similar to: new on E100P

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, 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

Hello I have installed my new E100P card in the same machin with two TDM04B cards. I have followthe instructions on http://www.digium.com/index.php?menu=configuration#T_E100P_PRI This is my zapata.conf span=1,0,0,cas,ami fxsls=1-15,17-31 dchan=16 fxsks=32-39 loadzone = us defaultzone=us I do *modprobe zaptel wct1xxp wcfxs ans seems to work ok. * But when i do ztcfg -vvv i got Zaptel

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 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

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

Hey all, ? I'm trying to get an E400P and 2 X100Ps working together in the one box and don't seem to be having much luck. ? I can get the two different types of boards working separately, but not together.? I've made calls on both the X100P and have seen sync and correct signalling on the E400P.? But when I try to enable to configs together I get the following: ? modprobe wcfxo

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

Hi, I have been using asterisk for a couple of months now and for thee most part, I love it. However, I'm having a problem with the drivers of the Digium TE110P. I have tried both the Debian package and the CVS. I have tried several kernels, and am now at 2.6.11. This has been working before (with 2.6.8.1), but after a reboot it stopped working and I am not able to consistently make

Hello, I have a E100P card from digium and I try to implement a conference bridge in asterisk. I wonder since I got the E100P card do I still need to load ztdummy for caller from h323 endpoints to work with Meetme? I load the E100P driver but i did not load the ztdummy driver. My h323 caller does not hear any voice play by Meetme. Looks like ztdummy is required as long as h323 is concern and

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

<Correction in my zapata.conf file I used> Hey Everyone, So this is a problem I've been having for sometime now. I sent a few messages to the list with no luck. The problem is that when people dial into the Asterisk system using DID numbers, it works the first time or 2, then I get busy signals. A friend recommended I clear out the zapata and zaptel, start over, and recreate my

Hi All- I'm working on a project that will have remote (internet)access to an * server through SIP phones, either soft or hard ones. Does anyone have any experience to share about which SIP product they are using under similar conditions, as well as which codec is being used and bandwidth usage? TIA! PauloHM

I'm trying to load ztdummy on my Asterisk, running in a XEN domain. I've modified the code to disable the use of an RTC. I can load the zaptel module just fine, the ztdummy also loads without problem. But when running ztcfg I get this error. ----- s n i p ----- graham:~# ztcfg -vvvv Zaptel Version: 1.4.6 Echo Canceller: MG2 Configuration ====================== SPAN 1: CCS/ AMI

Hi List, I'm setting up a system with one TDM400P (2*FXO + 2 * FXS) and one Junghanns QuadBRI on a Fedora Core 6 (Kernel 2.6.20-1.2944.fc6). I'm using the bristuff-0.3.0-PRE-1y-e kit. It download zaptel-1.2.16, libpri-1.2.4 and asterisk-1.2.17 When it's the time for ztcfg to do its job it complains with <ZT_SPANCONFIG failed on span 2: No such device or address (6)>

Hi, I?m having a serious problem at one customer. After 6 hours answering a PRI line, * stops responding in a very similar situation as described here ... http://lists.digium.com/pipermail/asterisk-users/2003-July/015391.html I took a look at "/proc/first * PID/fd" and there are hundreds of file descriptors; If I try to connect using asterisk -r I get the "broken pipe"

Hi folks, where can I find the R2 beta code for Asterisk? Best, PauloHM -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030806/9c7a0660/attachment.htm