similar to: question on nortel CS 1000 PBX and PRI connection to built in PA system

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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

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 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

All, I am connecting to a CS 1000 nortel PBX. I can call out, I have limited success with call in. I get debug traffic that a call is coming in but I get the message "Unable to create/find channel". I was expecting that incoming calls over the trunk would be handled from my sip definition and goto the nortel context. It is not. Below is the actual incoming call debug information. I am

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

We are currently working to trunk from a Nortel 81C to an Asterisk Server 1.4 running on Red Hat Linux. We have two PRI trunks which work with the exception of the clock slips, which is causing the Nortel to reset the PRIs once a hour. Thanks for any suggestions. 81C MSDL Asterisk Digium TE110P REQ prt TYPE adan dch 10

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

We would like to present a toll free CallerID when making outbound toll calls. In the past, when our PRIs were directly connected to a Nortel CS1000 we could do this, without issue. Now that the PRIs are front ended by a mediagateway facing asterisk, we can no longer do this. Is it possible to set the billing number via a SIP header and set what should be presented as callerid as another header

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

Hi I want to connect asterisk 1.0.9 ( kernel 2.6.8-2 debian )with TE110P and Nortel meridian option 11c release 25.40 with NTBK50AA card which is An E1 card. But the main problem is the first stage that no sync occurs the * card never syncs with meridian card I am using euroisdn, ccs , hdb3, crc4 , pri_net on asterisk And I am assuming that meridian is using same as it is

Actually, there was recently a bug fixed regarding multipart SDP parsing in chan_sip. That should have fixed the issue with CS1000s and SIP (among other things). I haven't actually tried it yet on my CS1000, but it should work. Regards, - Brad ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Jerry Geis Sent: Tue 12/26/2006 3:51 PM To:

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

Skipped content of type multipart/alternative-------------- next part -------------- Nov 21 14:17:47 VERBOSE[32580] logger.c: <-- SIP read from 172.25.103.222:5060: INVITE sip:1715;phone-context=exp_net.ascom@ascom.be:5060;maddr=172.25.96.48;transport=udp;user=phone;x-nt-redirect=redirect-server SIP/2.0 From:

I've googled / voip-info'd / searched until my eyes are blurry, but couldn't see the info I was looking for. I've turned here for help! Asterisk CVS head (9/7/04) Fedora Core 2 (updated to 2.6.6 kernel) DE405P (jumpers set to E1) I want to put asterisk in the middle of our current pbx (Meridian Option11) Currently the meridian has a 2MB pri EuroISDN card linked via a rj-45 into

Hello R Folks, I am new to R. I have been struggling to overlay a barchart with a xyplot together on one plot but did not get this worked out. Any help and idea are greatly appreciated. I attached R scripts for barchart and xyplot below and also data I used. What I am trying to do is just to put the barchart and xyplot together on one plot. Huapeng <<dispersal_infestation.csv>>

Hi all, I'm trying to insert a Rhino Ceros box equipped with a Rhino R2T1 dual-T1 card and running the latest version of Trixbox (2.4.2) between the central office and a Nortel Option 11. The switch at the CO is a DMS100. Basically, I'm taking the T1, connecting it to port 0 on the R2T1 card, and then connecting port 1 to the Nortel. (Actually a CSU and then the Nortel) We're running

I am getting the following information and asterisk 1.0.7 is not answering the call. Any ideas? jerry ------------------ Sip read: INVITE sip:2828;phone-context=cdp.udp@qg.com:5060;maddr=161.49.198.102;transport=udp;user=phone;x-nt-redirect=redirect-server SIP/2.0 From: <sip:3173241052;phone-context=+1@qg.com;user=phone>;tag=c22da8c0-13c4-429efa71-657b8da-2ed To:

Well I am entering into a realm that I don't know. 3 sites with Asterisk 1 site with Nortel Asterisk/Sip calls working fine between the 3 sites. Asterisk to Nortel set calls working fine. (call comes from asterisk to nortel and rings telephone, people answer and talk happens, hangup call clears) Nortel to Asterisk. Set on Nortel gets a busy signal. Any suggestions on what to look

Is anyone successfully using this phone with Asterisk? There is a lot mentioned about CISCO but nothing about Nortel... Alex.