<html><head><style type="text/css"><!-- DIV {margin:0px;} --></style></head><body><div style="font-family:tahoma,new york,times,serif;font-size:10pt"><div>Hi list, I am trying to make work to the parking of calls with
openser and asterisk, which I want to do is when two UAC are in the
middle of a call one of them can transfer a call with *700 and this is
sent to asterisk to the extension by default...<br><br>the problem here is that when I make the transfer to the extension 700 the asterisk it doesn't return it to the extension that I originate the transfer, the call it returns to the extension in delay ..<br><br>I can see when I make the transfer in the SDP that the openser puts me <span style="font-family: monospace;"></span>c=IN IP4 0.0.0.0 , but asterisk doesn't return the call to the extension that I originate the transfer<br><br><pre>U +2.857758 192.168.10.40:5060 -> 192.168.10.1:5060<br>INVITE sip:*700@192.168.10.1:5070 SIP/2.0 <br>Via: SIP/2.0/UDP 192.168.10.40:5060;branch=z9hG4bK1a7bc3cdf6e22598 <br>Route: <sip:192.168.10.1;lr=on;ftag=0d380f28344df9f2> <br>From: "Ventas" <sip:112@192.168.10.1>;tag=0d380f28344df9f2 <br>To: <sip:*700@192.168.10.1>;tag=as5aea7a9e <br>Contact: <sip:112@192.168.10.40:5060;transport=udp> <br>Supported: replaces, timer, path
<br>Referred-By: <sip:120@192.168.10.38:5060> <br>Proxy-Authorization: Digest username="112", realm="192.168.10.1", algorithm=MD5, uri="sip:*700@192.168.10.1:5070", nonce="495ac4ad695509e755aba895780497e8116e6353", response="40021b3138cbfefcd079505a55c6043f" <br>Call-ID: f69f46f8461d45e0@192.168.10.40 <br>CSeq: 55344 INVITE <br>User-Agent: Grandstream GXP2020 1.1.6.16 <br>Max-Forwards: 70 <br>Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE <br>Content-Type: application/sdp <br>Content-Length: 352 <br> <br>v=0 <br>o=112 8001 8002 IN IP4 192.168.10.40 <br>s=SIP Call <br>c=IN IP4 0.0.0.0 <br>t=0 0 <br>m=audio 5006 RTP/AVP 0 18 3 97 2 9 101 <br>a=sendonly <br>a=rtpmap:0 PCMU/8000 <br>a=rtpmap:18 G729/8000 <br>a=rtpmap:3 GSM/8000 <br>a=rtpmap:97 iLBC/8000 <br>a=fmtp:97 mode=20 <br>a=rtpmap:2 G726-32/8000 <br>a=rtpmap:9 G722/16000 <br>a=ptime:20 <br>a=rtpmap:101 telephone-event/8000 <br>a=fmtp:101 0-11
<br><br>#<br>U +0.000646 192.168.10.1:5060 -> 192.168.10.40:5060<br>SIP/2.0 100 Giving a try <br>Via: SIP/2.0/UDP 192.168.10.40:5060;branch=z9hG4bK1a7bc3cdf6e22598 <br>From: "Ventas" <sip:112@192.168.10.1>;tag=0d380f28344df9f2 <br>To: <sip:*700@192.168.10.1>;tag=as5aea7a9e <br>Call-ID: f69f46f8461d45e0@192.168.10.40 <br>CSeq: 55344 INVITE <br>Server: OpenSER (1.3.4-notls (i386/linux)) <br>Content-Length: 0 <br> <br><br>#<br>U +0.000078 192.168.10.1:5060 -> 192.168.10.1:5070<br>INVITE sip:*700@192.168.10.1:5070 SIP/2.0 <br>Record-Route: <sip:192.168.10.1;lr=on;ftag=0d380f28344df9f2> <br>Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK537.cbf8a716.0 <br>Via: SIP/2.0/UDP 192.168.10.40:5060;branch=z9hG4bK1a7bc3cdf6e22598 <br>From: "Ventas" <sip:112@192.168.10.1>;tag=0d380f28344df9f2 <br>To: <sip:*700@192.168.10.1>;tag=as5aea7a9e <br>Contact: <sip:112@192.168.10.40:5060;transport=udp> <br>Supported: replaces, timer, path
<br>Referred-By: <sip:120@192.168.10.38:5060> <br>Proxy-Authorization: Digest username="112", realm="192.168.10.1", algorithm=MD5, uri="sip:*700@192.168.10.1:5070", nonce="495ac4ad695509e755aba895780497e8116e6353", response="40021b3138cbfefcd079505a55c6043f" <br>Call-ID: f69f46f8461d45e0@192.168.10.40 <br>CSeq: 55344 INVITE <br>User-Agent: Grandstream GXP2020 1.1.6.16 <br>Max-Forwards: 69 <br>Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE <br>Content-Type: application/sdp <br>Content-Length: 352 <br> <br>v=0 <br>o=112 8001 8002 IN IP4 192.168.10.40 <br>s=SIP Call <br>c=IN IP4 0.0.0.0 <br>t=0 0 <br>m=audio 5006 RTP/AVP 0 18 3 97 2 9 101 <br>a=sendonly <br>a=rtpmap:0 PCMU/8000 <br>a=rtpmap:18 G729/8000 <br>a=rtpmap:3 GSM/8000 <br>a=rtpmap:97 iLBC/8000 <br>a=fmtp:97 mode=20 <br>a=rtpmap:2 G726-32/8000 <br>a=rtpmap:9 G722/16000 <br>a=ptime:20 <br>a=rtpmap:101 telephone-event/8000 <br>a=fmtp:101 0-11
<br><br>#<br>U +0.000263 192.168.10.1:5070 -> 192.168.10.1:5060<br>SIP/2.0 100 Trying <br>Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK537.cbf8a716.0;received=192.168.10.1 <br>Via: SIP/2.0/UDP 192.168.10.40:5060;branch=z9hG4bK1a7bc3cdf6e22598 <br>Record-Route: <sip:192.168.10.1;lr=on;ftag=0d380f28344df9f2> <br>From: "Ventas" <sip:112@192.168.10.1>;tag=0d380f28344df9f2 <br>To: <sip:*700@192.168.10.1>;tag=as5aea7a9e <br>Call-ID: f69f46f8461d45e0@192.168.10.40 <br>CSeq: 55344 INVITE <br>User-Agent: Asterisk PBX <br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY <br>Supported: replaces <br>Contact: <sip:*700@192.168.10.1:5070> <br>Content-Length: 0 <br> <br><br>#<br>U +0.000104 192.168.10.1:5070 -> 192.168.10.1:5060<br>SIP/2.0 200 OK <br>Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK537.cbf8a716.0;received=192.168.10.1 <br>Via: SIP/2.0/UDP 192.168.10.40:5060;branch=z9hG4bK1a7bc3cdf6e22598 <br>Record-Route:
<sip:192.168.10.1;lr=on;ftag=0d380f28344df9f2> <br>From: "Ventas" <sip:112@192.168.10.1>;tag=0d380f28344df9f2 <br>To: <sip:*700@192.168.10.1>;tag=as5aea7a9e <br>Call-ID: f69f46f8461d45e0@192.168.10.40 <br>CSeq: 55344 INVITE <br>User-Agent: Asterisk PBX <br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY <br>Supported: replaces <br>Contact: <sip:*700@192.168.10.1:5070> <br>Content-Type: application/sdp <br>Content-Length: 285 <br> <br>v=0 <br>o=root 9758 9759 IN IP4 192.168.10.1 <br>s=session <br>c=IN IP4 192.168.10.1 <br>t=0 0 <br>m=audio 15948 RTP/AVP 0 18 101 <br>a=rtpmap:0 PCMU/8000 <br>a=rtpmap:18 G729/8000 <br>a=fmtp:18 annexb=no <br>a=rtpmap:101 telephone-event/8000 <br>a=fmtp:101 0-16 <br>a=silenceSupp:off - - - - <br>a=ptime:20 <br>a=recvonly <br><br>#<br>U +0.000104 192.168.10.1:5060 -> 192.168.10.40:5060<br>SIP/2.0 200 OK <br>Via: SIP/2.0/UDP 192.168.10.40:5060;branch=z9hG4bK1a7bc3cdf6e22598
<br>Record-Route: <sip:192.168.10.1;lr=on;ftag=0d380f28344df9f2> <br>From: "Ventas" <sip:112@192.168.10.1>;tag=0d380f28344df9f2 <br>To: <sip:*700@192.168.10.1>;tag=as5aea7a9e <br>Call-ID: f69f46f8461d45e0@192.168.10.40 <br>CSeq: 55344 INVITE <br>User-Agent: Asterisk PBX <br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY <br>Supported: replaces <br>Contact: <sip:*700@192.168.10.1:5070> <br>Content-Type: application/sdp <br>Content-Length: 285 <br> <br>v=0 <br>o=root 9758 9759 IN IP4 192.168.10.1 <br>s=session <br>c=IN IP4 192.168.10.1 <br>t=0 0 <br>m=audio 15948 RTP/AVP 0 18 101 <br>a=rtpmap:0 PCMU/8000 <br>a=rtpmap:18 G729/8000 <br>a=fmtp:18 annexb=no <br>a=rtpmap:101 telephone-event/8000 <br>a=fmtp:101 0-16 <br>a=silenceSupp:off - - - - <br>a=ptime:20 <br>a=recvonly <br><br>#<br>U +0.054311 192.168.10.40:5060 -> 192.168.10.1:5060<br>SIP/2.0 200 OK <br>Via: SIP/2.0/UDP
192.168.10.1;branch=z9hG4bK745a.fb2f15e.0 <br>Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK61abb915;rport=5070 <br>Record-Route: <sip:192.168.10.1;lr=on;ftag=as6fb8efad> <br>From: "Ventas" <sip:112@192.168.10.1>;tag=as6fb8efad <br>To: <sip:112@192.168.10.1>;tag=368cbb40ae863e2a <br>Call-ID: 3492c4a24c10596e6e3063c361c67eb9@192.168.10.1 <br>CSeq: 102 INVITE <br>User-Agent: Grandstream GXP2020 1.1.6.16 <br>Contact: <sip:112@192.168.10.40:5060;transport=udp> <br>Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE <br>Content-Type: application/sdp <br>Supported: replaces, timer <br>Content-Length: 234 <br> <br>v=0 <br>o=112 8000 8000 IN IP4 192.168.10.40 <br>s=SIP Call <br>c=IN IP4 192.168.10.40 <br>t=0 0 <br>m=audio 5004 RTP/AVP 0 101 <br>a=sendrecv <br>a=rtpmap:0 PCMU/8000 <br>a=ptime:20 <br>a=rtpmap:101 telephone-event/8000 <br>a=fmtp:101 0-11 <br>m=video 0 RTP/AVP 99 <br><br>#<br>U +0.000097
192.168.10.1:5060 -> 192.168.10.1:5070<br>SIP/2.0 200 OK <br>Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK61abb915;rport=5070 <br>Record-Route: <sip:192.168.10.1;lr=on;ftag=as6fb8efad> <br>From: "Ventas" <sip:112@192.168.10.1>;tag=as6fb8efad <br>To: <sip:112@192.168.10.1>;tag=368cbb40ae863e2a <br>Call-ID: 3492c4a24c10596e6e3063c361c67eb9@192.168.10.1 <br>CSeq: 102 INVITE <br>User-Agent: Grandstream GXP2020 1.1.6.16 <br>Contact: <sip:112@192.168.10.40:5060;transport=udp> <br>Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE <br>Content-Type: application/sdp <br>Supported: replaces, timer <br>Content-Length: 234</pre><br>I have openser and asterisk with realtime<br><br>any ideas ?<br><br>regards list <br><br>rickygm<br><br><br> </div></div><br>
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