similar to: MGCP and Cisco ubr924

Hello, I have just received an MGCP Phone for test purpose and I can't place a call from my MGCP Phone. I can call my MGCP phone from a SIP Phone. Here is my mgcp.conf: ; ; MGCP Configuration for Asterisk ; [general] ;port = 2427 ;bindaddr = 0.0.0.0 ;[dlinkgw] ;host = 192.168.0.64 ;context = default ;line => aaln/2 ;line => aaln/1 [192.168.10.10] host = 192.168.10.10 context =

Hi. I'm new in Asterisk with MGCP. I set up a MGCP user agent and start asterisk with the next configuration files. '--------------- extensions.conf ---------------------------------------------------- [general] static=yes writeprotect=yes [globals] ap1 => mgcp/aaln/ap200@64.76.148.186 [macro-apl1] exten => s,1,Dial(${ARG1},30,Ttmr) ;exten => s,2,Voicemail(u${MACRO_EXTEN})

I strange error messages when using mgcp and ata186 . This session is simply dial into 600 demo extension - echo test ... Handling request 'NTFY' on aaln/1@10.0.1.19 Transmitting: 200 29 OK to 10.0.1.19:2427 -- Endpoint 'aaln/1@10.0.1.19-1' observed '0' -- MGCP Asked to indicate tone: on aaln/1@10.0.1.19-1 in cxmode: sendrecv Posting Request: RQNT 306

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:

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 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

I'm trying to use a 4 port ATA that was designed for use with MGCP 1.0 / NCS 1.0 -- is the channel_mgcp compat with NCS -- anything I can do to make it compat?? This is what happens - below *CLI> mgcp reload Reloading MGCP == Parsing '/etc/asterisk/mgcp.conf': Found Use EXIT or QUIT to exit the asterisk console == MGCP Listening on 10.1.22.39:2427 == Using TOS bits 0 mgcp

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

Hi, I have some problems transfering call from phone to phone with my Asterisk. When I dial Flash I can hear for half a second the dial tone, but it stops suddenly. The other phone hear the on hold music and pressing flash key another time I get back to the previous channel. On the asterisk consolle seems to be all ok, this is whant I can read: asterisk1*CLI> -- Swapping 0 for 1 on

see below I tried to call 98013356 from the following phone (from mgcp.conf) [iptlf03] host = 192.168.33.3 context = default inbanddtmf = 1 callerid = 22545062 line => aaln/1 Console output: == Spawn extension (capiring, 9988001133335566, 1) exited non-zero on 'MGCP/aaln/1@iptlf03-1' -- MGCP mgcp_hangup(MGCP/aaln/1@iptlf03-1) on aaln/1@iptlf03 -- Delete connection 4

I know this is not the most appropriated list to this, but I will try: Does anyone know what is the criteria to the generation of the transaction identifiers in MGCP? I mean, are they generated by a randomic method? I'm using Asterisk and MGCP eyeP Phone and observed that the RSIP and NTFY (methods created by the gateway) use high values in the transaction identifiers, while the RQNT, AUEP,

Hey list, Does anyone have a current working config example of a cisco ubr924 and * ? I think the 924 only supports MGCP. I want to get VoIP on this device, I was wondering if anyone has already tackled the problem, if not, I'll go in blind :) Thanks Duane Cox -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

OK, I've looked till im blue. I have a couple of the Cisco ubr924 cable modems that I wish to set up with asterisk using MGCP,  but, the IOS that I have seems to not support this. <br />The IOS that I have is ubr920-k1o3sv4y556i-mz.121-4.bin  I have got one  for 12.2(19)c however, it doesnt support MGCP and the SGCP support also isnt available.<br /><br />Where can I get a

Hi, in following of a recent discussion I got to work on MGCP with the Cisco ATA186 again, and got it to work very nicely. However, there is a little thing with transfers I would like to get comments on: Call comes in from PSTN and goes to an ATA186 (MGCP) Call is answered and then, using flash, transferred to another extension If the extension is available, there can be an announcement 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

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 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 all, I'm looking for some help to try to understand why my CPE doesn't work good with Asterisk in MGCP. Here is what I want to do : - Register a TECOM AH4021 on Asterisk in MGCP with the following profile in mgcp.Conf : [general] port = 2727 bindaddr = 10.95.20.1 disallow=all allow=g729 allow=alaw 020202020202] context=mgcp host=dynamic canreinvite=no dtmfmode=rfc2833 nat=yes

Hey All, Thought I'd take a bash at trying to get Festival to work here on my lab system with the aim of using it to create our IVR menu prompts. I've spent most of the afternoon searching through the Wiki, the Festival website and Google and I've got a couple of questions. First one is that the 'Asterisk+festival+installation' page on the Wiki mentions the RedHat 9 RPMs