onsip.org "getting started"
note: do not try to use mediaproxy before you solved REGISTER handling
klaus
unplug wrote:
As title, any where I can find such howto?
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
Thanks! I have tried nat-mediaproxy.5.0.cfg and gw-pstn.5.0.cfg with modification. However, none of them can work completely.
For nat-mediaproxy, I have mediaproxy1.4.2 installed and then follow the document to adjust the parameter of the configuration file. Then there are 2 IP phones (A & B) connect (NAT) to the SIP server. The result is A can contact B but B failed to contact A.
For gw-pstn, the case is more worst. A & B is the internal IP phone. C is PSTN phone. None of them can make a call to any other.
Just want to know anyone can make both configuration work. Any tricky action that needed to take care. Please advise.
On 12/14/05, Klaus Darilion klaus.mailinglists@pernau.at wrote:
onsip.org "getting started"
note: do not try to use mediaproxy before you solved REGISTER handling
klaus
unplug wrote:
As title, any where I can find such howto?
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
I just can tell you that it is able to handle all this situation. But I can't help you if you just say "A cant call B". You find to dig into the problem. Learn how SIP works. Learn how to use ethereal and ngrep to watch the SIP call flow. Use ethereal to watch the RTP streams. Use xlog to watch how SIP messages yre routed by your SIP proxy.
Then post the excat problem and also post the correspondig SIP call flow (e.g. use ngrep -W byline)
regards klaus
unplug wrote:
Thanks! I have tried nat-mediaproxy.5.0.cfg and gw-pstn.5.0.cfg with modification. However, none of them can work completely.
For nat-mediaproxy, I have mediaproxy1.4.2 installed and then follow the document to adjust the parameter of the configuration file. Then there are 2 IP phones (A & B) connect (NAT) to the SIP server. The result is A can contact B but B failed to contact A.
For gw-pstn, the case is more worst. A & B is the internal IP phone. C is PSTN phone. None of them can make a call to any other.
Just want to know anyone can make both configuration work. Any tricky action that needed to take care. Please advise.
On 12/14/05, Klaus Darilion klaus.mailinglists@pernau.at wrote:
onsip.org "getting started"
note: do not try to use mediaproxy before you solved REGISTER handling
klaus
unplug wrote:
As title, any where I can find such howto?
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
Thanks! Unless I know the scripts are working. Well, I am using the sample gw-pstn.5.0.cfg. I can make call from IP phone to PSTN phone, the callee rings but there is no ring tone in caller. Below is the log with error 500. Is it the configuration problem of mediaproxy?
A message from G/W to openser U 203.193.46.2:5060 -> 203.193.46.6:5060 SIP/2.0 500 FQDN in SDP Media cannot be resolved..Via: SIP/2.0/UDP 203.193. 46.6;branch=z9hG4bK92a3.b70bebf.0,SIP/2.0/UDP 10.0.0.54:5060;rport=15060; received=210.184.43.1;branch=z9hG4bKm2iYDC7IcO4digOk..From: "8498" <si p:8498@o01.ol.com>;tag=VzXcnm2ihb05wKiC..To: "93634" <sip :93634@o01.ol.com>;tag=28B191AC-B8F..Date: Fri, 16 Dec 2005 0 7:30:46 GMT..Call-ID: wKLVEZGcY3rHMY8K@10.0.0.54..Server: Cisco-SIPGateway/ IOS-12.x..CSeq: 1 INVITE..Allow-Events: telephone-event..Content-Length: 0.
...
There is a warning from the log: Warning: sl_send_reply: I won't send a reply for ACK!!
When I make a call from PSTN to IPPhone, the following error shown. Is it related to the same problem as by previous post?
message from openser to G/W U 203.193.46.6:5060 -> 203.193.46.2:5060 SIP/2.0 483 Too Many Hops..Via: SIP/2.0/UDP 203.193.46.2:5060..From: <si p:3634@203.193.46.2>;tag=2945E878-2498..To: <sip:8498@203.193.46. 6>;tag=b5ab8e75af536221ca172f8fc39505eb.a169..Call-ID: 52A60849-6D5311DA- 8286A518-578F6E21@203.193.46.2..CSeq: 101 INVITE..Server: OpenSer (1.0.0 (i386/linux))..Content-Length: 0..Warning: 392 203.193.46.6:5060 "Noisy f eedback tells: pid=13742 req_src_ip=203.193.46.6 req_src_port=5060 in_ur i=sip:8498@203.193.46.6:5060 out_uri=sip:8498@203.193.46.6:5060 via_cnt==16".... On 12/16/05, unplug maillisting@gmail.com wrote:
Thanks! Unless I know the scripts are working. Well, I am using the sample gw-pstn.5.0.cfg. I can make call from IP phone to PSTN phone, the callee rings but there is no ring tone in caller. Below is the log with error 500. Is it the configuration problem of mediaproxy?
A message from G/W to openser U 203.193.46.2:5060 -> 203.193.46.6:5060 SIP/2.0 500 FQDN in SDP Media cannot be resolved..Via: SIP/2.0/UDP 203.193. 46.6;branch=z9hG4bK92a3.b70bebf.0,SIP/2.0/UDP 10.0.0.54:5060;rport=15060; received=210.184.43.1;branch=z9hG4bKm2iYDC7IcO4digOk..From: "8498" <si p:8498@o01.ol.com>;tag=VzXcnm2ihb05wKiC..To: "93634" <sip :93634@o01.ol.com>;tag=28B191AC-B8F..Date: Fri, 16 Dec 2005 0 7:30:46 GMT..Call-ID: wKLVEZGcY3rHMY8K@10.0.0.54..Server: Cisco-SIPGateway/ IOS-12.x..CSeq: 1 INVITE..Allow-Events: telephone-event..Content-Length: 0.
...
There is a warning from the log: Warning: sl_send_reply: I won't send a reply for ACK!!
You just should read the error message:
SIP/2.0 500 FQDN in SDP Media cannot be resolved
Take a look at the SDP of the corresponding SIP request (before the SIP proxy and after the SIP proxy)
klaus
unplug wrote:
Thanks! Unless I know the scripts are working. Well, I am using the sample gw-pstn.5.0.cfg. I can make call from IP phone to PSTN phone, the callee rings but there is no ring tone in caller. Below is the log with error 500. Is it the configuration problem of mediaproxy?
A message from G/W to openser U 203.193.46.2:5060 -> 203.193.46.6:5060 SIP/2.0 500 FQDN in SDP Media cannot be resolved..Via: SIP/2.0/UDP 203.193. 46.6;branch=z9hG4bK92a3.b70bebf.0,SIP/2.0/UDP 10.0.0.54:5060;rport=15060; received=210.184.43.1;branch=z9hG4bKm2iYDC7IcO4digOk..From: "8498" <si p:8498@o01.ol.com>;tag=VzXcnm2ihb05wKiC..To: "93634" <sip :93634@o01.ol.com>;tag=28B191AC-B8F..Date: Fri, 16 Dec 2005 0 7:30:46 GMT..Call-ID: wKLVEZGcY3rHMY8K@10.0.0.54..Server: Cisco-SIPGateway/ IOS-12.x..CSeq: 1 INVITE..Allow-Events: telephone-event..Content-Length: 0.
...
There is a warning from the log: Warning: sl_send_reply: I won't send a reply for ACK!!
-
Klaus Darilion -
unplug