[Kamailio-Users] trying to use call park for similar a music on hold

Ricky Gutierrez xserverlinux at yahoo.com
Wed Dec 31 02:05:31 CET 2008


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 at 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 at 192.168.10.1>;tag=0d380f28344df9f2 
To: <sip:*700 at 192.168.10.1>;tag=as5aea7a9e 
Contact: <sip:112 at 192.168.10.40:5060;transport=udp> 
Supported: replaces, timer, path 
Referred-By: <sip:120 at 192.168.10.38:5060> 
Proxy-Authorization: Digest username="112", realm="192.168.10.1", algorithm=MD5, uri="sip:*700 at 192.168.10.1:5070", nonce="495ac4ad695509e755aba895780497e8116e6353", response="40021b3138cbfefcd079505a55c6043f" 
Call-ID: f69f46f8461d45e0 at 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 at 192.168.10.1>;tag=0d380f28344df9f2 
To: <sip:*700 at 192.168.10.1>;tag=as5aea7a9e 
Call-ID: f69f46f8461d45e0 at 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 at 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 at 192.168.10.1>;tag=0d380f28344df9f2 
To: <sip:*700 at 192.168.10.1>;tag=as5aea7a9e 
Contact: <sip:112 at 192.168.10.40:5060;transport=udp> 
Supported: replaces, timer, path 
Referred-By: <sip:120 at 192.168.10.38:5060> 
Proxy-Authorization: Digest username="112", realm="192.168.10.1", algorithm=MD5, uri="sip:*700 at 192.168.10.1:5070", nonce="495ac4ad695509e755aba895780497e8116e6353", response="40021b3138cbfefcd079505a55c6043f" 
Call-ID: f69f46f8461d45e0 at 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 at 192.168.10.1>;tag=0d380f28344df9f2 
To: <sip:*700 at 192.168.10.1>;tag=as5aea7a9e 
Call-ID: f69f46f8461d45e0 at 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 at 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 at 192.168.10.1>;tag=0d380f28344df9f2 
To: <sip:*700 at 192.168.10.1>;tag=as5aea7a9e 
Call-ID: f69f46f8461d45e0 at 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 at 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 at 192.168.10.1>;tag=0d380f28344df9f2 
To: <sip:*700 at 192.168.10.1>;tag=as5aea7a9e 
Call-ID: f69f46f8461d45e0 at 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 at 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 at 192.168.10.1>;tag=as6fb8efad 
To: <sip:112 at 192.168.10.1>;tag=368cbb40ae863e2a 
Call-ID: 3492c4a24c10596e6e3063c361c67eb9 at 192.168.10.1 
CSeq: 102 INVITE 
User-Agent: Grandstream GXP2020 1.1.6.16 
Contact: <sip:112 at 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 at 192.168.10.1>;tag=as6fb8efad 
To: <sip:112 at 192.168.10.1>;tag=368cbb40ae863e2a 
Call-ID: 3492c4a24c10596e6e3063c361c67eb9 at 192.168.10.1 
CSeq: 102 INVITE 
User-Agent: Grandstream GXP2020 1.1.6.16 
Contact: <sip:112 at 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


      
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