similar to: Zaptel Not Releasing Channel (PRI)

I have a user who's system crashes on pri hangup request. Tried 1.4.19.1 and 1.4.20 as well as the latest libpri no change Progress is as follows...... < Supervisory frame: < SAPI: 00 C/R: 0 EA: 0 < TEI: 000 EA: 1 < Zero: 0 S: 0 01: 1 [ RR (receive ready) ] < N(R): 025 P/F: 1 < 0 bytes of data -- ACKing all packets from 24 to (but not including) 25 -- Since

Hi all, I was looking to build a SIP-to-PRI gateway using Asterisk (as in my other post), but there is also an alternative of using a Cisco router with something like an NM-HDV module with a T1 VIC module and DSP channel banks. The question is, would it be more reliable to offload all dahdi/zaptel/libpri type stuff to a dedicated gateway device (Asterisk or Cisco) and have the Asterisk PBX only

Hi, Quick question to the real world. Approx what specs would I need on server to handle 95 ZAP or Dahdi -> SIP gateway using G729 on the SIP to carrier side (nothing else, just media conversion)? Does the latest Asterisk/DAHDI significantly improve these numbers over say, Asterisk 1.2.X? Sure, there is plenty to read but nothing I could find quickly on my exact needs that was clear and I

Great for Chan_Mobile and GSM modem for SMS in Kannel or if Asterisk supports SMS over GSM modem. I know chan_mobile had SMS in the future at one point but have not revisited the project since. "America Movil's MVNO TracFone Wireless quietly unveiled a prepaid, nationwide unlimited offering for $45 per month that includes unlimited text messaging and 30 MB of data."

Hi there, I'm having some fairly serious asterisk problems, which seem to be spread quite liberally across all asterisk versions, I've tried 1.4.2, 1.6.0.10, 1.6.2beta4 and still had exactly the same problem with static and echo on the line when using the PRI interface. A little background: Server: HP DL145 G3 Dual Opteron 246 with 2GB ram and a brand new OpenVox D110P

Hi I have problem with TE121 Digium card. I connected it to modem keymile Music 200 (provided by telco) but I can see 2 red lights on modem (both bellow words rx) and my card is red too. I tried to make experiment and made loopback (pins 1 4 , 2 5) and put it in card and card become green (I hope that is sign card is ok) but on CLI i can see following error message WARNING: We think we are cpe but

Also posting this question to people working on manager interface and dialers. I have a simple auto dialing script (using Originate) that forwards all incoming calls to a queue full of waiting agents instead of a meetme conference room. I use queues rather than meetme so I can leave the automatic call distribution to the queue itself. The problem is when the calls reach the agents, some of the

Hi all, We have just set up trixbox latest with a Rhino r1t1 card, hooked up to a plain E1 PRI line. We call fine SIP to SIP, but as soon as we make a call from SIP to PSTN all sounds become unintelligible screeching or static kind of noise on both ends, when we call PSTN to SIP the PSTN side seemingly OK at least we hear no screeching sound, but the SIP side is a even worse screeching

Hello, Raw stats: Version:1.8.3.2 OS:Centos 5.6 Special setup: postgre database I am having a few queue issues with Asterisk specifically relating to breaking out from queues while on hold. The intent is that while someone is on hold they can press a key (lets say *) to break from the queue and go elsewhere (in this case to leave a message). However In all of my testing I am unable to get

I have 12 T1's going into 3 servers, 4 in each into "Digium, Inc. Wildcard TE410P Quad-Span togglable E1/T1/J1 card 3.3v (rev 02)" cards. Each "group" of T1's have the primary D on 24 and the secondary D on 96. The first server (ts20) and the last server (ts22) can playback "demo-congrats" fine. The "middle" server (ts21) cannot -- just dead air.

Hi, I am new user on asterisk (for that matter linux) and i have lot of embedded programming experience. We have a new project from our client, to design a box that takes the telelphone line as input and route the line to the respective user with different ring tones. The box should be programmed by the users with buttons. Features. 1. I should be able to store some .wav files for different

A) Modify res_musiconhold.c and the application "WaitMusicOnHold" to accept DTMF breakout B) Create a call queue with a timeout of X and configure the DTMF options properly. Then you can drop callers into this queue and effect a music on hold for X seconds and allow DTMF breakout with no C code. -----Original Message----- From: asterisk@billheckel.com [mailto:asterisk@billheckel.com]

People, When I did: rsync -av /home/ /mntb5/ # about 4TB I got errors like: 'rsync [sender] expand file_list pointer array to xxx bytes, "did move"' with rsync hanging - after breakout on /home for writing I then get: "Read-only file system" So after unmounting and remounting /home I did: cd /home find /home/ -type d | sort >

Slightly OT, but I'm wondering if anyone here has come across a "soft ATA". That is, software that will perform the functions of a basic POTS line ATA on Linux with a zaptel driven card. I have a Linux machine with a zaptel card in it and I want to have another Linux machine running Asterisk utilize the zaptel card in the first Linux machine to make outgoing and receive

Just a reminder...The submission deadline is in one week. thanks, -KVM Forum 2010 Program Commitee -- ================================================================= CALL FOR PAPERS KVM Forum 2010 ================================================================= DESCRIPTION The KVM Forum is back! After a break last year we're proud to present this year's gathering around KVM

Just a reminder...The submission deadline is in one week. thanks, -KVM Forum 2010 Program Commitee -- ================================================================= CALL FOR PAPERS KVM Forum 2010 ================================================================= DESCRIPTION The KVM Forum is back! After a break last year we're proud to present this year's gathering around KVM

Could someone explain to a non-US dummy the following phrases I have seen on the list. "I can provide you with tier 1 termination 6/6. I can blend or NPANXX breakout." "We provide US48 termination, blended rate for 1 MOU and above is .008 with 6/6." What is 6/6? What is US48? What is blended? What is MOU? What is NPANXX breakout? -------------- next part --------------

Groups for implementing =91GSM Gateways=92" which leads me to believe (or h= ope at least) that more than one phone can be paired to a dongle. Dongles are so cheap I guess it doesn't really matter other than more complexity. Anyone know for sure? --=20 Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) --001636c92545598447046efba265 Content-Type:

Greetings List, I have configured chan_mob for Nokia 7610. I can succefully dial from softphone to mobile and land line numbers, Softphone (PC) =====> Asterisk ====> FXO (Nokia 7610) ====> Destination Number When call is established I have to use Nokia 7610 for conversation. Is it possible to use softphone, dial via mobile phone and have conversation using softphone?

Ok Guys, I've done a tonne of hunting around on this problem, but can't find much help. I'm running: asterisk 1.4.19.1 libpri 1.4.3 and zaptel 1.4.9.2 which I believe has been modified by RedFone to add the ztd-ethmf module. My interface is a RedFone foneBridge2 4 Span; and I'm connecting to a BT E1 PRI / ISDN30e with 15 lines on span 1, and a legacy Panasonic PBX on span 4. Upon