search for: channles

Good day all We got a Wildcard TE110P I installed linux,zaptel,libpti and asterisk I coped over my zaptel.conf and zapata.conf from a previous E100P config But when I try to start asterisk it gives error not bying able to load zap channles: == Parsing '/etc/asterisk/zapata.conf': Found Jan 10 08:17:18 WARNING[-1084595552]: chan_zap.c:9308 setup_zap: Ignoring switchtype Jan 10 08:17:18 ERROR[-1084595552]: chan_zap.c:9131 setup_zap: Unknown signalling method 'pri_cpe' Jan 10 08:17:18 ERROR[-1084595552]: chan_zap.c:878...

...hunt but just CallForwarding No Answer/Busy, what that means is that if I have asterisk setup to first ring a phone for 5 times and then go to an IVR and answer the phohe, it will go to the next line and stop ringing the first line, and therefore never end up in Voicemail or my IVR. 2. No CPC, hung channles, blank voicemails, and all the other goodies that come with no hangup supervison, is a daily thing. Anybody else seen this? TIA

Hi all what is the difference between show channels sip show channles i see the difference in both show channels show me 30 channels sip show channels shows me 221 channels any description on this ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070909/6dd07e17/atta...

I have one TDM2404E digium card on asterisk box, after configuring the zaptel and zapata configuration files, I am getting these errors when reloading asterisk: ast_unregister_indication_country: Removed default indication country 'us' setup_zap: Ignoring signalling setup_zap: Ignoring answeronpolarityswitch unable to recognize channel 13-5 what is the reason for that? Thanks,

I've got problem with connecting asterisk to CCM. Our side has Asterisk system other side CCM , ehrn i dial a number on other side channles created , connections established but nothing happend , just silence , and after some time busy tone. Sides sending ad reciving g711 codec , but we need that sides send and recive g729 (we have licenses) , if in h323 conf i try to : disallow=all , allow=g729 i doesn't help. What semas t...

Hi all, We're trying to hook up our Asterisk config (Card: TE410P) with a Siemens EWSD switch. The link is ok on both ends (green), with no errors. The problem is when we try to make a call from our side (via call files), we get the pri/E1 error Ext: 1 Cause: Temporary failure (41), class = Network Congestion (2) Our Telecom partner (they checked with Siemens) mentioned that we need to

I'm using T100P with CAC AB II, only FXS ports. After upgrading, asterisk stoped sending caller IDs to the phones. Even inside - port to port. I got 2 errors in the debug: __zt_exception: Exception 23, channle 2 (i'm ringing to channel 2) zt_handle_event: Didn't finish Caller-ID spill. Canceling.

With best regards Abdul Ahad Anwer Khan, M.Sc(CME, in progress) University of Applied Sciences Offenburg Germany Phone:+497814748226 Mobile:+4917623468462 From: abdulahadanwer at hotmail.com To: asterisk-users-bounces at lists.digium.com Subject: New in asterisk Date: Sun, 27 Sep 2009 14:50:59 +0600 Hello All I am a student and doing my thesis which is related to asterisk. I am new in

Hi again, shortly after beta 4 came out i noticed that rumbling artifact oggenc puts into a certain tune of mine. Back then, it was even audible at an ABR of 350kbps. Now with RC2 that bug is completely gone at bitrates of 128kbps and higher. I can hear it again at 96kbps and it becomes more apparent with lower bitrates. At 64kbps, I assume, everybody should be able to hear it. This is not a

...but I would like to avoid this. Here is my setup: PSTN < Adit 600 FXO card > Asterisk Digium single span T1 <Adit 600 TDM controller> Avaya Partner T1 card. It realy is the same Adit 600 that connects to both the Asterisk and Avaya system. I just cross connected the FXO to the first 8 channles on the first T1 in the Adit, and then cross connected the 16 leftover channels from the first T1 to the top 16 (9-24) channels on the second T1 of the Adit, Asterisk is connected to the first T1, and the Avaya is connected to the second T1. Asterisk is getting the CallerID from the FXO cards, it...

Hi, I added a Fritz!PCI card to my asterisk system which is running on fc2 !. I installed the avm modules (fcpci) and when i try an capiinit and a capiinfo then everything seems to be OK (you can see capiinfo and lsmod at the end of the text) !. I installed the asterisk chan_capi module and when i start asterisk and enter the command capi info on the command line i get back: *CLI> capi info

On Thu, Dec 10, 2009 at 10:06 AM, Monty Montgomery <monty at xiph.org> wrote: > Vorbis-dev might be better.... > My issue is with theora/thusnelda for xiph tools. > > What I mean is, you quoted a response from Frank Barchard, but I never > saw his response on the list. Are there blind CCs on the conversation? > I'm just trying to figure out where the rest of the

...RED span=3,3,0,ccs,ami # termtype: te bchan=7-8 hardhdlc=9 echocanceller=mg2,7-8 # Span 4: B4/0/4 "B4XXP (PCI) Card 0 Span 4" span=4,4,0,ccs,ami # termtype: te bchan=10-11 hardhdlc=12 echocanceller=mg2,10-11 # Global data loadzone = us defaultzone = us *******************dahdi-channles.conf********************************** ; Autogenerated by /usr/sbin/dahdi_genconf on Wed Jun 16 22:49:10 2010 ; If you edit this file and execute /usr/sbin/dahdi_genconf again, ; your manual changes will be LOST. ; Dahdi Channels Configurations (chan_dahdi.conf) ; ; This is not intended to be a com...

Does anyone have any recommendations on how to connect 160 analogue phones to an asterisk PBX? Background information: A client wishes to replace their current PBX with a new VoIP system. Currently they have 2 PRIs. I intent to set up 2 asterisk PBXs with Debian GNU/Linux on raided drives. These drives will be mounted only read-only to recover gracefully from power-cycles. I am considering 2

...RED span=3,3,0,ccs,ami # termtype: te bchan=7-8 hardhdlc=9 echocanceller=mg2,7-8 # Span 4: B4/0/4 "B4XXP (PCI) Card 0 Span 4" span=4,4,0,ccs,ami # termtype: te bchan=10-11 hardhdlc=12 echocanceller=mg2,10-11 # Global data loadzone = us defaultzone = us *******************dahdi-channles.conf********************************** ; Autogenerated by /usr/sbin/dahdi_genconf on Wed Jun 16 22:49:10 2010 ; If you edit this file and execute /usr/sbin/dahdi_genconf again, ; your manual changes will be LOST. ; Dahdi Channels Configurations (chan_dahdi.conf) ; ; This is not intended to be a com...

Hello, more and more operators in Europe offer music instead of ring tunes. E.g. instead of the 400 Hz or whatever tunes, the caller will hear J-Lo, or Mozart.... Currently I will have to answer the line to do that. Is there a way to do this with asterisk? Regards, Marc -- CTO Marc Storck MS Networks SA mstorck@luxadmin.org Internet Service

I'm staring at an RFP--this company wants to replace a 2000 position PBX (at eight locations) with a new system. Their mindset is Nortel/Avaya because they talk about 28-button digital sets. The do specify a few IP phones for just one location, so they are aware of VoIP. I'm going to bid on this--there's nothing to lose except the time it takes to write the proposal. I'll

from voip-info wiki Asterisk automatic daily restart To automatically restart Asterisk you can add something like this to cron # Restart Asterisk PBX once a day to prevent any problems from piling up 10 7 * * * root /usr/sbin/asterisk -rx "restart now" >/dev/null 2>&1 or 10 7 * * * root /usr/sbin/asterisk -r -x "restart gracefully" >/dev/null 2>&1

Hi, I hope someone can help me with this.... Asterisk 1.0.6 Zaptel 1.0.6 Libpri 1.0.6, 1 Digium E100P card installed Panasonic TDA200 firmware v2.0.6 E1 Card Firmware 1.0.2 System is located in Australia, so as technologies go, I believe it is twist on the euro standard for the E1 signalling. Here is the situation. The TDA E1 card is set in cross over mode and I am using a functional