similar to: connecting to nortel CS1000 (half way there)

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

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:

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

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'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

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

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

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:

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

Hello, I'm having this issue on my pbx, it appears that asterisk is refusing the codecs that my providers is proposing. My trunk configuration is: --- username=5x5x7x9x0x3 type=friend secret=CRcxn7sqwm qualify=yes port=5060 insecure=port,invite host=sip.txtxlxoxp.it fromuser=5x5x7x9x0x3 fromdomain=sip.txtxlxoxp.it disallow=all context=from-trunk allow=alaw --- A typical invite from my

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

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*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

Hello all, My name is Danilo and I have a problem with the ISDN. I hope I have the wrong section. =P I have a CS1000 Nortel central with release 5.50. This central is attached to an Asterisk server with Sangoma PRI ISDN. I need to read the headers of ISDN and comes running from Nortel to Asterisk. How can I read them? Thank you, Danilo. --

Hi, I need make a VPN between a FreeBSD and a Nortel... The IPSec of FreeBSD is compatible for this ? I will have that to use racoon to make ISAKMP or I can make without it? -- [ Diego Linke - GAMK ] System/Network/Security Administrator E-Mail/Site: gamk@gamk.com.br - http://www.gamk.com.br Public Key: http://www.gamk.com.br/gamk.asc Phone Number: (+5541) 9967-3464

Hi, Was wondering if anyone had any tips or experience in getting a Nortel CS1K and Asterisk 1.4.19 to talk to each other via NRS? So far I've gotten asterisk to place calls to the CS1k via the NRS, however calls originated by the CS1K get rejected by the NRS with a 404 Not Found message. If I take the NRS out of the equation by replacing the IP address of the NRS in the CS1K with that

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

I am connected with 5ess to a nortel switch. Running dahdi 2.1.04 and libpri 1.4.7 and asterisk 1.4.22 Everything is working for a short time. The other side reports back to me that he is getting slip-r errors from the TE122P card. After to many errors (like 11) the d channel shuts down on their side. Any idea what to do about this? I have not ran into it before. Jerry

Hi all, I have asterisk 1.4.26 (and I tried 1.4.29) connected PRI all 23 lines and for the most part everything works. Dialing out on 23 lines to phones works fine. I have to use the Local channel to call the intercom system (from call files). If I only call 1 intercom system at a time so it uses DAHDI/1 everything seems to work as I can call all 8 intercom systems and play a message. The