similar to: pridialplan=unknown ?

Hello, I have an Asterisk system in the UK using ISDN service from BT. My problem is that the called number is always passed to the provider with the Type Of Number declared as ?national? despite pridialplan (and prilocaldialplan) is set to ?unknown?. My questions are: - How can I find out what the effective value is for the above two variables without actually making a call? (I have very

I have connected asterisk 1.0.7 with Avaya Definity via E1 with a TE100P Digium Card. Inbound calls are working perfectly and I dont have any problem. But when I try to make an outgoing call with my softphone (xlite) I am getting the following messages. Hungup 'Zap/13-1' Executing Dial("SIP/IZ-bc0a", "Zap/g1/3118") in new stack Called g1/3118 Channel 0/1, span 1 got

Hi list! What exactly is the meaning / function of the pridialplan & prilocaldialplan? I've been trying to find out what the different possibilities for these settings are but couldn't find a clear answer. The possible parameters I could find are are : local,unknown,dynamic,national,international and maybe there are more? Thanks!

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.

Only a warning, and doesn't seem to do anything bad. But I can't seem to figure out what these warnings mean? -- Requested transfer capability: 0x00 - SPEECH [Feb 17 12:33:03] WARNING[10750]: chan_dahdi.c:3096 dahdi_call: Unrecognized prilocaldialplan NPI modifier: k [Feb 17 12:33:03] WARNING[10750]: chan_dahdi.c:3096 dahdi_call: Unrecognized prilocaldialplan NPI modifier: o [Feb 17

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 have Asterisk working great for a while now, and until now, only needed to forward calls which originate from one country (Holland, +31). We have a Wildcard TE410P card and configured it in zapata.conf as: switchtype: euroisdn pridialplan: unknown ...etc This was all fine, but now we want to also forward calls to the voice-switch with a calling-number with a different country code.

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

I have a PRI link up and running between Asterisk and a Nortel Norstar MICS v4.1 . I'm having a problem getting the textual Caller Name across the link from Ast to Ns, however numeric Caller ID arrives and displays fine. From Ns to Ast both elements come through fine. I'm forcing dummy values for testing using: exten => s,1,SetCIDName(Test) exten => s,2,SetCallerID(1234561234)

Trouble getting chan_sip2 to compile below is what I have done -download acl.c.patch,acl.h.patch,chan2s_sip.c to /root/software cp /root/software/chan_sip2s.c /usr/src/asterisk/channels cd /usr/src/asterisk/ patch -p0 acl.c /root/software/acl.c.patch cd /usr/src/asterisk/include/asterisk patch -p0 acl.h /root/software/acl.h.patch - added the follow to /usr/src/asterisk/channels/Makefile

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

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, 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

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 --

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

When receiving multiple calling numbers via a PRI for a call setup, I cannot find the ability to select between either first or last. Is there a way to do this currently? If not, can anyone help me in getting this to work? Here is the dump of the PRI in intensive debugging. Please note that I want the first calling number, not the last. Any help would be much appreciated. Below the PRI trace

Hello, I recently bought a Junghanns Octobri Card. I have some problems with this card to make outbound calls but I can receive calls. I have 3 lines to PSTN and 3 lines to my existing PBX FRANCE TELECOM <-- OctoBRI --> Asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1h <-- OctoBRI --> PABX e-Generis <----> ISDN Phones |

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

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

Hi, ? I'm trying to figure out what the issue is splicing Asterisk between our Telewest PRI and a GDK-186 with a PRI card. ? We're using the Digium TE405P ? Our telco provider is Telewest, and Telco directly into switch is fine. ? When I splice Asterisk in, I can make and receive calls from Asterisk extensions, I can make outbound calls from the GDK, but inbound calls do not seem to pass