similar to: Touch Tones not working after call is made

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:

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

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

Is this a new bug, or am I doing something wrong here... ? Same config files as I have always been using, maybe there is a config file change needed? I have been running CVS for a while and just upgraded to 1.0.0 Now I get an RTP error with MGCP, MGCP dies afterwards. Sep 23 11:28:33 WARNING[98310]: rtp.c:711 ast_rtp_offered_from_local: rtp structure is null -- MGCP

-------------- next part -------------- An embedded message was scrubbed... From: Deon Rodden <drodden@webunited.net> Subject: Re: [Asterisk-Users] Strange Quality problems with Asterisk, Gentoo, Redhat and Kernels - /dev/dsp Date: Thu, 30 Sep 2004 09:05:39 -0400 Size: 5509 Url: http://lists.digium.com/pipermail/asterisk-users/attachments/20040930/289c69cc/dsp.eml

Config: AAH 2.2 Digium TDM card connecting to 3 x Telus POTS lines Polycom 501 phones pretty basic setup, working mostly just fine... When I dial a number such as: 96045551212 Telus automation will sometimes come online and tell me that the number I have dialled cannot be completed as dialled. If I hang up the Polycom 501 and redial the EXACT same number, it will work the second time. I

Good Morning, I'm having an issue with callerid display when calles are placed _from_ an mgcp device (8x8 ata w/mgcp firmware). Internally, there are several different sip devices and one mgcp device. Calls from any of the sip devices to any other device (sip or mgcp) have name/number displayed properly by the called party's phone. Calls from the mgcp device to any other device display

hi all I'm being spammed by these messages in the console (see below) and sound doesn't work with today's cvs. I rolled back a week, and it works fine. In addition to the sound problems, I had to enable inband dtmf squelch on the dilnk mgcp phones. if not, each pressed key was counted twice NOTICE[245776]: File chan_mgcp.c, Line 710 (mgcp_rtp_read): MGCP ast_dsp_process

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

*This message was transferred with a trial version of CommuniGate(tm) Pro* Hey all, Just noticed something that might be an issue. I have just made asterisk crash consistently by doing the following. I have a D-Link DG1102s running MGCP into asterisk and an extension *9 setup which dumps me into my inbound context to simulate calls coming in from my X100P. This usually works with no hassles

Okay, so I'm a noob. Asterisk looks very promising, so I say "thanks" and "good job" to all who contribute. My * test box is up and running with soft phones using IAX and SIP, so now I'm on to testing hard phones. I borrowed a couple ShoreTel 210 phones from somebody who had them on hand but they only support MGCP. I see that there's an mgcp.conf in

Hi, I have the following situation: I've installed Asterisk at Machine 1 (M1 - IP: 192.168.1.145) and X-Lite (X_lite-Xten-Win32-1103m.exe from www.xten.com) at Machine 2 (M2 - IP: 192.168.1.100) and Machine 3 (M3 - IP: 192.168.1.200). I need to catch the SIP and MGCP messages that will appear when M2 calls to M3 and vice versa. The SIP messages are working (I don't have problems with the

Hi All, I'm trying to hook up asterisk (CVS-HEAD-02/09/05-13:44:11 ) to a ADIT 600 via MGCP. Got it working really nice but now have a pretty bad problem: 1. When I perform a flash on the telephone, I usually get a second dialtone, but when I dial, dialtone doesn't break. If I flash back and forth a few times, it will eventually give me no dialtone.. here if I dial, it successfully

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

Hi, My system is like this currently: ATA-186 <-> *1 <-> IAX2 to Europe <-> *2 <-> i4l <-> voicemail at cell provider When I dial up to my voicemail at my European cell phone provider I can't press '#' to get into their menu. It seems like it just ignores any DTMF tones or doesn't get them. When I call a human on the other side of the i4l they

Guys. I have a te100p with unicall and an E1 and Im having problem with DTMF tones but the weird thing is, I only have problems sending the tones to certain phone numbers, anybody seen this behavior? Asterisk shows on the console the dtmf tone been pressed but seems the other side is not getting them, and this just happens with certain phone numbers, not all.. Any clues(tips?

Asterisk 1.4 We are experiencing an issue on transfers where no progress tones are heard by the caller: 1. Call from 1593 (SPA525G 0026998D2FFFF) to 1595 (SPA922 000B820AFFFFF). 1595 answers 2. From 1593 initiate transfer to 1597 (SPA508G 1CDF0F4AFFFF). 1595 hears MoH. 3. 1597 starts ringing and 1593 presses transfer again. MoH stops but 1595 hears no ringing When xfer is pressed and the

Greetings, I am PRIs connected to a Cisco 36xx gateway, which in turn connects to Asterisk via SIP. The problem I am having is that DTMF tones originated on the PSTN side are not heard on the SIP device. On the other hand, tones originating on the PSTN side are received by Asterisk when talking to voicemail or an autoattendant. >From the Cisco debug, I can see the Cisco sending NTE (RFC2833)

Help! I've been using mysql for cdr storage, I need to switch to MS SQL. I must be stupid or something but I cannot figure out how to setup the cdr_tds. I have FreeTDS configured properly, but my unixodbc is not working properly either... I'd be happy with either solution, but I'm in need of assistance. Luke Catranis

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):