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