RE: [Asterisk-Users] SIP, Microsoft RTC, and Originate problem
Hi, What is your setup? By MS RTC do you mean Office Communicator? If you are using MS OC, do you use SER in between (to convert SIP UDP2TCP)? Please share some more details J Cheers, Ohad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Sent: Wednesday, June 14, 2006 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SIP, Microsoft RTC, and Originate problem It seems that Microsoft RTC has some problems with originated calls from Asterisk. If I execute Manager API originate application, with SIP channel as parameter, the Microsoft RTC softphone will start to ring after a couple of seconds delay, but nothing more happens after when I answer there is no second call to an extension. When I looked through the sip debug, I noticed that Microsoft RTC fails to properly respond to INVITE messages (I have attached the sip debug). Asterisk has to retransmit INVITE message for 6 times and even then the RTC still doesn't respond in a proper time. However, if I do direct call to that problematic Microsoft RTC based softphone, everything works fine, eventhough very same INVITE messages are being transmited to it from Asterisk. Does anyone have any ideas for a workaround? Regards, Alex ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP, Microsoft RTC, and Originate problem
Nope, it's just the Microsoft RTC Core 1.3 library ... more or less a single DLL J. And I'm almost sure there is no SER in between should there be one? It's pretty much a straightforward thing I have a few SIP clients defined in my sip.conf, like this: [general] context=default allowguest=yes realm=timd.si bindport=5060 bindaddr=0.0.0.0 srvlookup=yes domain=timd.si,from-sip domain=111.111.111.8,from-sip videosupport=yes disallow=all allow=alaw allow=ulaw musicclass=default rtptimeout=100 rtpholdtimeout=100 tos=0x18 canreinvite=yes [SIPClient001] username= SIPClient001 secret= mysecret type=friend host=dynamic context=from-sip disallow=all allow=alaw allow=ulaw qualify=yes [SIPClient002] username= SIPClient002 secret= mysecret type=friend host=dynamic context=from-sip disallow=all allow=alaw allow=ulaw qualify=yes And there is an MS RTC based Softphone, that I made, on the other side that registers to Asterisk, using this profile XML string: provision key=5B29C449-29EE-4fd8-9E3F-04AED077690E name=Asterisk user account=SIPClient001 uri=sip:[EMAIL PROTECTED] / sipsrv addr=111.111.111.8 protocol=udp auth=digest role=registrar session party=first type=pc2ph / /sipsrv /provision Now, doing an originate to CHANNEL=SIP/SIPClient002, and some extension, will randomly fail, for example (see OriginateFailure reponse as well): action: Originate actionid: 123 exten: 03020846051635424 channel: SIP/SIPClient002 timeout: 3 priority: 1 context: asttel async: true Event: OriginateFailure Privilege: call,all ActionID: 123 Channel: SIP/ SIPClient002 Context: asttel Exten: 03020846051635424 Reason: 1 Uniqueid: null From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, June 14, 2006 10:14 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] SIP, Microsoft RTC, and Originate problem Hi, What is your setup? By MS RTC do you mean Office Communicator? If you are using MS OC, do you use SER in between (to convert SIP UDP2TCP)? Please share some more details J Cheers, Ohad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Sent: Wednesday, June 14, 2006 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SIP, Microsoft RTC, and Originate problem It seems that Microsoft RTC has some problems with originated calls from Asterisk. If I execute Manager API originate application, with SIP channel as parameter, the Microsoft RTC softphone will start to ring after a couple of seconds delay, but nothing more happens after when I answer there is no second call to an extension. When I looked through the sip debug, I noticed that Microsoft RTC fails to properly respond to INVITE messages (I have attached the sip debug). Asterisk has to retransmit INVITE message for 6 times and even then the RTC still doesn't respond in a proper time. However, if I do direct call to that problematic Microsoft RTC based softphone, everything works fine, eventhough very same INVITE messages are being transmited to it from Asterisk. Does anyone have any ideas for a workaround? Regards, Alex ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP, Microsoft RTC, and Originate problem
Hmm.. Interesting, I didnt try to implement it this way... but, if its the same libraries used for Office communicator, than it supports only SIP over TCP or TLS, since asterisk doesnt support any of those its impossible to connect them directly... If udp works, maybe the registration part is problematic, try configuring asterisk with autocreatepeer (just for testing) to see if you can dial out without being registered. Ohad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Sent: Wednesday, June 14, 2006 11:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] SIP, Microsoft RTC, and Originate problem Nope, it's just the Microsoft RTC Core 1.3 library ... more or less a single DLL J. And I'm almost sure there is no SER in between should there be one? It's pretty much a straightforward thing I have a few SIP clients defined in my sip.conf, like this: [general] context=default allowguest=yes realm=timd.si bindport=5060 bindaddr=0.0.0.0 srvlookup=yes domain=timd.si,from-sip domain=111.111.111.8,from-sip videosupport=yes disallow=all allow=alaw allow=ulaw musicclass=default rtptimeout=100 rtpholdtimeout=100 tos=0x18 canreinvite=yes [SIPClient001] username= SIPClient001 secret= mysecret type=friend host=dynamic context=from-sip disallow=all allow=alaw allow=ulaw qualify=yes [SIPClient002] username= SIPClient002 secret= mysecret type=friend host=dynamic context=from-sip disallow=all allow=alaw allow=ulaw qualify=yes And there is an MS RTC based Softphone, that I made, on the other side that registers to Asterisk, using this profile XML string: provision key=5B29C449-29EE-4fd8-9E3F-04AED077690E name=Asterisk user account=SIPClient001 uri=sip:[EMAIL PROTECTED] / sipsrv addr=111.111.111.8 protocol=udp auth=digest role=registrar session party=first type=pc2ph / /sipsrv /provision Now, doing an originate to CHANNEL=SIP/SIPClient002, and some extension, will randomly fail, for example (see OriginateFailure reponse as well): action: Originate actionid: 123 exten: 03020846051635424 channel: SIP/SIPClient002 timeout: 3 priority: 1 context: asttel async: true Event: OriginateFailure Privilege: call,all ActionID: 123 Channel: SIP/ SIPClient002 Context: asttel Exten: 03020846051635424 Reason: 1 Uniqueid: null From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, June 14, 2006 10:14 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] SIP, Microsoft RTC, and Originate problem Hi, What is your setup? By MS RTC do you mean Office Communicator? If you are using MS OC, do you use SER in between (to convert SIP UDP2TCP)? Please share some more details J Cheers, Ohad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Sent: Wednesday, June 14, 2006 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SIP, Microsoft RTC, and Originate problem It seems that Microsoft RTC has some problems with originated calls from Asterisk. If I execute Manager API originate application, with SIP channel as parameter, the Microsoft RTC softphone will start to ring after a couple of seconds delay, but nothing more happens after when I answer there is no second call to an extension. When I looked through the sip debug, I noticed that Microsoft RTC fails to properly respond to INVITE messages (I have attached the sip debug). Asterisk has to retransmit INVITE message for 6 times and even then the RTC still doesn't respond in a proper time. However, if I do direct call to that problematic Microsoft RTC based softphone, everything works fine, eventhough very same INVITE messages are being transmited to it from Asterisk. Does anyone have any ideas for a workaround? Regards, Alex ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP, Microsoft RTC, and Originate problem
I tried your suggestion and found out that someone/something I don't know whether that is an MS RTC or Asterisk is having problems if the same Windows application is using Manager and SIP at the same time. At least for now, it has always worked, if I tried to initiate Originate command from one application, and had MS RTC in another. As soon as I put these two things in the same application, it stops working...weird. Has anyone experienced anything like that before? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, June 14, 2006 12:50 PM To: asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] SIP, Microsoft RTC, and Originate problem Hmm.. Interesting, I didnt try to implement it this way... but, if its the same libraries used for Office communicator, than it supports only SIP over TCP or TLS, since asterisk doesnt support any of those its impossible to connect them directly... If udp works, maybe the registration part is problematic, try configuring asterisk with autocreatepeer (just for testing) to see if you can dial out without being registered. Ohad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Sent: Wednesday, June 14, 2006 11:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] SIP, Microsoft RTC, and Originate problem Nope, it's just the Microsoft RTC Core 1.3 library ... more or less a single DLL J. And I'm almost sure there is no SER in between should there be one? It's pretty much a straightforward thing I have a few SIP clients defined in my sip.conf, like this: [general] context=default allowguest=yes realm=timd.si bindport=5060 bindaddr=0.0.0.0 srvlookup=yes domain=timd.si,from-sip domain=111.111.111.8,from-sip videosupport=yes disallow=all allow=alaw allow=ulaw musicclass=default rtptimeout=100 rtpholdtimeout=100 tos=0x18 canreinvite=yes [SIPClient001] username= SIPClient001 secret= mysecret type=friend host=dynamic context=from-sip disallow=all allow=alaw allow=ulaw qualify=yes [SIPClient002] username= SIPClient002 secret= mysecret type=friend host=dynamic context=from-sip disallow=all allow=alaw allow=ulaw qualify=yes And there is an MS RTC based Softphone, that I made, on the other side that registers to Asterisk, using this profile XML string: provision key=5B29C449-29EE-4fd8-9E3F-04AED077690E name=Asterisk user account=SIPClient001 uri=sip:[EMAIL PROTECTED] / sipsrv addr=111.111.111.8 protocol=udp auth=digest role=registrar session party=first type=pc2ph / /sipsrv /provision Now, doing an originate to CHANNEL=SIP/SIPClient002, and some extension, will randomly fail, for example (see OriginateFailure reponse as well): action: Originate actionid: 123 exten: 03020846051635424 channel: SIP/SIPClient002 timeout: 3 priority: 1 context: asttel async: true Event: OriginateFailure Privilege: call,all ActionID: 123 Channel: SIP/ SIPClient002 Context: asttel Exten: 03020846051635424 Reason: 1 Uniqueid: null From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, June 14, 2006 10:14 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] SIP, Microsoft RTC, and Originate problem Hi, What is your setup? By MS RTC do you mean Office Communicator? If you are using MS OC, do you use SER in between (to convert SIP UDP2TCP)? Please share some more details J Cheers, Ohad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Sent: Wednesday, June 14, 2006 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SIP, Microsoft RTC, and Originate problem It seems that Microsoft RTC has some problems with originated calls from Asterisk. If I execute Manager API originate application, with SIP channel as parameter, the Microsoft RTC softphone will start to ring after a couple of seconds delay, but nothing more happens after when I answer there is no second call to an extension. When I looked through the sip debug, I noticed that Microsoft RTC fails to properly respond to INVITE messages (I have attached the sip debug). Asterisk has to retransmit INVITE message for 6 times and even then the RTC still doesn't respond in a proper time. However, if I do direct call to that problematic Microsoft RTC based softphone, everything works fine, eventhough very same INVITE messages are being transmited to it from Asterisk. Does anyone have any ideas for a workaround? Regards, Alex ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
