similar to: PRI Issue - Calls being rejected with unacceptable channel

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.

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

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, 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

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

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

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

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

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

We've got a EuroISDN (32 channels) with a TE405p, running cvs head as of 5 days ago. In the past couple of days, we've hit a scenario where incoming calls to the * pbx from the PSTN are being marked as busy, but outgoing calls work just fine. When we reboot *, the problem goes away. Has anyone else had this ? I've attached a PRI debug below. I've changed the phone numbers (x

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

Having a problem w/ not getting CID name from a PRI. CID Name appears in the PRI debug, but even after a Wait(4) it still appears after the phone is ringing. Here is the relevant info from my PRI debug output. Line 4 is a NoOp showing me trying to echo Name and Number. Line 6 dials the extension, and you can see callerid name get presented on line 29. Again, there is a Wait(4) before the

Hello all, Recently I purchased an QuadBRI card from junghanns.net after some playing around, reconfiguring dialplans etc with the exception of 1 thing everything seems to work: I seem to be unable to set the outbound callerid. The dutch telecom operator (KPN) provided me with 4 MSN's on 1 BRI interface. In the past years I'm more then used to setting the MSN without the leading 0, this

Hi all, I'm trying to insert a Rhino Ceros box equipped with a Rhino R2T1 dual-T1 card and running the latest version of Trixbox (2.4.2) between the central office and a Nortel Option 11. The switch at the CO is a DMS100. Basically, I'm taking the T1, connecting it to port 0 on the R2T1 card, and then connecting port 1 to the Nortel. (Actually a CSU and then the Nortel) We're running

We've been working for the past 2 weeks to get a new V400P working with our PRIs from the telephone company. We're trying to get the Asterisk server setup as a VoIP gateway for SIP and AIX. We can make SIP-SIP calls, but all calls from or to the PRI fail. This is the applicable entries from the Asterisk log (configuration files follow) for a call coming from the PSTN on the PRI. I

Hello! Is it possible to connect Siemens HiPath 4000 to Asterisk? What equipment required on Siemens side? I mean IP not E1. Sorry for asking here. Siemens-related websites use "salesperson language". There is no technical information.

Hi all, I tried to connect the bristuffed(0.3.0-PRE-1i) * to an Alcatel PBX via BRI (zaphfc) and Q.SIG. The Alcatel PBX is connected to the outside world and should forward our calls to the telco. This setup works correctly as far as I use euroisdn as the switchtype. The first problem was that it is only possible to run the * side in CPE-mode -- I wanted NET. Anyway, I configured * this way:

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

Hi Guys, I am calling out 416-999-1111 on Channel 1 of PRI and then calling 416-999-2222 on Channel 2 of PRI. When the two channels are going to be ZAP native bridged, both channels hangup and CLI show PRI cause (16). Asterisk Verbose *(Channel 1 already connected to party)*: -- Requested transfer capability: 0x00 - SPEECH -- Called g0/4169992222 -- Zap/2-1 is proceeding passing it