search for: dlcx

...a 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, 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@t...

...CP/aaln/1 at 020202020202-1 and SIP/mgcp-out-09d998c0 -- Resetting interface aaln/1 at 020202020202 == Spawn extension (mgcp, 0141909872, 2) exited non-zero on 'MGCP/aaln/1 at 020202020202-1' -- MGCP handle_request(aaln/1 at 020202020202-1) ast_channel already destroyed, resending DLCX. -- MGCP handle_request(aaln/1 at 020202020202) set vmwi(-) - Receive incoming calls through SIP PROXY : OK too, but as with the outgoing calls, the interface seems to self restart : -- Executing Dial("SIP/XXXSIPCALLID", "MGCP/aaln/1 at 020202020202") in new stack --...

...1.137 2427 MGCP 200 14831 OK 4 0.008150 192.168.0.5 2727 192.168.1.137 2427 MGCP RQNT 7 SHOR_001049007E83@[192.168.1.137] MGCP 1.0 5 0.008186 192.168.0.5 2727 192.168.1.137 2427 MGCP AUEP 8 SHOR_001049007E83@[192.168.1.137] MGCP 1.0 6 0.008267 192.168.0.5 2727 192.168.1.137 2427 MGCP DLCX 9 SHOR_001049007E83@[192.168.1.137] MGCP 1.0 7 0.009226 192.168.0.5 4569 192.168.1.135 4569 IAX2 IAX, source call# 2, timestamp 61056ms HANGUP 8 0.012188 192.168.1.135 4569 192.168.0.5 4569 IAX2 IAX, source call# 10281, timestamp 61056ms ACK 9 0.015216 192.168.1.137 5004 192.168.0.5...

...00-S1 MGCP 1.0 X: 0fa02584 R: hu(N), hf(N), D/[0-9#*](N) S: cg to 10.0.2.6:2427 -- MGCP mgcp_hangup(MGCP/aaln/S0/SU0/0@MRP200-S1-1) on aaln/S0/SU0/0@MRP200-S1 -- Delete connection 5 aaln/S0/SU0/0@MRP200-S1-1 with new mode: sendrecv on callid: 6b49143d0fa02584 Posting Request: DLCX 11 aaln/S0/SU0/0@MRP200-S1 MGCP 1.0 C: 6b49143d0fa02584 X: 0fa02584 I: 5 to 10.0.2.6:2427 -- MGCP Asked to indicate tone: ro on aaln/S0/SU0/0@MRP200-S1-1 in cxmode: sendrecv Posting Request: RQNT 12 aaln/S0/SU0/0@MRP200-S1 MGCP 1.0 X: 0fa02584 R: hu(N), hf(N), D/[0-9#*](N) S...

hello, my questns are about few * functionality. 1) how can I make call tranfer. Not call parking. If I'm talking with some one a I want to tramnfer call to the another extension, to the other person. 2) how can I make call confernece. Not Meetme If I'm talking with some one and I want to join another person to our talk . I haven't found this in any manual :( hudecof -- mail:

We are using an MGCP configuration. There seems to be some incompatibilities between our Netergy T2 VOIP chip and Asterisk. This is how our Vendor sees it: Here's what I see. 1. The first call is initiated. (CRCX) The interesting thing here is that the CA (Call Agent) tells us to go directly into sendrecv mode which means that we start streaming audio immediately. All other CAs that

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 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 =