similar to: RE: mgcp <-> h323 problem

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'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

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, 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

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

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})

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, 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, After spending some time trying to get a DG-104S working behind NAT, I finally found the problem. I made the incorrect assumption that nat=yes in mgcp.conf works just like sip.conf. The channels within a gateway are treated more closely to zap channels than sip channels (from a .conf standpoint). What this means is that you have to put nat=yes BEFORE any subchannel definitions: This

I am have trouble getting caller id to work here is my current mgcp.conf am I missing something? ========================================================= Mgcp.conf [general] port=2427 [Egraph-1] ;dlink 104s-1 host = 12.151.207.2 context = local line => aaln/1 callerid = "jim's office 1" <321> line => aaln/2 callerid = "jim's office 2" <322>

I have a DG-104S (which I reset to factory settings, it's DHCP'ing off my network, plugged into the WAN port). The system comes up, and I through the web browser set under Call Agent IP Address to: Notify Entry: dlinkgw@[192.168.1.1]:2427 (192.168.1.1 is the * server) I have RGW Name: and DNS IP addres the DNS IP of the MGCP box and DNS State disabled (not sure what to set it to) --

I am have trouble getting caller id to work here is my current mgcp.conf am I missing something? ========================================================= Mgcp.conf [general] port=2427 [Egraph-1] ;dlink 104s-1 host = 12.151.207.2 context = local line => aaln/1 callerid = "jim's office 1" <321> line => aaln/2 callerid = "jim's office 2" <322>

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 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 ...

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

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

> 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'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