similar to: Different MGCP issues

Hi I have kind of an odd problem. When dialing in from an outside line via a TE410P card it seems like MusicOnHold and MP3Player doesn't work properly (for me anyway). The remote end which is calling * doesn't hear the music but just keeps ringing. But if I insert a Playback("file_which_dont_exist") just before the Moh or MP3Player I can hear the music. Actually I observed the

Hi everybody I've just installed a new Redhat 8.0 and configured it with Asterisk, zaptel and libpri. Afterwards I installed a TE410P and configured this as well. But when starting Asterisk I get the following error message: ------------------------------------------------------- -- Registered channel 1, PRI Signalling signalling ..... -- Registered channel 15, PRI Signalling

Hi I have an Asterisk installation with some SIP and MGPC devices, and I also have a TE410P on a E1 line. If I make an outside ISDN data call to asterisk the phone rings as usual and if I answer it, I just hear some clicks. I've read that the D-Channel has information about the call, if its voice or data. Is it somehow possible to end/ignore this call already before it is ringing?

Hello * users I've have a rather disturbing problem, which I don't know how to debug or how to solve, but first a brief description of the set up. One Asterisk server with a TE410P card installed (first line used on this only), and a number of Wellgate 3504A (4 port FXS devices with SIP firmware). There is no connection from the Asterisk server to the outside world or any other

Hello * users Sorry I forgot to send the mail in plain text the first time... I've have a rather disturbing problem, which I don't know how to debug or how to solve, but first a brief description of the set up. One Asterisk server with a TE410P card installed (first line used on this only), and a number of Wellgate 3504A (4 port FXS devices with SIP firmware). There is no connection

Hi there How do I change the dialplan runtime, if I for example wants all calls on the main number to be answered by a voicemail (when it is out-of-office hours). I want to be able to change the configuration by pressing a DTMF combination e.g. *82. Can't figure out whether it is necessary to change contexts or how to do it. I have read a lot of examples and config documentation, but I

Hello fellow * users Hope this isn't a stupid question; I've done my research but could not find a proper answer. I have 8 different destinations which I want to match. The numbers are: ###### 00 ###### 20 ###### 30 ###### 40 ###### 15 ###### 35 ###### 12 ###### 44 Right now I've solved it by doing this: exten => _######[0234]0,1,HangUp exten => _######[13]5,1,HangUp exten

I must be getting something wrong about this call pickup. In zapata.conf I have just the default callgroup=1 and pickupgroup=1. If I call from my mobile to * and then try to dial *8 from any other phone than the one which is ringing I just get a "Nothing to pick up" answer on my * console. I also have experimented with those parameters in sip.conf but are not aware of exactly where to

Hi there I have a Welltech Wellgate SIP device and I want to be able to do a supervised transfer. I've read that in order to do that I have to use flash hook. The problem is just that I can't flash hook with this device. I'm in contact with the developer of the SIP device but don't know what to tell him in order to get him to fix this. What is happening when you flash hook, I

Hello there Yesterday I updated my pwlib, openh323 and Asterisk from CVS. After making "clean opt" in pwlib and openh323 and make "clean install" in Asterisk i get an "Undefined symbol" error when I try to start Asterisk. As far as I can see its when loading the h323 channel driver the error occurs. Do I have to update other things as well, by reading the various

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

Hello all I've discovered that SIP channels sometimes get stuck in *. I've read some posts from Fri 29 Aug 2003 which mentions this issue, but there doesn't seem to be any final answers I don't know if this is related to the 0001604 bug? Below is a list from one of the incidents: I know the (d) means that it is scheduled for destruction but the 10.1.1.45 channel hasn't

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

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,