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...terisk 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, CRCX and DLCX (by the call agent) use low values for it. I hope I could explain my doubt! Here are some examples: RSIP 18696 aaln/1@tramontina MGCP 1.0 RQNT 8 aaln/1@tramontina MGCP 1.0 AUEP 9 aaln/1@tramontina MGCP 1.0 RSIP 27232 aaln/1@tramontina MGCP 1.0 RQNT 3 aaln/1@tramontina MGCP 1.0 NTFY 27219 aaln/1@tramontina MGCP 1.0 CRCX 2 aaln/1@tramontina MGCP 1.0 DLCX 26 aaln/1@tramontina MGCP 1.0 Thanks, Leonardo J. Tramontina ---------...

...ems to be allright (also the "extensions.conf"), but the messages are not working... Does anyone have an example of "mgcp.conf" and "extensions.conf"? Is there another file to config? And at X-Lite, is it necessary setup something else? Thanks in advance! Leonardo J. Tramontina -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041126/eaf414dd/attachment.htm

Hi, I've just got a TE110P card and installed at Asterisk. I configured zapata.conf, according to www.digium.com/index.php?menu=configuration, but the following error is happening: ... ... ... [chan_phone.so] => (Linux Telephony API Support) == Parsing '/etc/asterisk/phone.conf': Found == Registered channel type 'Phone' (Standard Linux Telephony API Driver)

Sirs, According to RFC 2705 (MGCP), these are the parameters that are used in the transactions: ReturnCode, Connection-parameters <-- DeleteConnection(CallId, EndpointId, ConnectionId, [Encapsulated NotificationRequest,] [Encapsulated

Hi, I've created a test at "extensions.conf" like this with 3 steps: ; When dial 5555, get the first available channel and dial do 482343400 exten => 5555,1,Dial(Zap/g1/482343400,5,rt) ; When dial 5555, get the channel 20 and dial do 482343400 exten => 5555,2,Dial(Zap/20/482343400) ; Go to Voicemail 1234 exten => 5555,3,Voicemail(u1234) I've tried using just the

Does anyone know some free MGCP softphone? Nowadays I'm using one from eyeP Media, but it is trial for 30 days and it's expiring... Any ideas? Thanks, Leonardo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050108/b393104c/attachment.htm

Sirs, I have a question about CreateConnection (CRCX) at MGCP. For example, I have the phone number "5220107" and want to make a call for it using MGCP through a media gateway. How can I proceed? I know the command I must send to the media gateway should be like this: CRCX <trans_id> <endpoint> MGCP 1.0 "C", "L", "M" and "X"

Sirs, I have the following situration: 1) AudioCodes Stretto 2000 media gateway running MGCP 2) E1 Digium card at a PC with Asterisk 3) My application running as Call Agent (CA) from Stretto 2000 | My app |----------| Stretto 2000 |----------| E1 card + Asterisk | As my application is the CA of Stretto 2000, everything it sends (RSIPs, acks, etc.) my app answers. And everything I send,

I will implement a SIP application and I'm considering using Asterisk for mixing the media streams (audio). Does anybody know if Asterisk supports or contains a RTP mixer? If so, how to use it? Just to be a little more clearer: I will send to Asterisk more than one RTP stream and they must be mixed. The result must be a single stream to be forwarded to a SIP phone or to the PSTN. Thanks,

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