similar to: need help to setup a sip trunk between a Nortel CS1000 and asterisk

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

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

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:

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

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

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

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

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:

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'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

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

> From: David Hopwood <david@bl...> > [image removed] Re: trusted computing > 2004-10-18 19:24 > Tim Freeman wrote: > > > not about Xen in particular, but as a side note, because I think some > > people are interested in trusted computing and virtualization? If > > you"re not, sorry for the intrusion! > > > >

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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. --

I've noticed a problem with calls to Hangup when talking to my Norstars over channelised T-1 E&M trunk lines - it's been present since I started to fiddle with Asterisk last December and it's still present in 'Asterisk CVS-HEAD-08/13/04-10:37:13'. Specifically, when a call is connected to Asterisk from the Norstar DTI card to my T100p I get the following conditions

I currrently have Asterisk 1.2.8 with a TE110P Zaptel card tied to our Nortal DMS-10 switch via T1. This T1 trunk is configured for Feature Group D MF. The purpose of this is for LD delivery to a VoIP LD provider as a LD choice for our customers off of the DMS-10. When I sieze a channel on the zaptel, I get a spill of digits from the switch that is the called number, and I can see that in

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

Has anyone had any luck getting an asterisk box to talk to a Nortel C15K softswitch? Or any Nortel "sip" products? I've been playing with it for several days and can't seem to pass calls either direction. I know that whike the Nortel says the C15K speaks SIP, it really speaks nortel's implementation of SIP, but I thought I could get it to at least pass simple calls back

Dear List, Need to know if anyone on this list has had any experience with using the Nortel BCM 50 for Call Account Reporting using an IP connection to a Linux / Asterisk interface? Presently, I have a BCM 50 installed that uses a local Lenova Small Form Factor PC with a windows XP / os that quit reporting because of a up-grade to the Nortel reporting software. Nortel support is now telling