 3.5.19 /asterisk-1.8.x I didn't understand what is the relation between wanpipe and dahdi ? do i need to start wanrouter service ? I am getting weird errors and my system got kernel panic many time when i restart dahdi service. any idea ? what is the startup sequence of all these service ? root at example:/etc/asterisk#

Hi, I'm using Sangoma's wanpipe together with dahdi, all software downloaded today at most recent version. Hardware is Sangoma A104, a 4xE1 card. Installation went well. Anyway, wanrouter status shows a different result than dahdi_tool or dahdi_scan. I've just put a hardware loop on port 1. All the other ports are open. wanrouter status shows the expected result: Device name |

I've replaced 2XTE110 with an A102 with echo cancellation specifically to deal with echo problems. However, user feedback has indicated to me that on some calls (not a lot, but some) the call is unusable, with audio artifiacts, described by one user, as: "very bad phasing reverb & feedback (from my rock & roll days)". This is quite intermittent, as in most cases, the user

[priv] type=3Dfriend dbsecret=3Ddundi/secret context=3Dlongdistance Hope this helps, in your case Dundi will save you a world of work on configuring that many systems, in fact if you structure Dundi like spokes around a small number of master servers, the config gets real easy.Let me know how it goes. On Wed, Apr 16, 2008 at 8:41 AM, Jeremy Mann <jmann at txhmg.com> wrote: > > >

[priv] type=friend dbsecret=dundi/secret context=longdistance Hope this helps, in your case Dundi will save you a world of work on configuring that many systems, in fact if you structure Dundi like spokes around a small number of master servers, the config gets real easy.Let me know how it goes. On Wed, Apr 16, 2008 at 8:41 AM, Jeremy Mann <jmann at txhmg.com> wrote: > > > >

I am sure many of you have seen my post asking question that I cannot seem to resolve. While the responses i have been getting have been helpful i still cannot seem to get this working 100%. So I have waving the white flag here. I give up. I need someone to come to my office and help me get this working. If anyone is interested the office is in Lexington KY. If someone is interested we can

Hey Guys! Please help me to find out issue. I have two PRI ## Span 1: WPT1/0 "wanpipe1 card 0" span=1,1,0,esf,b8zs bchan=1-23 hardhdlc=24 echocanceller=mg2,1-23 ## Span 2: WPT1/1 "wanpipe2 card 1" span=2,2,0,esf,b8zs bchan=25-47 hardhdlc=48 echocanceller=mg2,25-47 Sometime my calls got through but some time i am getting pri cause 44 sebpbx1*CLI> == Using SIP RTP

I think you are absolutely right. The audio I heard earlier sounds exactly like a timing issue. So: wanpipe1.conf: TE_CLOCK = NORMAL TE_REF_CLOCK = 0 wanpipe2.conf: TE_CLOCK = MASTER TE_REF_CLOCK = 1 zaptel.conf: span=1,1,0,esf,b8zs span=2,0,0,esf,b8zs I'm going to make this change and reload at lunchtime, I'll document it and post it to the list if it works.

Followup on this issue, it appears that using a single PRI's clock as the master clock avoids clock drift between the PRI's and we get no more artifacts. So, : wanpipe1.conf: TE_CLOCK = NORMAL TE_REF_CLOCK = 0 wanpipe2.conf: TE_CLOCK = MASTER TE_REF_CLOCK = 1 zaptel.conf: span=1,1,0,esf,b8zs span=2,0,0,esf,b8zs -----Original Message----- From: Michael L. Young

Hi Group, I am currently facing a dead end and any help will be much appreciated. I have an a104d installed in an asterisk box, two of which is configured on ISDN pri. One is facing pstn and the other one is facing a hipath 300e Siemens. I am getting one way audio when a local on the hipath tries to make a pstn call but no issue on incoming calls from pstn going to the hipath locals. local