similar to: (still) Channel problem - solved partially

Hi folks, I'm still having the following problem, maybe someone can help me out of it. Two IDENTICAL MACHINES (same motherboard, same RH 7.2, same *) communicate through IAX2. Everything works ok on machine 1. On machine 2, if I try to use 4 fxo's from a TDM400 card, sound gets lousy. If I manually destroy one of the zap channels (e.g. zap destroy channel 4), sound gets good again.

Hi, I'm trying to build a call center application that allows attendants to come in the morning and dial a certain extension to make their extension available. I wouldn't like to use the AgentLogin app because their line would need to stay off-hook (is this correct?) Is there any SET channel status command that would allow me to do something like this? PauloHM -------------- next

Hi all, In a forthcommming project, I'll have one * server tentatively calling 10 PSTN numbers through IAX2 and an * gateway. Can someone tell me if bandwidth is being used for each of these calls/channels even while my gateway tries to call and connect the destination numbers? Best, PauloHM

Hi, I'm trying to use iaxcomm. I can place a call from the softphone, but when I place a call to it, when I answer I get ... NOTICE[16401]: File channel.c, Line 1094 (ast_read): Dropping incompatible voice frame on IAX2[paulohm]/3 of format GSM since our native format has changed to ALAW My iax.conf looks like this .. [paulohm] type=friend host=dynamic username=... secret=...

Hi all, We are about to make our first channel bank install. This will be a one PRI outside connection and up to 70 extensions. As the schedule (and the budget) is pretty tight, I would like to learn a little bit more about general experiences with channel banks, like echo cancellation problems, Caller ID usage, etc. TIA, Paulohm

Talking to myself ... ;-) Solved this by ... disallow=all allow=gsm ;allow=ulaw ;allow=alaw -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Paulo Mannheimer Sent: quinta-feira, 11 de dezembro de 2003 09:02 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Iax, Iax2 and Iaxcomm Hi, I'm trying

Count me in too. -----Original Message----- From: sip [mailto:sip@intology.com] Sent: Friday, October 17, 2003 1:56 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Beta testers for visual configuration tool for asterisk count me in ----- Original Message ----- From: "Paulo Mannheimer" <paulohm@instant.com.br> To: <asterisk-users@lists.digium.com>

Hi, I'm having random hangup problems with zap channels. If I turn busydetect off in Zapata.conf, * fails completely to detect a user hangup in the middle of a script. On the other hand, if I turn it on, everything works much better, but long calls tend to be hung up without a motive. Any other parameter that I can try? Any #define that I can tweak and recompile? Will callprogress

Hi All, We've been developing for a while an IDE for Asterisk, and the time has come to open it for beta testers. You can check at www.instant.com.br/viv.html for a snapshot of the application. Current modules are Dialplan and VoiceMail configuration. As you may see, it is all-visual, with drag and drop support and integrated sound recording, saving and cross-checking, so you dialpland

Hi All, Can anyone tell me what are the <dev1> <dev2> parameters that I should use to run pridump? I took a look at the source code but couldn't figure this one out. Best, PauloHM

Hi, I would like * to treat a SIP extension as a normal extension, when it comes to the busy functionality. In other words, if someone tries to call the SIP phone and there is already an ongoing conversation, the new caller should get a busy message/tone Is there any parameter that I can set? Is this something that should be configured at my softphone? Best, PHM

Hi, I'm testing an E1 with E&M signaling. Some of the problems I'm running into are the following: 1) if I try to configure any channel above channel 15, I start getting a "multiframe alignment error" on my telco test equipment. So I have my zaptel file only configured for 15 channels, like this span=1,1,0,cas,hdb3 e&m=1-15 2) When the test equipment tries to send me

Hello, In my quest to figure out the source of the random echo on our shiny new asterisk install, I have been using ztmonitor on the TDM400p channels for the good part of today. I have 2 TDM400p cards with 8 FXO modules and 6 pots lines connected to them (last 2 channels are unused but configured in zaptel). Even when the lines are onhook, the Tx values settle down to 0 but the Rx values still

Hi, we are setting a 120-channel conference server and would like to learn if someone already did this (hardware, problems, etc...) Best regards, PauloHM

Hi, I'm installing my first E100P. My zaptel reads the following: Span=1,0,0,ccs,hdb3,crc4 E&m=1-31 My Zapata.conf reads the following: Signaling = em_w Channel =1-15 Channel =16-31 After starting the zapter service I get: ZT_SPANCONFIG failed on span 1: No such device or address (6) ??? PauloHM -------------- next part -------------- An HTML attachment was scrubbed...

Hi All, Does anyone have any experience with the ArrayCom VoIP phone? I bought one a couple of weeks ago, it used to work quite well with * until I misconfigured one option. I now cannot make it work anymore, because the phone boots up, doesn't find a valid SIP gateway, resets itself and keeps rebooting indefinetely ;-( Their technical support refuses to answer my questions. Any hint on a

Hi all, I have a problem running bind on C5.4 when the messagebus service is activated. When I switch off the dbus message service bind runs well. With dbus activated bind crashes with a segfault. How can I get rid of this error? What shall I change? SElinux (the source of most unexpedted things) is switched off. Following versions are installed: bind-9.3.6-4.P1.el5

Shouldn't that acutally pull the value of EscapeChar out of the users environment (and possibly default to the old message if they don't have one defined at all)? Thanks, --Jason Lacoss-Arnold, Systems Technical Specialist Technical Services - Unix Arch. 314-955-8501 -----Original Message----- From: bugzilla-daemon at mindrot.org [mailto:bugzilla-daemon at mindrot.org] Sent: Thursday,

I've gotten around to testing the generator hang bug using stunnel under 2.6.7, and I still occasionally get the same problem. It seems to be around 50% of the time. Here's some strace output from the generator and receiver from two runs, one where the generator exits properly, and one where it doesn't. Run 2 hangs indefinately. Run 1 Generator: Process 14581 attached - interrupt

Hi All, I was able to track down what I believe is a bug when using AGI services. This bug may crash your system if your extensions.conf script is intensive in using AGI services. Depending on your system's ulimit, * keeps opening files until it reaches the system limit and then stops responding. Function app_agi/launch_script seems to leave an open and unused file. Can someone confirm this?