similar to: Wildcard TE410P still making trouble

Hi all, I have to asterisk-1.0.9BRIstuffed-0.2.0n boxes each equipped with a TE410P. Box A is connected with pri1 to the PSTN. Box B is connected with pri1 (cpe) to the Box A at pri2 (net). Now I want Box B to dial out to the PSTN tunneled thru Box A and have it get all ISDN indications in case of call failure, eg. unallocated destination number etc. But currently Box B always gets only

Dear Gurus, i am facing some unknown problem here.. first, let me describe my case. my office is using a nortel merridian option 11c PBX. and it is connected to the provider thru a E1 card, which is working fine with no problem. my plan it to slot in a server with 2 TE11XP card to intercept the outgoing calls. Provider <--te11xp---> asterisk <---te11xp------> nortel merridian

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

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

I have this setup: (PSTN E1 PRI) -- Asterisk -- (crosscable) -- Alcatel PBX --- analog phones and a few of VoIP phones directly connected to Asterisk. Calling a number (just one!) - an automatic responder (IVR) - from VoIP phones works, from analog phones doesn't work: NOANSWER after a few seconds. I'm using no 'r' in dial options (this caused a problem with an IVR some time

handled. So....what do I do? Thanks, MD =1=================================================== !! Invalid Protocol Profile field 0x11 -- Accepting call from '2004000' to '111' on channel 0/23, span 1 -- Executing NoOp("Zap/23-1", "Incoming call from Meridian1") in new stack -- Executing NoOp("Zap/23-1", " From number: 2004000|

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 all, the scenario: Carrier ----S2M------ * -----S2M------Siemens | | SIP Clients and many other features With much help from the list, the PRI links are without alarms and inbound calls are working fine (from both: Carrier and Siemens). But I am not able to dial wether outbound nor to the Siemens PBX. I allways get the message: == Everyone is busy

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:

Hi, Ive been playing on a asterisk to orion gsm box E1 pri setup. I have achieved incoming calls to be passed to my asterisk box successfully but outgoing calls will just I have tried playing with various pridialplan and overlapdial settings and with no success. If anyone can make more sense from the log, I'd certainly appreciate it. I am sending a 10 digit number to be dialed. I guessed

I have a working system with asterisk 1.4.26.2 libpri 1.4.1 and zaptel 1.4.12.1 With a digium TE205p. I am trying to update to libpri 1.4.10.2. When I do, incoming calls work but outgoing does not. When I do this I "rm /usr/lib/libpri*" then just install libpri-1.4.10.2 as normal. I then do a make clean in asterisk and make distclean ,then configure, make and make install. I do

Hi all, I have some problem to send a FAX to a given number. I use asterisk 1.2.18, on a openSUSE 10.2, i586 host. The FAX is sent out via an ISDN PRI interface, I'm in Germany, and the destination FAX devices are in Germany too, but in different areas, so I have to use a city prefix. I did set the pri device in debug mode, below are two calls, to two different FAX numbers, the first is

Salve, I'm somewhat stuck on how to get DTMF working with my setup and googling didn't yield anything similar. My setup consists of one CAPI-capable board (AVM Fritz!DSL) connected to a BRI (T-ISDN), one HFC-S board running in NT-mode connected to an internal S0 bus with some ISDN devices (DECT stations, TA) and, of course, some ethernet interfaces. ISDN standard used is Euro-ISDN.

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.

picture. I know the firmware on the Nortel is old, so I'm guessing that libpri is sending something that the Nortel does not know how to handle. Is there a way to dumb down what libpri sends? From everything I've read PRI is an evolving standard - and older devices may struggle with newer extensions/developments. (This might be very handy for users trying to talk to old pbx's.) Is

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

Yes its a T3 split into seven trunk groups with one D channel and NFAS on each. Can you explain the cut function or point me somewhere please? Thanks, Steve -----Original Message----- From: Alexander Lopez [mailto:Alex.Lopez@OpSys.com] Sent: Mon 4/10/2006 12:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: RE: [Asterisk-Users] ANI and DNIS Seperation on

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 Folks, I've Asterisk Bristuffed up and running behind an Auerswald Commander Basic ISDN PBX on the internal ISDN Bus (BRI/PTMP). The HFC Card works marvelleous for outgoing calls (as the parallely installed avm fritzcard with chan_capi does), but when I'm trying to call in, I get a short ring signal and then the connection is terminated. This does not happen with chan_capi and

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