similar to: seeking H.323 <-> MGCP (User Agent) gateway

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:

Hi, I havn't actually tried this yet, but would it be possible to use asterisk as a mgcp <-> h.323 translator? For example, I have mgcp service from Next Gen telephone company. But i only have a h.323 phone. Would there be a way to the mgcp signalling to hit asterisk, and then have it fire the call out h.323? And vice versa? Just brainstorming. Sean Watkins

Hello folks, I'm trying to get an 8x8 DTA-310 running mgcp to work. I get no dialtone & can't get it to ring. My mgcp.conf says: ; ; MGCP Configuration for Asterisk ; [general] port = 2427 bindaddr = 0.0.0.0 [172.16.2.25] host = 172.16.2.25 context = default line => aaln/1 And here's the interesting bits of extensions.conf: [globals] ... TRUNK=H323/BYEXTENSION@pstn_gw ...

HI i am trying to register mgcp gateways(Polycom 501, 601) to asterisk as a call agent, mgcp gateways are not registering to the call agent. Please help me on this if any one knows how to congigure the mgcp.conf on asterisk as well as an MGs. The following are the details of mgcp.conf on asterisk. mgcp.conf [general] port = 2427 bindaddr = 0.0.0.0 [0004f205c258] //MG MAC Address host =

I'm looking for config examples to use a Cisco 2610 with an FXO card as an MGCP gateway for my Asterisk server. I would like the cisco config along with the Asterisk MGCP config. This is a test environment and this is the only equipment I have at the moment. I do have the phones also. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Philip, Unfortunately, * speaks MGCP only as the Call Agent, rather than as the Media Gateway. MGCP is a master/slave protocol, and it would take some effort to make * work as the slave. I have the same problem: Free Telecom here in Paris includes MGCP service with their DSL. You can call any fixed phone in France at no charge! Rates to mobiles and international are quite aggressive, too.

Hey, I have a voip provider that uses mgcp and I would like to connect that provider to my asterisk. Anyone succeed in doing this? Thanks, David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060218/aea2042a/attachment.htm

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

I am sure Primus has a SIP platform because we have played with it. We managed to use it on MSN's SIP phone as well as couple Zultys ZIP2x2 hard phones. Their PC-Phone app is also a SIP soft phone. If you are registering to sip.iprimus.net then it is definitely their SIP platyform not MGCP. David >>> asterisk-users@eol.ca 1/21/2004 6:39:34 AM >>> I'm not sure Primus

If you look at the specs on the Dlink box that Primus gives you, you will see that it is SIP. I am sure Primus has a SIP platform because we have played with it. We managed to use it on MSN's SIP phone as well as couple Zultys ZIP2x2 hard phones. Their PC-Phone app is also a SIP soft phone. If you are registering to sip.iprimus.net then it is definitely their SIP platyform not MGCP.

Hi folks, The obligatory newbie disclaimer: "Hi, I'm new to Asterisk and I have a couple questions..." OK, now that that's over with: I've just started working for a small CLEC, and I'm trying to sell * to my boss as a replacement for some expensive/inflexible/closed-source software he's been using to provide residential dialtone with for a couple years now.

I've got a ShoreTel 210 MGCP phone drops calls. My packet capture indicates that the phone may be trying to renew its registration with *, but reports Restart Method of Disconnected (frame 2), then * seems to take that as a sign that it has lost the connection and closes things down. The phone, meanwhile, seems to think it can continue the conversation until a few ICMP "port

Hi, I picked up a router with 8 voice ports that supports MGCP, but it has several options that I am not familiar with or do not seem apparent in the mgcp.conf. Enter the default IP address for the Notified Entity: [0.0.0.0] Enter the listening port of the Notified Entity: [2427] Enter the IP address for MGCP signalling (Data): [192.168.0.210] Enter the local port for MGCP signaling (Data):

Hey guys, can you shead some light on this? I will copy my mgcp.conf and post below, but here is the problem. I can't get call waiting to work with my MGCP device. I already have one call going, and I can hear the second call come in, I flash over to it, but all I get is a dial tone, * puts the 1st call on mute/hold, but I never get the second, and it terminates. I flash back over and pick

Hello I have a problem with my MGCP voice gateway. I use D-Link DG104S Boot PROM Version 3.0B38-D Firmware Version 3.0T86-D I tried asterisk v 0.7.2 and I am using latest CVS version now. When I dial a number very fast, or when I use a redial function, my asterisk receives coupled digits. My co-worker called number 245005111, these are a few lines of my debug. The identifier of first digit

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

> Hello all, > Can someone help me on the problem which I have on MGCP phone test . I test mgcp - asterisk- zap. But I got several NOTICE message from rtp.c. > NOTICE[20501]: File rtp.c, Line 221 (process_rfc3389): RFC3389 support > incomplete. Turn off on client if possible > > -- Endpoint 'aaln/1@VG101-1-1' observed '9' > NOTICE[20501]: File rtp.c,

I have 1 IP phone (Swissvoice IP10S) and 1 POTS phone. When I dial the number for the IP phone off the POTS phone, the IP phone rings. But when I pick up the handset on the IP phone, I get a busy signal and this message on *: Aug 23 09:38:57 NOTICE[1142106560]: chan_mgcp.c:2243 handle_response: Terminating on result 502 from svip10@00059002042b-1 Here is the entire session. svip10 is the 1 and

Hello, I`m try to make the attended transfer work Dlink DG-104S via FLASH, when somebody calls my phone I pickup and press flash to get a second line to call another extension. When I press flash I hear no dialtone, and only a long and then small beep. When I try to dial digits I hear again those long+short beeps, but the extension dialed is not ringing. If I pres flash again I get back to

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