similar to: Asterisk and Cisco PRI gateway config

Hello to all, I have a problem with an asterisk qsig. I have three machines: Nortel CS1000 --- Card Sangoma PRI ---> Asterisk QSIG ---SIP Trunk---> Asterisk I use Snom phones on Asterisk. If I call from Asterisk to Nortel, Nortel reminds me of the name of the person i'm calling and I visualize on the display of Snom phone, but if I call from Nortel to Asterisk, the QSIG does not send

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

Hello, I am using a Cisco 2,811 gateway to connect Asterisk over PRI to our Nortel TX-1 PBX. Up to now I had only the calling party names passed both ways. After upgrading the Cisco to the latest release (12.4.24T) it began honoring the "remote-part-ID" field sent by Asterisk and sends the *called*name to the Nortel. However, I still do not get the called name from the Nortel to

[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):

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

Hi All, I wish to connect * to a Cisco using a E100P board. When I load the driver I got this error message: -bash-2.05b# modprobe wct1xxp ZT_CHANCONFIG failed on channel 1: Function not implemented (38) /lib/modules/2.4.21-SAX/misc/wct1xxp.o: post-install wct1xxp failed /lib/modules/2.4.21-SAX/misc/wct1xxp.o: insmod wct1xxp failed Follows Cisco configuration: isdn switch-type primary-qsig

Hi all,as good? It would like to know if already they had had success, in the integration of the functions of asterisk, with one pabx legacy (Avaya)for that pabx avaya, could use the voicemail of asterisk. For sample, user of pabx avaya, it would have its calls directed for not attendance and busy, for asterisk and asterisk, it would send the same one for the voicemail. Best Regards Josu?

Hi, I am getting this message on the * console on my first pri span. Pri show span show it is down, and I can't make any calls from the span. Apr 27 07:40:23 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:23 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27

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, I have been pulling my hair out trying to get a Cisco MC3810 to interface my Asterisk box with a T1. I am able to make outgoing calls but incoing calls never reach my Asterisk box. The cisco give a fast busy when I try to call one of the DID's. When playing around with the dial-peers I can get the cisco to pick up the call, but then it forwards the call back to the ANI that is dialing.

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've been going back and forth with my telco for several days, trying different configurations to get a new PRI to come up. The bchannels are all up and the T1 is not in alarm status. The dchannel refuses to come up however. We've tried ni2, qsig, and now dms100 for the switchtype. The telco tech I've been working with says that he's been sending "reset all channels"

We've been trying unsuccessfully to connect our Meridian Option 81C to a TE110P via PRI. We've followed the directions in asterisk-meridian-a1.pdf (link on http://www.voip-info.org/wiki/view/Asterisk+legacy+integration), but it doesn't seem to work on our 81C (even though many, many users report it works very well on Option 11's). Has anyone had any success in getting the above

Hi all, We we send a server out to a customer with a PRI card I basically tell the PBX person that we want national, NI2, b8zs, D channel is on 24, and we are the CPE and your the NET and us a cross over cable. Allow extensions, local calls and long distance. on the surface I would have thought this would be enough for them to connect the digium card to the PBX and get calls working. The PBX

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, 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

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