similar to: Cisco 2610 as an MGCP gateway

Can anyone tell me what MGCP error that I'm getting means? The hardware is a MRP200 in a Cisco 7750 PBX. (Its a FXO blade with 2 slots, first one has a 4 port FXO card and the second has 2 port FXO card. It recognises those correctly, at least to the point of this error.) MGCP Debugging Enabled MGCP read: NTFY 13 aaln/S0/SU0/0@MRP200-S1 MGCP 0.1 X: 1adace42 O: L/hd from

I repost a message I put a week ago: I have a Cisco IAD 2431 which has MGCP protocol; I cannot make it to work againts Asterisk; at least there is some MGCP conversation between them but when I offhook a phone attached to IAD I get no tone at all. As anybody managed to get working Asterisk against an MGCP Cisco gateway ? Which MGCP version should I use ? Also I recently

Hi! I have strange problem, I hope it's just a configuration problem, but maybe not. I'm trying to make a call between a MGCP gateway and Cisco 5300 talking SIP. Everything is fine except that audio is one way (from 5300 to MGCP gateway only). It seems that during reinvite Cisco gets confused by session ID and version and excpects RTP stream on different port. The call flow on SDP

Does anybody has the expirience configuring Asterisk with Cisco ATA 186 MGCP firmware ? I have Cisco software v3.1.1 atamgcp (Build 040629A) Asterisk 1.0-RC1 On ATA i only put domain test. mgcp.conf looks like this [test] host = 192.168.195.55 context = default line => aaln/2 line => aaln/1 Asterisk CLI shows this: Jul 31 16:05:40 WARNING[135449600]: chan_mgcp.c:485

I've been trying to figure this one out for a while, but to no avail. I have my cisco ubr924 setup for MGCP with Asterisk as the call-agent. I have manually registered the endpoint in mgcp.conf. When I pick up the phone, I get no dialtone and debug shows errors. IOS on the ubr924 is 12.2. Any help is appreciated. from mgcp.conf: [ubr924] host=65.37.86.203 context = from-sip (just as a

Hello: I want to receive calls from my SIP proxy and re-route them to one of the analog lines on my Cisco VG200 ia MGCP and Asterisk. Inbound SIP calls will arrive with the five digit called number preficed with an "m" by the proxy. I'd like to them match these calls against a rule like exten => _mXXXXX,1,Dial(MGCP/aaln/1@128.100.10.11) . This however results in an error.

I'm trying to link up Cisco MGCP-enabled router (residential gateway) with Asterisk, and it looks like some sort of protocol mismatch, could it be MGCP 0.1 vs 1.0? Look at this (x.x.x.99 is the router, x.x.x.98 is Asterisk): MGCP read: NTFY 2 aaln/0@voip-gw1 MGCP 0.1 X: 0 O: hd from 192.168.154.99:2427MGCP read: NTFY 2 aaln/0@voip-gw1 MGCP 0.1 X: 0 O: hd from 192.168.154.99:2427Verb:

I have a fxo endpoint installed in a Cisco router. I would like in my dialplan to get an extension call a telephone number through that fxo endpoint. Since with zaptel channels it is done like: exten => 0999,1,Dial(DAHDI/2-1/111) --> being 111 the phone number I want to call. I thought that for mgcp it would be the same, and I did: exten => 5200,1,Dial(MGCP/aaln/S0/SU3/0 at

Anyone has a working CARD.XML for cisco MGCP phones? The one on the cisco site is old and it's not working with the new firmwares. Thanks Nico -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060505/f7b965bb/attachment.htm

Hi all, I am looking for a software package (free or not), or an inexpensive hardware device, which can route calls between an H.323 network and an MGCP-based voice service. Unfortunately, I believe (based on documentation and other forum posts -- I have not looked at the code) that Asterisk can only act as an MGCP Call Agent. Has anyone added code that can talk the 'slave' side of the

Hi all, i am using CVS-HEAD-08/13/04-12:00:00-BRI-stuffed-0.1.0-RC4a with Cisco ATA-186 3.1.1 atamgcp We are used to make an special ;) blind transfer like (Flash)Number(Hangup before anyone answers or ring). Then * crashes (see below) if the man in the middle is an cisco-ata-186-mgcp If one waits until the last one rings, then hangup, everything is fine. If one waits until the last one

Hello asterisk community, I have successfully set up asterisk as a SIP PBX and now would like to connect to the outside world using a Cisco 2600 with VIC-BRI as an ISDN gateway. This works already in the lab, but I have security concerns before conecting the gateway to the internet. I currently don't know exactly what VoIP services the Cisco runs by default besides SIP (H.323, MGCP, ...)

Hello, I am using a Cisco-2,811 router with PRI as a gateway between Asterisk and Nortel TX-1. I had problems with name transfer and with the help of Cisco support I've fixed it. Enclosed here are the definitions needed for it. BTW, Cisco's CCM is using MGCP thus the Q.sig is handled by CCM. Here I am using SIP so the router must decode/encode the Q.sig. The Nortel should be defined

If you're using MGCP, we need your help. There's a patch in bugs.digium.com that needs testing by the community. Please spend some time testing and adding your comments to the bug tracker. The author writes: ------------------ I'm trying to make work Asterisk against a Cisco IAD2431 with chan_mgcp. Since chan_mgcp assumes the Line package is the default which is not the usual with

I have a Cisco ATA-188 with two POTS phones and latest stable cvs. In any situations with call waiting (existing connection and calling again) the second call cause both calls to drop. This is the same for "internal" extensions and from external (ZAP and SIP). It seems to be a "502 - The transaction could not be executed, because the endpoint does not have sufficient

Hi All, I just setup an asterisk system equiped to with a 4 FXO port board to interface it with my phone system (Inter-Tel). I realize that I'll be running our of ports once I start showing what can be done with it. My Inter-tel system supports MGCP connectivity to a gateway. Is there any planned development to make chan_mgcp work as a gateway? Another alternative, was to rig * and the

Hello, I try to place a call between CISCO IP phone 7905 (h323) and an analogue telephone on DLINK DG-104S (mgcp). Call setup is handled ok - both way, but after connection of the call I get both site no-audio, even codec are Set to alow-64 both. The call is terminated with this debug output - Didn't get a frame from channel: MGCP/aaln/1@dlink-1 and is Hanged up. Jan 18 20:32:55

Hi folks I seted up the asterisk with an active ISDN B1 AVM Card (german vendor) and it works fine, various SIP clients (IP fon snom, xlite under MacOSX) and also incoming and outgoing connectins. Ok. No problem. After that I configured a CP7940G with a MGCP IOS. It's connected to the asterisk too via switch. No NAT deivce between. And of course, I can call out to PSTN and also to one of the

I'd like to use the VoIP service from Free with Asterisk, but am having a couple of problems. Here are some details: ADSL from Free T?l?com comes bundled with VoIP and TV services. Most users access the VoIP via the supplied Freebox, which is an integrated DSL modem, router, ATA, and media player. It is of course possible to connect the Freebox to Asterisk via an X100P or other FXO

Hi, For MGCP users. Is there any success stories with any MGCP IAD vendor. I?m trying to find an IAD which works with Asterisk. I?ve tried the Cisco IAD 2430 without success; but SIP on this IAD works but it?s limited (no authentication, no notify messages, etc) and with higher density IAD (16 or more ports) it?s nice to control using MGCP. Any information will be apreciated ! Thanks. --