search for: cs1000

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 and have the asterisk box send the calls back to the CS1000 to the correct extensions. Now I have a clean Asterisk install with noth...

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 n...

...c4-429ecb30-59f22c0-57e8 Call-ID: 2a67a2493820c3c153d0b87c7a97ceca@161.49.198.102 CSeq: 102 OPTIONS Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE Via: SIP/2.0/UDP 161.49.198.102:5060;branch=z9hG4bK6a2f6214 Supported: 100rel,sipvc,replaces User-Agent: Nortel CS1000 SIP GW: release=4.0 version=sse-4.00.31 Content-Length: 0 10 headers, 0 lines Destroying call '2a67a2493820c3c153d0b87c7a97ceca@161.49.198.102'

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...

...80d To: <sip:1715;phone-context=Exp_Net.ascom@ascom.be;user=phone> Call-ID: f8b5ad8-de6719ac-13c4-45630b31-5f87fa39-d37@ascom.be CSeq: 1 INVITE Via: SIP/2.0/UDP 172.25.103.222:5060;branch=z9hG4bK-45630b31-5f87fa39-63c2 Max-Forwards: 70 Supported: 100rel,sipvc,replaces,timer User-Agent: Nortel CS1000 SIP GW release_4.5 version_sse-4.50.88 P-Asserted-Identity: <sip:1649;phone-context=Exp_Net.ascom@ascom.be;user=phone> Privacy: none History-Info: <sip:1715;phone-context=Exp_Net.ascom@ascom.be;transport=udp;user=phone>;index=1 x-nt-corr-id: 000000b70e1423150b@0001af0807d6-f087c208 x-nt...

...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 on the intercom. It eventually times out and fails. Does anyone know how to setup either asterisk or the CS1000 to GIVE asterisk the proper signaling in this case so the call is "answered" and messages can be played over the intercom system. This is a PRI call into the CS1000 routed to a line level audio port. It appears to not be giving signallying that the route was established or the call wa...

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 for presentation? We can't possibly be the...

...9efa71-657b8da-2ed To: <sip:2828;phone-context=cdp.udp@qg.com;user=phone> Call-ID: 103548e8-c22da8c0-13c4-429efa71-657b8da-517@qg.com CSeq: 1 INVITE Via: SIP/2.0/UDP 192.168.45.194:5060;branch=z9hG4bK-429efa71-657b8da-2522 Max-Forwards: 70 Supported: 100rel,sipvc,replaces User-Agent: Nortel CS1000 SIP GW: release=4.0 version=sse-4.00.31 P-Asserted-Identity: <sip:3173241052;phone-context=+1@qg.com;user=phone> Privacy: none History-Info: <sip:2828;phone-context=cdp.udp@qg.com;transport=udp;user=phone>;index=1 x-nt-corr-id: 000000460c18100206@0001af0d4517-c0a82de1 x-nt-calling-id:...

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

...429efa71-657b8da-2ed To: <sip:2828;phone-context=cdp.udp@qg.com;user=phone> Call-ID: 103548e8-c22da8c0-13c4-429efa71-657b8da-517@qg.com CSeq: 1 INVITE Via: SIP/2.0/UDP 192.168.45.194:5060;branch=z9hG4bK-429efa71-657b8da-2522 Max-Forwards: 70 Supported: 100rel,sipvc,replaces User-Agent: Nortel CS1000 SIP GW: release=4.0 version=sse-4.00.31 P-Asserted-Identity: <sip:3173241052;phone-context=+1@qg.com;user=phone> Privacy: none History-Info: <sip:2828;phone-context=cdp.udp@qg.com;transport=udp;user=phone>;index=1 x-nt-corr-id: 000000460c18100206@0001af0d4517-c0a82de1 x-nt-calling-id: &...

...(Ludovic Bou?) > 5. Re: Gateway? (A J Stiles) > 6. Re: Can't register to Asterisk 1.6 with old Aastra phones > (Bob Kyeyune) > 7. Re: Gateway? (Eric Wieling) > 8. hello! (Rahul Pachauri) > 9. Asterisk QSIG doesnt send the calling name to Nortel CS1000 > (Danilo Dionisi) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 29 Apr 2013 12:51:42 -0500 > From: "David M. Lee" <dlee at digium.com> > Subject: Re: [asterisk-users] Asterisk 11.3.0 - Mask...

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: asterisk-users@lists.digium.com Subject:...

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 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

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