similar to: [offtopic] Asterisk <-IP-> Siemens HiPath 4000

Hello, Configuration: Asterisk CVS HEAD 20050730 on RH EL3+ DIGIUM TE110P PRI card + euroISDN E1 I am trying to sort out the problem: 1. Provider's switch sends "SETUP"; 2. Asterisk receives "SETUP", rings allocated extension but does not send "Setup acknowledge" (or any other messages) to switch; 3. After 4 seconds of waiting of *'s response switch sends

Hello all, I have Asterisk 1.2.1 and a TE110P connected to a Nortel Meridian Option 81C system. The PRI line is currently setup as DMS100. Here are the relevant lines from zaptel.conf and zapata.conf: zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone = us zapata.conf: [channels] language=en context=from-internal musiconhold=default switchtype=dms100

Hi, I have this setup: E1 PRI PSTN -- Asterisk -- Alcatel PBX - analog phones Can someone tell me what's wrong with this call initiating from an analog phone connected to Alcatel PBX? It dies with NOANSWER but all works if I call other destination numbers. Dialplan is a simple Dial(zap/g1/0984465691) statement. At the end you'll find also zapata.conf.

Hey all, I am running a Asterisk 1.2.9.1 with Sangoma A101 card, newest firmware, configured with a help from Sangoma Tech Support, running fine. It is connected to a PRI circuit split from Cisco MC 3810, which in turn is connected to a Converged T from CTC Communications. While Asterisk works fine and I can call in/out on my BV account, I am only able to dial in through CTC. I have spent

hello everyone, I have a trixbox server with an E1 card(not digium).It connects to an AVAYA pbx use E1. It works fine.But when i change the E1 card to digium TE220B,there is a problem. When sip extension A(on trixbox) call PSTN extension B(on avaya),A must wait longtime before B start to ring.From the log I find there are two times call. I don't know why the first request be rejected

Is there a setting somewhere in * to define whether I am receiving callerID or true ANI? Global Crossing claims they are sending ANI but I dont think so. My understanding of ANI is that it is always sent, regardless if callerID is blocked. If I dial *67 and my DID, I get "Presentation: Presentation prohibited of network provided number" and no number. Before I call GC on Monday

I there, I have some trouble to do working caller id name for outgoing calls on the PRI we just installed. Caller id name work on incoming calls only. Caller id number work on incoming and outgoing calls. The provider, Goup Telecom, said that's in what i'm sending. They said that I send the cid name in ascii code and to do it working, I need to send it in hex. So I take some traces

Hi! For demonstration purposes I try to bridge an incoming video call from a 3G mobile handset to another 3G mobile handset using asterisk as "switch". On the incoming call leg I see all expected bearer capabilities (Digital, 64k Transparent, G.7xx 384k video) but on the outgoing call leg the bearer capability G.7xx 384k video get lost and therefore the call is dropped from the mobile

I have a T1 link from asterisk 1.2.23 (also tried with 1.4.13) to a Meridian Option 61C. Calls either way drop with error "Channel 0/23, span 1 got hangup, cause 100". Can anyone offer insight into the cause and solution/workaround? (I tried upgrading to Ast 1.4.13, and upgrading matching zaptel & libpri, put the problem is identical). For testing, I tried a call from the

OK I am going to do it again. Global Crossing is now sending ANI but it is not in the format I expected. Any one know of a way to get this data into two seperate variables? The first number is ANI and the second is DNIS so it is "*tendigits*tendigits* on one line like below. < Called Number (len=26) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan

I posted a problem earlier thinking it was due to a lack of sound card. Several members stated that you do not need a sound card to play audio to a PRI channel. I did some further testing and discovered that there is a problem with call progress tones or signaling on my PRI. I think that the reason I am not hearing audio from the MeetMe() or Playback() apps. is because the the calling side of

I have a PRI E1 link between Asterisk 1.4.24 and Alcatel-Lucent OmniPCX Enterprise R9.0. As EuroISDN it works fine. However, I need to move to QSIG because of a firmware upgrade on the Alcatel PBX which doesn't support EuroISDN (please don't ask why). Besides, I've read somewhere that 2 B Channel Transfers "should" work with * 1.4, the latest 1.4 libpri and QSIG. So this

Hi, When we make an outgoing call on ISDN (zaphfc) with overlap dialing we get immidiate hangup after answer. But when we place a full number before dialing everything is ok. Any help appriciated!! Thanks here is info with debug: == Primary D-Channel on span 1 up -- Executing Dial("SIP/200-164c", "zap/g1/|100|tc") in new stack -- Making new call for cr 192 --

Hey all. We have a DS3 circuit with GBLX split off into 7 systems with a 4 port sangoma card (A104D) in the first 2 systems, and digium T410P cards in the other 5. GBLX numbers their spans from 0 to 3 instead of 1-4 and we have a NFAS configuration with the d-channel on chan 96. All of our systems are running 1.0.7 for stability reasons (and no good time for maintaince, the entire platform

Hi all, I have an asterisk-server (cvs-head from august) connected to a carrier's switch (DMS/Euroisdn) via a te410p, and I am having problems with DDI (standard 'official pstn' number plus extra digits for 'internal' use) Basically, when the entire number (including the extra digits) is dialled via a redial or a programmed key, I see the entire called party number (including

So I reached the point where my PRI is accepting incoming calls, but I cannot dialout. I must be doing something stupid, but I can't figure it out. The Asterisk box is sitting between the Mitel and the phone company, and has PRI lines to each. Asterisk was built from CVS r1-0 Log for a call from mitel heading outbound: ------------------------- -- Accepting call from '' to

Hi all, using latest asterisk-svn I want to reflect an video call incoming via an PRI EuroISDN channel to another outgoing PRI channel, and I want the the outgoing channel to have the exact same bearer capability < Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Unrestricted digital information (8) < Ext: 1 Trans mode/rate:

Hello All, I'm having problems connecting Asterisk to Telco in Poland (using E1). The telco guys are saying that the RING message is missing. How can I make Asterisk to send the RING message? Does anyone have any samples of zaptel and zapata for Poland? Best Regards, Alex ____________________________________________________________________________________ Do you Yahoo!? Everyone is

Hi I have recently upgraded my Asterisk system (from 1.2 to 1.4) and I have started to notice the following messages when I recieve a call on my Zap channel :- [Mar 11 09:20:17] WARNING[4294]: chan_zap.c:8851 zt_pri_error: 1 copying 5 bytes LLC I have a single PRI ISDN 30 link to a Siemens Realitis DX. Here is my zapata.conf :- [channels] echocancel=no echocancelwhenbridged=no rxgain=-5.0

Hi. recently I have been trying to setup a PRI on asterisk. Inbound calls are working just fine but I am not able to make outbound calls. Does anyone know what I need to change to make outbound calls work? Right now the PRI is instantly hanging up on the outbound calls. I have included full debug info as well as config files. /etc/zaptel.conf span=1,1,1,esf,b8zs bchan=1-23 dchan=24