similar to: random one way audio and noise betweenSIP phoneson same LAN

I'm having the same "random" problem. I have "canreinvite=no" on all extensions. I have "qualify => yes" on all non-NAT extensions. I do have several NAT extensions, but I've not had reports of problems from those. 95% of my extensions (all polycom 501/601) are on a brand-new network comprised of 2 48-port Cisco 3560 1GB switches. In all cases, the

We are having an occasional one w-way audio problem that occurs about every 25 - 30 calls on a system configured as follows: Asterisk 1.2.12.1 Sangoma A101 w/wanpipe beta9 Polycom 500s w 1.5.3 This happens only on inbound calls from the PRI. The external caller can hear our customer answer and say hello, however, our customer cannot here their caller. Typically, the caller calls right back

Hi, sometimes I have one way calls and noise between sip phones connected to the same LAN so no nat/firewall is involved. I tried with different sip phone models soft phones and the result is the same. I searched inside every log file but found nothing. I made different PBX with different hardware but the result is always the same. Is there anybody experiencing this terrible problem?

pbx18*CLI> module load codec_g729-ast14-gcc4-glibc-pentium3.so Unable to load module codec_g729-ast14-gcc4-glibc-pentium3.so Command 'module load codec_g729-ast14-gcc4-glibc-pentium3.so ' failed. [Dec 22 15:52:45] WARNING[4491]: loader.c:757 inspect_module: Module 'codec_g729-ast14-gcc4-glibc-pentium3.so' does not provide a license key. [Dec 22 15:52:45] WARNING[4491]:

Hi all, since Asterisk 1.4 seems to have too many differences from previous versions, wouldn't be nice to have a new mailing list? Giorgio Incantalupo

Hi, I'd like to build a personal PBX connecting 4 or 5 analogic phones with a ADSL line and I'd like to know what is the right card I need I visited digium site and I think TDM400 could be the right choice but I cannot understand how it works...I think it has 4 slots where 4 modules (FXS or FXO) can be inserted. How many cards do I need to connect my ADSL line to 5 phones? I think I

Hi, I have an asterisk 1.2.1 on a Debian Sarge distro with *three* TDM400P (12 fxo ports). I noticed Asterisk is slow to answer inbound calls so I connected an analog phone in parallel to make a test: __________Asterisk fxo ---- line -----| -----------------Analog phone The analog phone rings immediately when calling, while asterisk shows the message

Hi, I have a box with Debian Sarge, Asterisk 1.2.1 (and zaptel 1.2.1) and a TDM2400P with echo canceller. I installed the card but no echo cancellation is being made...seems like the echo canceller module does not work, infact the software cancellation is working. My zapata.conf has echocancel = 128 and echocancelwhenbridged = yes but no echotraining parameter which gives a warning. I found

Hi all, I have upgraded my Asterisk box from 1.2.x to 1.4.x version: it seems that sometimes some phones become paused and cannot receive calls anymore. I tried to set autopause = no in every section of my queues.conf but nothing changes.... Anybody knows why a phone becomes paused? Is it an Asterisk 1.4 bug or there is a particular reason for this behaviour? Thank you. Giorgio.

Hi, I'm using Asterisk 1.2.9.1. I have big problem with SIP VoIP providers registrations: Asterisk freezes when it cannot (re-)register with VoIP provider (registration timeout). The problem is related to DNS names resolution: if DNS server is very slow to respond Asterisk stops every activity (no zap or restart commands on CLI). The bad news is VoIP providers usually do not give their IP

Hi, we have an asterisk 1.2.1 box and 2 polycom SIP phones. We'd like to show the called name on the calling polycom display instead of his /her extensions as I do with the caller name on the called polycom. Is it possible? If yes, how? TIA Giorgio Incantalupo

I have been struggling over central provisioning for quite some time. I have eagerly watched each post with like problems but have yet to find a reliable answer. I have a Polycom 501 and I am trying to provision from an FTP server, and just to take any routing out of the issue it is on the same subnet. I am running the 2.0.1.0291 firmware and 3.2.2 bootrom. I set the IP info on the phone and

Hi, every second I get on the console: Remote UNIX connection Remote UNIX disconnected which gives no problem but makes console unusable. Is there anybody who has encountered the same problem? How did you solve it? TIA Giorgio Incantalupo

Hi, I'm using asterisk 1.2.1. Is there anybody out there who knows what this warning means? *WARNING: chan_sip.c:3470 process_sdp: Unknown SDP media type in offer: image 5004 udptl t38* Google does not help at all. TIA Giorgio Incantalupo

This is a function of the Phone itself. Asterisk has nothing to do with it as it does not know anything about the call until after the SIP device 'sends' it. To my knowledge it is not posible. I don't even think a SIP standard is available for this. This 'feature' along with changing CallerID Display after a call has been answered is something missing from the RFC. >

From what I know this log show everything working Perfect. Then it goes to the Welcome screen then after a long time of processing, it errors out with a 0x10000 error Any Ideas? 1005195711|so |4|00|---------- Initial log entry ---------- 1005195711|so |4|00|+++ Note that bootrom log times are in GMT +++ 1005195711|hw |4|00|Initial log entry. 1005195711|wdog |4|00|Initial log entry

Hi, I'm installing an Asterisk box with a TDM2400P in Brazil. I can make analog phones work while lines are not working. Since I do not know anything about brazilian lines, is there anybody who can tell me what is wrong/missing in my conf files (below)? TIA Giorgio _zaptel.conf:_ fxoks=9-16 fxsks=17-24 defaultzone=br loadzone=br* * _zapata.conf:_ context = inbound_zap echocancel = 128

Afternoon, I was hoping someone could point me in the right direction. I have an asterisk PBX deployed in China using a TDM400P based card. The incoming calls are being picked up correctly, but are not being hung up. I suspect that this might be an issue with the signaling that has been selected. If anyone here has deployed asterisk in china using an analog card, it would be a great help

Hi, I am trying to use zaptel module on an Ubuntu 5.10 distro (2.6.x kernel) using gcc 4.0.2. Compilation does not give me errors so after a 'make install' I try to load zaptel module with insmod but the following error arise: *insmod: error inserting '/lib/modules/2.6.12/misc/zaptel.ko': -1 Invalid module format* Is there anybody who can help me?? TIA Giorgio --

Hi, Is it possible for a Manager/Supervisor to intercept and listen in on live calls for training and evaluation purposes? Thanks -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze