|
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:
000003020846051635424 channel: SIP/SIPClient002 timeout: 30000 priority: 1 context: asttel async: true Event: OriginateFailure Privilege: call,all ActionID: 123 Channel: SIP/ SIPClient002 Context: asttel Exten:
000003020846051635424 Reason: 1 Uniqueid: <null> From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] 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 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 |
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