similar to: Asterisk QSIG doesnt send the calling name to Nortel CS1000

*I'm trying to build an application that provides statistics of calls*>* and call recording. Someone told me this could be done out of band*>* with a SPAN (?) port that would replicate SIP and media packets to a*>* separate NIC without having to actually pass the real-calls thru*>* asterisk. It was explained that this SPAN port would in the SBC*>* would replicate data

[Reposting, as was bounced for non-member, sorry if this is a dupe] Arrangement: { PSTN }--E1--[PBX]--E1--[*]--LAN--[SIP phones] \__[PBX system phones] Normal calls between PBX system phones and SIP phones work, in both directions. The call logs look like (ignore the no answer, it did ring):

Guys, I need to make asterisk connect to another device over a pri using QSIG signalling (master/slave). Is this feasible and has it been done before? Thank you! al. _________________________________________________________________ Get ready for school! Find articles, homework help and more in the Back to School Guide! http://special.msn.com/network/04backtoschool.armx

Hi, I have asterisk running a single T1 card with a connection to a nortel CS 1000. All calls to extensions, local and long distance are working just fine. My issue is this: The nortel CS 1000 supports connections to and intercom system that is just line level audio to speakers. When my PRI tries to call this "extension" it is never answered so my AGI never runs to play a message

folk, I see from the archives that the issue of nortel handsets not sending dtmf tones to asterisk has been discussed a couple of times, but there is something I quite havent seen answered yet. Is this dtmf issue a problem with the nortel handsets or the PBX itself? If the handset were changed to another one, would this issue be solved? Or must the PBX be changed completely? rgds eb

Hi everyone, I was referred to this mailing list by Richard Mudgett regarding the following thread on Issue Tracker as it's a feature request and not a bug:https://issues.asterisk.org/jira/browse/PRI-170 We've got a Nortel BCM (actually 2 BCM50s and 1 BCM400)... all showing the same problem. When a call comes in the number shows up OK but the name does not. In looking at the ISDN messages

Here is the CS1000 log. Again, the CS1000 using SIP accepts incoming calls just fine. However, using outgoing call files the CS1000 is hanging up after I answer the call. I dont know why? Thanks, for any assistance. Jerry my sip.conf entry is: [Nortel] type=friend dtmfmode=rfc2833 username=XXXXXXXXX disallow=all allow=ulaw allow=alaw

Hello, I am using a Cisco-2,811 router with PRI as a gateway between Asterisk and Nortel TX-1. I had problems with name transfer and with the help of Cisco support I've fixed it. Enclosed here are the definitions needed for it. BTW, Cisco's CCM is using MGCP thus the Q.sig is handled by CCM. Here I am using SIP so the router must decode/encode the Q.sig. The Nortel should be defined

I'm completely new to asterisk and while we have access to Nortel experts none of them know asterisk and since I'm the network guy I've been lumped with this. This is what I'm trying to accomplish We have a CS1000 that's sip capable. I want to be able to connect an Asterisk box to the cs1000 via sip as an IVR, so I can pass a call off to asterisk, go through some IVR menus

I am connecting to a Nortel CS 1000. I can place calls out to an extension so we are half way there. When calling into the box I get the following from sip debug ip X. I get dead air when calling into the box. In my sip.conf I have a context of nortel and in extensions.conf the nortel context just has a s,1,Playback(demo-congrats). Any suggestions as to why the call in might not be working but

I am using asterisk 1.4.32 and wish to connect using SIP to a nortel 1000 switch with the ability to have 90 calls at a one time outgoing or incoming. the nortel reseller is asking me what to do. I dont know nortel at all. I thought I just needed a "SIP trunk and IP address of the their server and an account name, and provide her my IP address". They didn't know what to do with

I am looking to get the info about QSIG support in Asterisk. Does Asterisk have QSIG support? Does Asterisk support QSIG SIP Tunneling or QSIG SIP Interworking? If so, How to configure that? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060503/3cb7f966/attachment.htm

Hi, Is Asterisk "fully QSIG-compliant"? I currently have an Alcatel 4400 connected to Asterisk 1.2 and 1.4. Zaptel versions are 1.2.26 and 1.4.11. I am using switchtype=euroisdn and all works fine. However, it seems that Alcatel's latest firmware has dropped support for euroisdn which is really despicable. So now I need to see if I can migrate to QSIG which is supported by

I have read most of Eicons information on Q.SIG, and I am able to load the Q.SIG protocol (instead of ETSI for example). No strange logging in divacrtl mlog. But how do I tell Asterisk to understand Q.SIG? My PBX is configured for QSIG, but I cannot see anything on my trace when trying to make a call via the S0(Q.SIG) Janne > -----Ursprungligt meddelande----- > Fr?n:

Hi all, I'm planning on connecting our Asterisk to our legacy PBX (an Avaya INDeX). I was originally going to sit it between our ISDN connection and the INDeX (tried it, worked ok) but now I intend to hang it off a spare PRI card just so should the * fail we keep our ISDN's at the INDeX, yes I've looked at ISDNguard but little info from the manufacturers. Anyhow, I'm now

That was my impression as well, I tried adding switchtype=qsig in zapata.conf, but all I see is the capi information... No connection... Janne > -----Ursprungligt meddelande----- > Fr?n: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] F?r Peter Svensson > Skickat: den 23 februari 2005 21:34 > Till: Asterisk Users Mailing List -

Hello, I have connected asterisk box with legacy PBX Alcatel OmniPCX 4400 (and also another * box connected to A4200). These PBXes have function to assign name to extensions and display it on phone. Asterisk box is connected via PRI with euroISDN signalling (also I have tried QSIG). Is it possible to set callerid with name and display it on alcatel digital phones? With command SetCALLERID

Hi, we are still trying to properly connect a Tenovis PBX to an Asterisk server (asterisk 1.2.6, libpri 1.2.2, zaptel 1.2.5, Digium Wildcard TE110P), this time with QSIG. Calling from a Tenovis phone to a SIP phone (i.e. traditional phone -> Tenovis PBX -> QSIG -> Asterisk -> SIP phone) works with the following messages: --- Don't know what to do if second ROSE component is of

I love it... I buy a half a million dollars worth of Trashiba's finest ... I download Asterisk for free... I now refer to it as legacy 18 months and 300 extensions later! Anyway, I am trying to integrate my dial plans acrossed platforms. PSTN>>>CTX670>>>Asterisk The dialplan I would like to setup, 1xx,2xx,3xx,7xxx to the CTX 670 4xx,6xxx to a remote ctx100 (this is setup

Hello, Does anybody know if Asterisk can support QSIG protocols to be interconnected with a Traditionnal PABX? (Using a HFC chipset based ISDN card to emulate NT Interface) Thank you in advance ;-) Ignace