similar to: SIP and ZAP

Greetings all, I'm trying to connect to an AT&T teleconference, but the call is never marked as ANSWERED by asterisk and therefore won't bridge and continue. The only work-around I've come up with so far is to dial like this: Exten => 744,1,Dial(Zap/g1,,p) The "private" mode keeps the line open without trying to do a bridge, but requires the

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users- > admin@lists.digium.com] On Behalf Of Matteo Brancaleoni > Sent: Monday, July 26, 2004 5:22 PM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Can't dial SIP<->EuroISDN (HFC-S based > PCIISDN card): Unable to create channel of type 'Zap'

Hi, Im working with the following versions: -asterisk-1.2.12.1 -zaptel-1.2.9.1 And with the following card: 00:0d.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device 8085:0003 Flags: bus master, medium devsel, latency 32, IRQ 201 I/O ports at c800 [size=256] Memory at fe000000 (32-bit, non-prefetchable)

Hi Guys!! I wrote a little patch for asterisk 1.2.5 and I will maintain it for future release unless somebody explains me how we can ask people at Digium to add it to the source tree... We are planning on using Asterisk as our main PBX for the office over the next few weeks. Our current setup uses TDM400 cards to bring our 8 lines into Asterisk, our Telco provides us an option for three

Hey, Im running Asterisk 1.2.2 and im having problems with the audio when bridging calls between the zap interfaces and sip. zap to zap work fine, as do sip to sip (but asterisk isnt in the media stream, as it doesnt need to be) and terminating the call and playing a test message via either sip or zap work fine. Basically, the only time I see this problem is trying to bridge between sip and

Hi! Is there a way to add a prefix to all called numbers that are coming in from zap-channels? I know that the "*prefix" options in zapata.conf will do this job for the caller-id but I need it for the callee-number. Right now, I'm using the following hack (in ael-Syntax) to add "99" to the called number before entering the default context, but I'd like to get rid of

Hello, When I receive calls in one FXO port (TDM400 or A200, occurs in both) and it dial to one invalid SIP extension, the call never hangup. The call would have to be dropped, but it seems that "Starting simple switch on 'Zap/1-1'" and "Hungup 'Zap/1-1'" occurs almost at the same time. If the dial is made to a valid SIP extension, the call is

Asterisk Users, I have 3 FXO modules with the TDM400P Digium Card. I can dial into the Asterisk rings my Sip phone, but dialing out with my SPA941 phone through the zap channel is a problem. I keep getting this message on the Asterisk CLI. What am I doing wrong? Thanks in advance. -- Executing [103 at default:1] Dial("SIP/200-006fa300", "{Zap/g0/{EXTEN:1}") in new

Hello list, I have a problem with called ZAP channels making an attended-transfer or blind-xfer. Signalling at the phones is ok, but the CDR of Asterisk is wrong. At the moment there is a bristuffed Asterisk 1.2.18 running with bristuff-0.3.0-PRE-1y-g. Here is my dialplan, I simplified it a bit: [default] exten => 0123456789,1,Macro(dialpstn,${EXTEN}) [macro-dialpstn] exten =>

Hello! I have several problems in the audio one belonging to asterisk at conferences between ZAP - SIP. I hope that you may help me. 1 Problem When the audio establishes a call between two canals, some zap and another sip itself one listens interrupted in one of the senses, exactly in zap sip, with audio cuts. Words lose at random, in that connection, however in the sense sip zap the

I am having a problem trying to do inband DTMF passthru via asterisk. My setup: PSTN gateway MAXTNT v11.0 SIP (T1 PRI/NT2) Asterisk HEAD or v1.0 makes no difference (I am using HEAD mostly) 12/10/04 and 01/17/05 (no difference) CAC ABII-T100P to/from analog lines to/from asterisk BTW, I have used a ABI and it works just like the ABII with asterisk. What I am seeing is: I make a call from a

Hi, Asterisk 1.4 Working (jitter buffers created as expected): ZAP -> SIP SIP -> ZAP Not working (no jitter buffers created): SIP -> chan_local (with /nj) -> ZAP SIP -> chan_local (with /j) -> ZAP SIP -> chan_local (with no flags) -> ZAP I have this in zapata.conf: jbenable=yes jbforce=no jbimpl=fixed jbmaxsize=300 Is there something I haven't tried that will make

Hello - I have 8 polycom 501s all setup great using ulaw. We have put them through a pretty rigorous torture over the last 4 months, and they've performed famously. No dropped calls ever. We invested in some g729 licenses. changed my ipmid.cfg so that g729 is priority 1 and ulaw is priority 2. I added allow=g729 to my extension's sip.conf entry, where existed before disallow=all

please help!!! I am dialing into our asterisk server(TDM400p) from the psnt. I hear our voicemail message and I press the extention 1000. The Polycom ip phone in the office rings. I pickup but neither side can hear one another. What have I done wrong? thanks sip.conf: [general] context=local-access ; Default context for incoming calls bindport=5060

This is weird, and could find nothing on the wiki or * google search. Regardless of the ackcall setting in agent.conf, if I have agents logged in to a phone on a zap channel, when a call is made the agent's phone rings, and when they answer, they have to press "#" in order to hear the announcement. If the same agent is logged onto a sip channel, then the announcement is played as

I have a TDM card in a asterisk machine. I found that once I used it to call out, the call status changed to "connected" even the callee is still ring. How could asterisk distinguish the "ringing" and "connected" in zap channel? thanks.

I'm getting this messages trying to call with one sip trunk: Feb 3 16:43:09 DEBUG[3389] channel.c: Avoiding initial deadlock for 'SIP/usa-e2ea' Feb 3 16:43:09 VERBOSE[3491] logger.c: -- SIP/usa-e2ea answered Zap/1-1 Feb 3 16:43:09 WARNING[3491] channel.c: No path to translate from Zap/1-1(68) to SIP/usa-e2ea(256) Feb 3 16:43:09 WARNING[3491] app_dial.c: Had to drop call

Here's our config: cisco 7960's running 6.3 sip code latest cvs of * t100p zaptel card adit 600 channel bank 7 pots lines and 2 fax machines on the adit 600 dialing out from the cisco phones gets sent out via the zap channels, but I'm having some serious echo problems. I currently have the adit set to +3 rxgain and -6 txgain, with my zapata.conf containing: echocancel=128

Hi everyone, our setup includes a NEC PBX connected to our asterisk via bri lines. The NEC has a doorphone feature, which is just an extension that calls you when someone rings. When connected to this extensions, a 'flash' signalling opens the door. So now, i'd like to trigger this from asterisk, too, but unfortunately wasn't able to do so. Setup: asterisk

Hi, I need little help with setting up new server i recently bought TE110P/E1. I have 1 PRI and I will be using Soft phones on the agents site. This is 1st time for me to work with Zaptel card and i am having some problem with that. if any one call help me with this or care to share extensions.conf with zap>sip sip>zap with me will be great. I want to be able to make call through SIP