Skipped content of type multipart/alternative-------------- next part -------------- Reliably Transmitting (no NAT) to 111.111.111.50:16666: INVITE sip:111.111.111.50:16666 SIP/2.0 Via: SIP/2.0/UDP 111.111.111.8:5060;branch=z9hG4bK360502ae;rport From: "asterisk" <sip:asterisk@111.111.111.8>;tag=as348de10b To: <sip:111.111.111.50:16666> Contact: <sip:asterisk@111.111.111.8> Call-ID: 104d16a9260f074f7d3dba3235b1870a@111.111.111.8 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 13 Jun 2006 07:25:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 242 v=0 o=root 10295 10295 IN IP4 111.111.111.8 s=session c=IN IP4 111.111.111.8 t=0 0 m=audio 12742 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- [Kasterisk*CLI> Retransmitting #1 (no NAT) to 111.111.111.50:16666: INVITE sip:111.111.111.50:16666 SIP/2.0 Via: SIP/2.0/UDP 111.111.111.8:5060;branch=z9hG4bK360502ae;rport From: "asterisk" <sip:asterisk@111.111.111.8>;tag=as348de10b To: <sip:111.111.111.50:16666> Contact: <sip:asterisk@111.111.111.8> Call-ID: 104d16a9260f074f7d3dba3235b1870a@111.111.111.8 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 13 Jun 2006 07:25:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 242 v=0 o=root 10295 10295 IN IP4 111.111.111.8 s=session c=IN IP4 111.111.111.8 t=0 0 m=audio 12742 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #2 (no NAT) to 111.111.111.50:16666: INVITE sip:111.111.111.50:16666 SIP/2.0 Via: SIP/2.0/UDP 111.111.111.8:5060;branch=z9hG4bK360502ae;rport From: "asterisk" <sip:asterisk@111.111.111.8>;tag=as348de10b To: <sip:111.111.111.50:16666> Contact: <sip:asterisk@111.111.111.8> Call-ID: 104d16a9260f074f7d3dba3235b1870a@111.111.111.8 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 13 Jun 2006 07:25:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 242 v=0 o=root 10295 10295 IN IP4 111.111.111.8 s=session c=IN IP4 111.111.111.8 t=0 0 m=audio 12742 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- [Kasterisk*CLI> Retransmitting #3 (no NAT) to 111.111.111.50:16666: INVITE sip:111.111.111.50:16666 SIP/2.0 Via: SIP/2.0/UDP 111.111.111.8:5060;branch=z9hG4bK360502ae;rport From: "asterisk" <sip:asterisk@111.111.111.8>;tag=as348de10b To: <sip:111.111.111.50:16666> Contact: <sip:asterisk@111.111.111.8> Call-ID: 104d16a9260f074f7d3dba3235b1870a@111.111.111.8 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 13 Jun 2006 07:25:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 242 v=0 o=root 10295 10295 IN IP4 111.111.111.8 s=session c=IN IP4 111.111.111.8 t=0 0 m=audio 12742 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- [Kasterisk*CLI> Retransmitting #4 (no NAT) to 111.111.111.50:16666: INVITE sip:111.111.111.50:16666 SIP/2.0 Via: SIP/2.0/UDP 111.111.111.8:5060;branch=z9hG4bK360502ae;rport From: "asterisk" <sip:asterisk@111.111.111.8>;tag=as348de10b To: <sip:111.111.111.50:16666> Contact: <sip:asterisk@111.111.111.8> Call-ID: 104d16a9260f074f7d3dba3235b1870a@111.111.111.8 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 13 Jun 2006 07:25:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 242 v=0 o=root 10295 10295 IN IP4 111.111.111.8 s=session c=IN IP4 111.111.111.8 t=0 0 m=audio 12742 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- [Kasterisk*CLI> Retransmitting #5 (no NAT) to 111.111.111.50:16666: INVITE sip:111.111.111.50:16666 SIP/2.0 Via: SIP/2.0/UDP 111.111.111.8:5060;branch=z9hG4bK360502ae;rport From: "asterisk" <sip:asterisk@111.111.111.8>;tag=as348de10b To: <sip:111.111.111.50:16666> Contact: <sip:asterisk@111.111.111.8> Call-ID: 104d16a9260f074f7d3dba3235b1870a@111.111.111.8 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 13 Jun 2006 07:25:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 242 v=0 o=root 10295 10295 IN IP4 111.111.111.8 s=session c=IN IP4 111.111.111.8 t=0 0 m=audio 12742 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- [Kasterisk*CLI> Retransmitting #6 (no NAT) to 111.111.111.50:16666: INVITE sip:111.111.111.50:16666 SIP/2.0 Via: SIP/2.0/UDP 111.111.111.8:5060;branch=z9hG4bK360502ae;rport From: "asterisk" <sip:asterisk@111.111.111.8>;tag=as348de10b To: <sip:111.111.111.50:16666> Contact: <sip:asterisk@111.111.111.8> Call-ID: 104d16a9260f074f7d3dba3235b1870a@111.111.111.8 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 13 Jun 2006 07:25:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 242 v=0 o=root 10295 10295 IN IP4 111.111.111.8 s=session c=IN IP4 111.111.111.8 t=0 0 m=audio 12742 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- [Kasterisk*CLI> Jun 13 09:25:36 WARNING[10305]: chan_sip.c:1217 retrans_pkt: Maximum retries exceeded on transmission 104d16a9260f074f7d3dba3235b1870a@111.111.111.8 for seqno 102 (Critical Request) Jun 13 09:25:36 WARNING[10305]: chan_sip.c:1234 retrans_pkt: Hanging up call 104d16a9260f074f7d3dba3235b1870a@111.111.111.8 - no reply to our critical packet. [Kasterisk*CLI> <-- SIP read from 111.111.111.50:1380: SIP/2.0 100 Trying Via: SIP/2.0/UDP 111.111.111.8:5060;branch=z9hG4bK360502ae;rport From: "asterisk" <sip:asterisk@111.111.111.8>;tag=as348de10b To: <sip:111.111.111.50:16666>;tag=60ddafdb3c924f2f87bcd1fe186f7e7f Call-ID: 104d16a9260f074f7d3dba3235b1870a@111.111.111.8 CSeq: 102 INVITE User-Agent: RTC/1.2 Content-Length: 0
Ohad.Levy@infineon.com
2006-Jun-14 01:13 UTC
[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@lists.digium.com [mailto:asterisk-users-bounces@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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060614/d2172558/attachment.htm
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@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@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of
Ohad.Levy@infineon.com
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 :-)
Cheers,
Ohad
_____
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@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
-------------- next part --------------
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URL:
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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@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of
Ohad.Levy@infineon.com
Sent: Wednesday, June 14, 2006 12:50 PM
To: asterisk-users@lists.digium.com
Cc: hjo@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@lists.digium.com
[mailto:asterisk-users-bounces@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@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@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of
Ohad.Levy@infineon.com
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 :-)
Cheers,
Ohad
_____
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@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
-------------- next part --------------
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