similar to: Fax on FXS

I am having a problem with a couple servers. They both have a Digium TE110P and a TDM04B card. I have setup the system so the TE110P uses channels 1-31 and the TDM04B 32-35. The problem is that when we reboot the server sometimes the TDM04B is recognized first and the TE110P second so the configuration fails. I do not know why this happens and to solve this I have to do a "service

I'm trying to get a Qwest PRI configured and working with my lab Asterisk server. They said that the switchtype is 5ess and the signaling is pri_cpe. My entries into zaptel.conf are: span=1,0,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone=us channels=1-23 And my entries in zapata.conf are: language=en context=telco-incoming switchtype=5ess signalling=pri_cpe

Hi I want to submit a bug about auto-dial , but I am not sure on which project the auto-dial comes, how to know about which project , auto-dial comes Thanks Joseph ___________________________________________________________ To help you stay safe and secure online, we've developed the all new Yahoo! Security Centre.

Hi All! I am facing some delay in conferencing. Using DIAX for Voip calls ,no hardware used yet I am using MeetMe to achieve conferencing and am having a lot of delays. Can anyone tell me how to reduce the delay Regards, Amna Saleem -------------- next part -------------- An HTML attachment was scrubbed... URL:

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Hi, Can someone walk me through compiling and loading the Zaptel 1.2.10 driver for Mandriva 2006 kernel 2.6.12-12? When I compile and attempt a modprobe I get "module zaptel not found" Thanks Julian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061109/53aed2ff/attachment.htm

I need to select a line from the Zap group channel using the SIP Phone (not FXO and not FXS ports). ignorepat does not work? Also, what is the method to let the second dial tone has another tone frequency? Regards Bilal ---------------- No, ignorepat is for FXS ports (FXS ports use FXO signaling). Also, ignorepat does not apply to SIP phones, because SIP phones provide their own dialtone,

Nov 12 08:54:27 steerpike kernel: Uhhuh. NMI received for unknown reason a0 on CPU 0. Nov 12 08:54:27 steerpike kernel: You have some hardware problem, likely on the PCI bus. Nov 12 08:54:27 steerpike kernel: Dazed and confused, but trying to continue Would my Digium TDM410P cause an NMI, or is my computer failing? - Mike

I have an Asterisk 1.2.16 server running CentOS 4.4 with a TE110P card and an OpenVox A1200P card. Up to today everything was working perfectly. The OpenVox card has 8 FXS and 2 FXO ports. The two faxo ports are used for a GSM adapter and for an ATA connected to Vonage. The problem we started noticing today was that the Vonage line will receive a call and then cannot connect to any of the SIP

I've got this low-ping 100%-up dsl connection between two asterisk 1.2.7.1 servers. And oftentimes one of them would declare its opposite UNREACHABLE. Why can this happen? The host stanzas in iax.conf have raw IP's, so no DNS monkey business here.. An inquiring mind wants to know.

Hi all, I have a Wildcard TE110P connected to a E1 line an I want to reserve channels in the following way: channels 1-15 and 17-21 for incoming calls channels 22-28 for outgoing calls channels 29-31 for emergency calls My zaptel.conf looks like this: ; incoming group = 1 signalling=pri_cpe context=from-zaptel channel => 1-15 channel => 17-21 ; outgoing group = 2 signalling=pri_cpe

Hi! Anybody familiar with streaming rtp from asterisk. Preferably with the xorcom asterisk patch which streams rtp from asterisk to oreka audio server. Any ideas will do just fine though! Regards / Marcus

Hi list, I'm running current SpanDSP http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.4pre6.tgz with Asterisk 1.2.22 somewhat successfully. Most Fax machines do work but I have problems with people having Tobit FaxWare and Shamrock CapiFax. http://www.tobit.com/login/mrd.asp?CategoryID=120 http://www.shamrock.de/ I've got black bars over the pages. In Tobit some content is Ok,

After an upgrade from asterisk 1.2.13 to 1.2.25 including upgrading libpri, zaptel, the incoming calls to a TDM400P REV I, with 3 FXO modules stop working. The board is working, I tested in another server with the 1.2.13 asterisk version. When a call is incoming, I do a ztmonitor to check the rx and tx values, but nothing appears on screen. Also changed the pci slot where the board is. The

I tried to test Asterisk 1.4 on FC6 x86_64. I have it working on FC5 x86_64 very good, but since FC keeps updating, I tried to follow newer kernel versions. I can't pass the zaptel compilation. Everything is OK, but when I finished, and tried to load it, allways got module not found when I run modprobe zaptel, and modprobe ztdummy. I already tried to modify is with the sed 1 option but

Hi list, This wll be my first post, so I want to thank all the developers for the great product they have created. Now, the question, I have installed asterisk 1.05 on debian sarge (binary package) with an I4l modem and 4 x-lite softphone and 2 SIP hardphones (Yuxin 100) This all works fine, exept for som echo on the ISDN channel, but I'll replace the I4L card with an AVM-C4 card next

Hi all I don't know what went wrong but i no longer seem to be able to compile asterisk. I first do : cd /usr/src/dahdi-linux-2.1.0.4 make clean ; make all ; make install cd /usr/src/dahdi-tools-2.1.0.2 ./configure ; make clean ; make all ; make install ; make config So far.... so good but then when i do : cd /usr/src/asterisk-1.6.0.6 make clean ; ./configure ; make all ; make install i

All, Can anyone point me to the best way to find the mac address of a phone on my system?? I can get the ip's just fine but dont seem to be able to pull mac addresses from any network tools. Thanks-John -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060619/b6e95d07/attachment.htm

Sorry I should have stated that I've tried +x, -x, x.y and x and I still get the same. Regards Lee -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Eric "ManxPower" Wieling Sent: 05 January 2007 08:10 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users]

I think A@Home allows a user to search a directory by either first OR last name, right? I don't know for sure since I don't run A@Home. I would like to offer that functionality in my system - and I'd have done it by now if there was a prompt where Allison asks "press 1 to search by first name, press 2 to search by last name". But I don't think that prompt exists. Can