Hello, Could anyone help to review the log and issue? Where I could post asterisk bugreport? I could help with testing if someone try to fix this error. Best regards, Vitali Fomine ----- Original Message ----- From: "Vitali Fomine" <support at officesip.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Sent: Friday, January 22, 2010 3:33 PM Subject: Re: [asterisk-users] OfficeSIP Softphone> Hello, > >> I would like to see this as well, from an Asterisk CLI log perspective >> with "sip debug" turned on. > > The .log file for login and invite is attached, I have use asterisk -vr > command. Is it correct? > >>> Yes, here is two INVITEs (I have missed first invite before), but the >>> server >>> respond 401 on first invite and softphone send ACK. Here is softphone >>> log. >> If Asterisk receives the ACK *after* the second INVITE I understand it. > > The softphone uses single tcp connection, so messages must arrive in same > order as them was sent. > > Best regards, > Vitali Fomine >--------------------------------------------------------------------------------> -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-------------- next part -------------- A non-text attachment was scrubbed... Name: login-invite.log Type: application/octet-stream Size: 20170 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20100128/380645ff/attachment.obj
You are running an Asterisk version for SIP TCP ? your SIP UA seems talking SIP over TCP Via: SIP/2.0/TCP 192.168.1.15:56298 Max-Forwards: 70 From: <sip:56 at trixbox1.local>;tag=2baacde98c;epid=aa3c1b27a7 To: <sip:MRASLoc.trixbox1.local at trixbox1.local> Call-ID: 28a90e7402da49159f343be9bc82b4d0 CSeq: 1 SERVICE Contact: <sip:56 at trixbox1.local :56298;maddr=192.168.1.15;transport=tcp>;proxy=replace;+sip.instance="<urn:uuid:1ADF8582-5BD5-531A-BC2A-C76FECED0C4E>" User-Agent: UCCAPI/2.0.6362.67 Authorization: Digest username="56", realm="asterisk", algorithm=MD5, uri="sip:MRASLoc.trixbox1.local at trixbox1.local", nonce="36662fdf", response="d6f90f263010891a42b3f7d46113796a" Content-Type: application/msrtc-media-relay-auth+xml Content-Length: 395 -- -- Adri? Vidal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100128/f8d09a02/attachment.htm
Hello,
Yes, unfortunately, the sip client lib does not support udp.
Best regards,
Vitali Fomine
----- Original Message -----
From: Adri? Vidal
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Thursday, January 28, 2010 3:49 PM
Subject: Re: [asterisk-users] Fw: OfficeSIP Softphone
You are running an Asterisk version for SIP TCP ?
your SIP UA seems talking SIP over TCP
Via: SIP/2.0/TCP 192.168.1.15:56298
Max-Forwards: 70
From: <sip:56 at trixbox1.local>;tag=2baacde98c;epid=aa3c1b27a7
To: <sip:MRASLoc.trixbox1.local at trixbox1.local>
Call-ID: 28a90e7402da49159f343be9bc82b4d0
CSeq: 1 SERVICE
Contact: <sip:56 at
trixbox1.local:56298;maddr=192.168.1.15;transport=tcp>;proxy=replace;+sip.instance="<urn:uuid:1ADF8582-5BD5-531A-BC2A-C76FECED0C4E>"
User-Agent: UCCAPI/2.0.6362.67
Authorization: Digest username="56", realm="asterisk",
algorithm=MD5, uri="sip:MRASLoc.trixbox1.local at trixbox1.local",
nonce="36662fdf",
response="d6f90f263010891a42b3f7d46113796a"
Content-Type: application/msrtc-media-relay-auth+xml
Content-Length: 395
--
--
Adri? Vidal
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On Thu, Jan 28, 2010 at 1:56 PM, Vitali Fomine <support at officesip.com>wrote:> Hello, > > Yes, unfortunately, the sip client lib does not support udp. > > Best regards, > Vitali Fomine > > >Then check you are using an Asterisk patched for TCP. -- -- Adri? Vidal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100128/67d515da/attachment.htm
Hello,
Thank you for your help. I have enable tcp by using tcpenable and tcpbindaddr.
The client can not connect w/o these settings. I am trying asterisk 1.6.0.10 (in
trixbox), need I install something else?
Best regards,
Vitali Fomine
----- Original Message -----
From: Adri? Vidal
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Thursday, January 28, 2010 4:20 PM
Subject: Re: [asterisk-users] Fw: OfficeSIP Softphone
On Thu, Jan 28, 2010 at 1:56 PM, Vitali Fomine <support at
officesip.com> wrote:
Hello,
Yes, unfortunately, the sip client lib does not support udp.
Best regards,
Vitali Fomine
Then check you are using an Asterisk patched for TCP.
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Adri? Vidal
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