[Asterisk-Users] SIP, Microsoft RTC, and Originate problem

[Asterisk]

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: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
Ohad.Levy at infineon.com
Sent: Wednesday, June 14, 2006 12:50 PM
To: asterisk-users at lists.digium.com
Cc: hjo at infineon.com
Subject: RE: [Asterisk-Users] SIP, Microsoft RTC, and Originate problem

 

Hmm..... Interesting, I didn't try to implement it this way... but, if
it's the same libraries used for Office communicator, than it supports
only SIP over TCP or TLS, since asterisk doesn't 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: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] 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 :-). 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:SIPClient001 at 111.111.111.8" />

    <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: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
Ohad.Levy at infineon.com
Sent: Wednesday, June 14, 2006 10:14 AM
To: asterisk-users at 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 :-)

 

Cheers,

Ohad

 

  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] 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

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