Also, did you happen to follow the "how-to" to configure the voip.ms portal? It sounds like you left the NAT feature on there.
http://wiki.voip.ms/article/PBXs#SIPfoundry On Tue, Oct 18, 2011 at 8:03 AM, Tony Graziano <[email protected] > wrote: > voip.ms uses 10000-20000 for rtp by default, so I don't understand how > they "clashed". Please explain what ports you used for media relay (which is > ONLY for remote users), and outbound ITSP requests (not inbound).. > > I see your source port is a low port. So explain your firewall > configuration please. > > > On Tue, Oct 18, 2011 at 7:58 AM, Philippe Bechamp < > [email protected]> wrote: > >> I did change the ports because the were clashing with the Media Relay and >> preventing it from starting up a while ago. This dd not break anything when >> I changed it.. >> >> I m connected to the outside world via voip.ms sip trunk. >> >> 101 is my phone, 100 is sipx - the call is initiated by 101, the IP-450. >> >> Sorry for the incomplete answer, I think I stayed up on this way too >> late... >> >> Thanks ! >> >> Sorry about the PSTN question, I worked on this till 3 a.m. >> >> On 2011-10-18, at 7:49 AM, Tony Graziano wrote: >> >> If the invite is FROM your sipx server, did you change the RTP ports? The >> default would be 30000-31000, but the diagram below shows 20000 range. So, >> how about explaining what is controlling the PSTN connectivity? Is it a >> POTS/PRI gateway, siptrunk, etc.? >> >> On Tue, Oct 18, 2011 at 7:47 AM, Philippe Bechamp < >> [email protected]> wrote: >> >>> Hi Tony, >>> >>> Thanks for the reply ! >>> >>> I didn't get a sip-trace but I will a bit later when I get in the office. >>> >>> In the mean time, I can offer Wireshark's VOIP Flow diagram. >>> >>> Thanks ! >>> >>> Philippe. >>> >>> |Time | 10.70.225.101 | >>> | | | 10.70.103.100 | >>> |15.526 | INVITE SDP (g722 g711U g711A g729 >>> telephone-ev...RTPType-101) |SIP From: "Philippe Bechamp" < >>> sip:[email protected] To:< >>> sip:[email protected];user=phone >>> | |(5060) ------------------> (5060) | >>> |15.527 | 100 Trying| |SIP Status >>> | |(5060) <------------------ (5060) | >>> |15.528 | 407 Proxy Authentication Required |SIP >>> Status >>> | |(5060) <------------------ (5060) | >>> |15.554 | ACK | |SIP Request >>> | |(5060) ------------------> (5060) | >>> |15.557 | INVITE SDP (g722 g711U g711A g729 >>> telephone-ev...RTPType-101) |SIP From: "Philippe Bechamp" < >>> sip:[email protected] To:< >>> sip:[email protected];user=phone >>> | |(5060) ------------------> (5060) | >>> |15.558 | 100 Trying| |SIP Status >>> | |(5060) <------------------ (5060) | >>> |24.601 | 183 Session progress SDP (g711U >>> telephone-even...PType-101) |SIP Status >>> | |(5060) <------------------ (5060) | >>> |24.712 | RTP (g711U) |RTP Num packets:17 >>> Duration:0.320s SSRC:0x12FE0FDC >>> | |(2230) ------------------> (20006) | >>> |24.712 | RTP (g711U) |RTP Num packets:2374 >>> Duration:47.460s SSRC:0x12FE0FDC >>> | |(2230) <------------------ (20002) | >>> |24.720 | RTP (g711U) |RTP Num packets:32 >>> Duration:0.299s SSRC:0x28A943DA >>> | |(2230) <------------------ (20002) | >>> |25.041 | RTP (g711U) |RTP Num packets:4722 >>> Duration:47.198s SSRC:0x489F3B37 >>> | |(2230) <------------------ (20002) | >>> |25.051 | 200 OK SDP (g711U telephone-eventRTPType-101) >>> |SIP Status >>> | |(5060) <------------------ (5060) | >>> |25.053 | RTP (g711U) |RTP Num packets:2357 >>> Duration:47.120s SSRC:0x12FE0FDC >>> | |(2230) ------------------> (20006) | >>> |25.070 | ACK | |SIP Request >>> | |(5060) ------------------> (5060) | >>> |72.229 | BYE | |SIP Request >>> | |(5060) ------------------> (5060) | >>> |72.259 | 200 OK | |SIP Status >>> | |(5060) <------------------ (5060) | >>> >>> >>> >>> On 2011-10-18, at 6:56 AM, Tony Graziano wrote: >>> >>> what is your pstn connectivity? did you grab a siptrace? if so, I. an >>> look at that if you like. >>> On Oct 18, 2011 2:27 AM, "Philippe Bechamp" < >>> [email protected]> wrote: >>> >>>> Hi there, >>>> >>>> Wonder if anyone can help me troubleshoot a one way audio problem. >>>> >>>> First of all, everything works fine with Bria - I can call anywhere in >>>> any direction. >>>> >>>> The issue happens with Polycom IP450 phones. >>>> >>>> The setup had been working fine for weeks and weeks. Then one day, sound >>>> seems to progressively degrade when making outgoing calls. After a while, >>>> boom, the receiving party of calls originating from the IP450 hears us but >>>> we can't hear them. The problem lasted about 36 hours the went away on it's >>>> own or I am not sure what i did to make it go away. >>>> >>>> Now, two weeks later, this has happened again. >>>> >>>> I have updated the IP450 to the recommended version on the wiki. >>>> >>>> I have traced and traced. It does not seem to be a firewall/nat/helper >>>> issue as a) Bria works fine on the same subnet b) doing port copies on my >>>> switch, I can see the packets being sent to the hardphone. >>>> >>>> From my limited but growing understanding of SIP calls, the server -> >>>> phone direction seems doubled (though it might just look ok to me…). I >>>> can't >>>> hear a thing and the diagnostics / media statistics on the phone increment >>>> in the outgoing direction but not in the incoming direction. >>>> >>>> Anyone have any idea that could help me pin this down ? Might it just be >>>> a phone firmware issue ? >>>> >>>> I have supporting material if that can help. >>>> >>>> Thanks, >>>> >>>> Philippe. >>>> >>>> >>>> >>>> |||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||** >>>> ** >>>> >>>> Philippe Bechamp | Manager, IT / Security**** >>>> >>>> Neuralitic Systems Inc. >>>> >>>> >>>> M. +1.514.812.9609**** >>>> >>>> E. [email protected]**** >>>> >>>> This message and attachments are for the use of the addressee only and >>>> may be confidential. If you are not the intended recipient, please notify >>>> the sender immediately by e-mail and delete this message and attachments. >>>> Thank you.**** >>>> >>>> >>>> _______________________________________________ >>>> sipx-users mailing list >>>> [email protected] >>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>>> >>> <ATT00001.c> >>> >>> >>> |||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||*** >>> * >>> >>> Philippe Bechamp | Manager, IT / Security**** >>> >>> Neuralitic Systems Inc. >>> >>> >>> M. +1.514.812.9609**** >>> >>> E. [email protected]**** >>> >>> This message and attachments are for the use of the addressee only and >>> may be confidential. If you are not the intended recipient, please notify >>> the sender immediately by e-mail and delete this message and attachments. >>> Thank you.**** >>> >>> >>> _______________________________________________ >>> sipx-users mailing list >>> [email protected] >>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>> >> >> >> >> -- >> ====================== >> Tony Graziano, Manager >> Telephone: 434.984.8430 >> sip: [email protected] >> Fax: 434.465.6833 >> >> Email: [email protected] >> >> LAN/Telephony/Security and Control Systems Helpdesk: >> Telephone: 434.984.8426 >> sip: [email protected] >> >> Helpdesk Contract Customers: >> http://support.myitdepartment.net >> >> <http://support.myitdepartment.net/>Blog: >> http://blog.myitdepartment.net >> >> Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 >> >> Ask about our Internet Fax services! >> >> <ATT00001.c> >> >> >> |||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||**** >> >> Philippe Bechamp | Manager, IT / Security**** >> >> Neuralitic Systems Inc. >> >> >> M. +1.514.812.9609**** >> >> E. [email protected]**** >> >> This message and attachments are for the use of the addressee only and may >> be confidential. If you are not the intended recipient, please notify the >> sender immediately by e-mail and delete this message and attachments. Thank >> you.**** >> >> >> _______________________________________________ >> sipx-users mailing list >> [email protected] >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >> > > > > -- > ====================== > Tony Graziano, Manager > Telephone: 434.984.8430 > sip: [email protected] > Fax: 434.465.6833 > > Email: [email protected] > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > sip: [email protected] > > Helpdesk Contract Customers: > http://support.myitdepartment.net > > <http://support.myitdepartment.net>Blog: > http://blog.myitdepartment.net > > Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 > > Ask about our Internet Fax services! > > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.465.6833 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Contract Customers: http://support.myitdepartment.net <http://support.myitdepartment.net>Blog: http://blog.myitdepartment.net Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services!
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