similar to: error "WARNING[28697]: File app_dial.c, Line 304 (wait_for_answer): Unable to forward voice"

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

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

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

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

When trying to make outbound calls I am getting the Warning: File app_dial.c line 313 (wait_for_answer) Unable to forward voice. When making the call it attempts to dial (pounds are actually numbers but replaced to not show numbers we are dialing): Executing Dial("Sip/donas-bd7b", Zap/g1/1##########") in new stack Called g1/1########## Channel 1, span 1 got hangup **Above

We are still having problems getting a Wildcard to work with a German E1 (PMX) interface. When starting asterisk it shows all B-channels starting up successfully (although our carrier told us only the first B-channel starts, if any at all). Incoming calls are not being signaled at all. (They seem to be intercepted by the carrier's switch, as no B-channel is up) Outgoing calls sometimes work,

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|

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

P.H.B. is insisting on having the ability to create a transparant SIP tunnel between old style ISDN telephony PBX with overlap dialing: PBX - ISDN - IAD - SIP - * - DAHDI - PRI The idea is that dialed numbers a the PBX are transmitted to the PRI as they are typed, whenever the PRI gets the signal that the number is complete the dialer instantly gets a ringing. This behavior is described in RFC

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

Hi, When have a incoming call from E1 to a extension FXS, and this extension is busy, the incoming call recive ring tone, and it is wrong. What can I do? Thanks in advance Pedro Here is the trace: asterisk-1*CLI> < Protocol Discriminator: Q.931 (8) len=41 < Call Ref: len= 2 (reference 66/0x42) (Originator) < Message type: SETUP (5) < Sending Complete (len= 4) < Bearer

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

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

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

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

I just downloaded the cvs version CVS-07/28/03-14:45:19 and now I cannot recieve the the calls from the zaptel interface which is a E100P with pri signaling. That is something with asterisk becouse rolling back to version from 06/23/03 using the new libpri and zaptel fixes the problem. Here is an exept from the config: [macro-stdexten]; ; ; Standard extension macro: ; ${ARG1} - Extension

We have set up an Asterisk PBX managing a EuroPRI in Italy. We have conneccted to the asterisk PBX some Cisco IP Phones and a Panasonic PBX with 10 analogic phones. If we dial an unassigned telephone number we are not able to listen to PSTN Operator message telling that the subscriber does not exist both on IP phones and analogic ones. Asterisk simply hungs up. We have repeated the test using a

Hello List I'm trying to figure out what is wrong between my asterisk and my NEC 2400 pbx We have been trying to link them with a spare PA-24DTG from the NEC, I'm able to call an extension on the Asterisk, however the extension rings, and then immediatly hangs up I traced it back to the debug of the PRI on the Asterisk... I would appreciate if anyone could pin point what is wrong The

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