similar to: Definity Asterisk CallerID Issue

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.

You should run the 4ESS protocol ("a" on the Definity command "change DS1 board#) as the Definity may not send Display Name for the NI-2 protocol (settings "b" or "d") Remember to make the Asterisk zapata settings consistent with the Definity. On the Definity trunk group form Page 1, change "Codeset to send Display" to "0 (zero)", also make

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

Yes, I can make a call on that extension from other definity phone, if you mean it. -----Original Message----- From: Ken Godee [mailto:ken@perfect-image.com] Sent: 30 ÉÀÌÑ 2004 Ç. 19:14 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Connecting Asterisk and Avaya Definity By E1. Incoming work, but not outgoing Roman Bessyadovskii wrote: > Hi All. > > I connect

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

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

I'm not a big fan of Line Side for Asterisk applications.... the individual channels have to be reserved (reminds me of the "old" days in telephony) and I don't like that... rather make it a tie trunk and use ARS/AAR routing in the Definity... much more general application. This also makes routing on the Asterisk a lot simpler. Easy to make the changes.... will advise if

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

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 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

Outbound calls on my Telus PRI aren't taking the Name portion of the callerID. I've looked at the logs, and it is being set (see below), but the PRI debug output doesn't show the name being sent anywhere. As a result, received calls always display from Unknown (or just the number). Is there some config that I've missed somewhere? I'm running NI-1 (Telus says NI-2 doesn't

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

Sorry Martin to bother you again! I have an ISDN flux with 100 numbers. The local PSTN is sending now the DNIS/DID (so they said!!!) (I have set the immediate=no in zapata.conf) but I have the same problem as before : NOTE : the number is alphanumeric-DID alphanumeric (I will make tests with numeric mumber!). WARNING[9226]: File chan_zap.c, Line 5275 (pri_fixup): Call specified, but not

root at cs6:~# zpool import pool: content3 id: 14184872052409584084 state: FAULTED status: The pool metadata is corrupted. action: The pool cannot be imported due to damaged devices or data. The pool may be active on another system, but can be imported using the ''-f'' flag. see: http://www.sun.com/msg/ZFS-8000-72 config: content3

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 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

Hi All, In my dialplan and standard asterisk CLI logging i see that i am able to restrict the callerid when dialing out with asterisk. however, on the receiving phone, the callerid is still displayed. When i increment the logging of the pri with "pri set debug on span 1" on the CLI i also get the lower level debugging info from the pri.

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

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, ? 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