[OpenSIPS-Users] Users Digest, Vol 38, Issue 65

Thu Sep 29 04:45:17 CEST 2011

Sent from my Verizon Wireless Sony Ericsson Xperia™ PLAY.

users-request at lists.opensips.org wrote:

>Send Users mailing list submissions to
>	users at lists.opensips.org
>
>To subscribe or unsubscribe via the World Wide Web, visit
>	http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>or, via email, send a message with subject or body 'help' to
>	users-request at lists.opensips.org
>
>You can reach the person managing the list at
>	users-owner at lists.opensips.org
>
>When replying, please edit your Subject line so it is more specific
>than "Re: Contents of Users digest..."
>
>
>Today's Topics:
>
>   1. Re: Sip invite sent,	not reaching dest from certain phones
>      (Vallimamod ABDULLAH)
>   2. Re: Sip invite sent,	not reaching dest from certain phones
>      (Schneur Rosenberg)
>   3. Re: Sip invite sent,	not reaching dest from certain phones
>      (Brett Nemeroff)
>   4. Re: Sip invite sent,	not reaching dest from certain phones
>      (Schneur Rosenberg)
>   5. Re: Sip invite sent,	not reaching dest from certain phones
>      (Vallimamod ABDULLAH)
>   6. Re: Sip invite sent,	not reaching dest from certain phones
>      (Schneur Rosenberg)
>
>
>----------------------------------------------------------------------
>
>Message: 1
>Date: Thu, 22 Sep 2011 00:03:01 +0200
>From: Vallimamod ABDULLAH <vallimamod.abdullah at imtelecom.fr>
>Subject: Re: [OpenSIPS-Users] Sip invite sent,	not reaching dest from
>	certain phones
>To: OpenSIPS users mailling list <users at lists.opensips.org>
>Message-ID: <26841AA5-4A22-4280-9DB2-1E21756F053D at imtelecom.fr>
>Content-Type: text/plain; charset=iso-8859-1
>
>Hi Schneur,
>
>What do you mean precisely by never hitting the asterisk server ?
>As your ngrep trace shows, both packets are sent over the wire to the exact same address (68.233.222.9:5060) so they should both reach Asterisk. But it's possible that the latter doesn't treat them the same way, depending on nat issues most of the time (Asterisk send replies to the contact header URI by default if I recall correctly...)
>
>Try to make a ngrep trace on the iface attached to 68.233.222.9 and check if Asterisk does not send the answer to a different IP. Also enable the debug log on the asterisk console to spot any error / warning messages or sip retransmissions.
>
>Hope this would help.
>
>Regards,
>-vma
>.
>
>On Sep 21, 2011, at 11:43 PM, Schneur Rosenberg wrote:
>
>> NO These are the invites going from the opensips to the asterisk NOT
>> the ones from the phone, I did a ngrep on the asterisk box and the
>> packet never reaches it, both opensips and asterisk are open no NAT,
>> the phones are behind a nat as you can see in the sip packets
>> 
>> 
>> On Thu, Sep 22, 2011 at 12:37 AM, Duane Larson <duane.larson at gmail.com> wrote:
>>> These are the INVITES that are coming from your Phones correct?  These won't
>>> help to troubleshoot I don't think.  You will need to show the INVITES that
>>> are leaving OpenSIPS and heading towards your Asterisk server.
>>> 
>>> Honestly if your opensips.cfg does the exact same thing for linksys and
>>> aastra phones I can't see it being an opensips issue.  That's just a guess
>>> since I don't have anything to go on.
>>> 
>>> On Wed, Sep 21, 2011 at 4:25 PM, Schneur Rosenberg
>>> <rosenberg11219 at gmail.com> wrote:
>>>> 
>>>> I'm pretty new to opensips, I'm having a interesting problem, I use my
>>>> opensips for loadbalancing purposes I'm trying to place a call, and
>>>> from My linksys phone everything works fine, call comes into opensips
>>>> and opensips sends it to my asterisk system and call goes through
>>>> properly, from other phone (Aastra) Opensips accept the call, it even
>>>> sends it to the Asterisk but in never hits the asterisk server, can
>>>> anyone please review the 2 invites and let me know why second invite
>>>> gets lost, and how I can fix it
>>>> 
>>>> Here is the invite from the Linksys that worked
>>>> 
>>>> U 64.69.40.120:5060 -> 68.233.222.9:5060
>>>> INVITE sip:61 at 68.233.222.9:5060 SIP/2.0.
>>>> Record-Route: <sip:64.69.40.120;lr=on>.
>>>> Via: SIP/2.0/UDP 64.69.40.120;branch=z9hG4bK9857.800b0681.0.
>>>> Via: SIP/2.0/UDP
>>>> 
>>>> 192.168.1.104:5060;rport=5060;received=173.220.6.65;branch=z9hG4bK-4194567e.
>>>> From: solhome5
>>>> <sip:solhome5 at opensips.myserverip.com>;tag=833ac73613f3482o0.
>>>> To: <sip:61 at opensips.myserverip.com>.
>>>> Remote-Party-ID: solhome5
>>>> <sip:solhome5 at opensips.myserverip.com>;screen=yes;party=calling.
>>>> Call-ID: 78a92c07-62e399fe at 192.168.1.104.
>>>> CSeq: 102 INVITE.
>>>> Max-Forwards: 69.
>>>> Contact: solhome5 <sip:solhome5 at 173.220.6.65:5060;nat=yes>.
>>>> Expires: 240.
>>>> User-Agent: Linksys/SPA2102-5.2.12.
>>>> Content-Length: 446.
>>>> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER.
>>>> Supported: x-sipura, replaces.
>>>> Content-Type: application/sdp.
>>>> 
>>>> Here is the invite of the Aastra that did not work
>>>> 
>>>> U 64.69.40.120:5060 -> 68.233.222.9:5060
>>>> INVITE sip:61 at 68.233.222.9:5060;user=phone SIP/2.0.
>>>> Record-Route: <sip:64.69.40.120;lr=on>.
>>>> Via: SIP/2.0/UDP 64.69.40.120;branch=z9hG4bK847d.93e9e112.0.
>>>> Via: SIP/2.0/UDP
>>>> 
>>>> 192.168.1.102;rport=32857;received=173.220.6.65;branch=z9hG4bK7d9e53a542fc6ebe6.6e57a6889920eccf1.
>>>> Max-Forwards: 69.
>>>> From: "test2" <sip:test2 at opensips.myserverip.com:5060>;tag=ef646132b8.
>>>> To: <sip:61 at opensips.myserverip.com:5060;user=phone>.
>>>> Call-ID: f12b5324f31c0d30.
>>>> CSeq: 20777 INVITE.
>>>> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE,
>>>> PRACK, SUBSCRIBE, INFO.
>>>> Allow-Events: talk, hold, conference, LocalModeStatus.
>>>> Contact: "test2"
>>>> 
>>>> <sip:test2 at 173.220.6.65:32857;transport=udp;nat=yes>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D2A6C9E>".
>>>> Supported: path, 100rel, replaces.
>>>> User-Agent: Aastra 57iCT/3.2.2.56.
>>>> Content-Type: application/sdp.
>>>> Content-Length: 630.
>>>> 
>>>> _______________________________________________
>>>> Users mailing list
>>>> Users at lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>> 
>>> 
>>> 
>>> --
>>> --
>>> *--*--*--*--*--*
>>> Duane
>>> *--*--*--*--*--*
>>> --
>>> 
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>> 
>>> 
>> 
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
>
>------------------------------
>
>Message: 2
>Date: Thu, 22 Sep 2011 01:07:01 +0300
>From: Schneur Rosenberg <rosenberg11219 at gmail.com>
>Subject: Re: [OpenSIPS-Users] Sip invite sent,	not reaching dest from
>	certain phones
>To: OpenSIPS users mailling list <users at lists.opensips.org>
>Message-ID:
>	<CANvjR0U66tgJeeQb+tO+AHF9=K_RTMu2GFDOVt2T93jjP9uESw at mail.gmail.com>
>Content-Type: text/plain; charset=ISO-8859-1
>
>The packet does not reach asterisk, I did a ngrep on the asterisk
>server and not a single packet arrives from the opensips when using
>the Aastra phone, therefore its not sending back anything, the
>asterisk CLI is also quiet nothing whatsoever :-(
>
>On Thu, Sep 22, 2011 at 1:03 AM, Vallimamod ABDULLAH
><vallimamod.abdullah at imtelecom.fr> wrote:
>> Hi Schneur,
>>
>> What do you mean precisely by never hitting the asterisk server ?
>> As your ngrep trace shows, both packets are sent over the wire to the exact same address (68.233.222.9:5060) so they should both reach Asterisk. But it's possible that the latter doesn't treat them the same way, depending on nat issues most of the time (Asterisk send replies to the contact header URI by default if I recall correctly...)
>>
>> Try to make a ngrep trace on the iface attached to 68.233.222.9 and check if Asterisk does not send the answer to a different IP. Also enable the debug log on the asterisk console to spot any error / warning messages or sip retransmissions.
>>
>> Hope this would help.
>>
>> Regards,
>> -vma
>> .
>>
>> On Sep 21, 2011, at 11:43 PM, Schneur Rosenberg wrote:
>>
>>> NO These are the invites going from the opensips to the asterisk NOT
>>> the ones from the phone, I did a ngrep on the asterisk box and the
>>> packet never reaches it, both opensips and asterisk are open no NAT,
>>> the phones are behind a nat as you can see in the sip packets
>>>
>>>
>>> On Thu, Sep 22, 2011 at 12:37 AM, Duane Larson <duane.larson at gmail.com> wrote:
>>>> These are the INVITES that are coming from your Phones correct? ?These won't
>>>> help to troubleshoot I don't think. ?You will need to show the INVITES that
>>>> are leaving OpenSIPS and heading towards your Asterisk server.
>>>>
>>>> Honestly if your opensips.cfg does the exact same thing for linksys and
>>>> aastra phones I can't see it being an opensips issue. ?That's just a guess
>>>> since I don't have anything to go on.
>>>>
>>>> On Wed, Sep 21, 2011 at 4:25 PM, Schneur Rosenberg
>>>> <rosenberg11219 at gmail.com> wrote:
>>>>>
>>>>> I'm pretty new to opensips, I'm having a interesting problem, I use my
>>>>> opensips for loadbalancing purposes I'm trying to place a call, and
>>>>> from My linksys phone everything works fine, call comes into opensips
>>>>> and opensips sends it to my asterisk system and call goes through
>>>>> properly, from other phone (Aastra) Opensips accept the call, it even
>>>>> sends it to the Asterisk but in never hits the asterisk server, can
>>>>> anyone please review the 2 invites and let me know why second invite
>>>>> gets lost, and how I can fix it
>>>>>
>>>>> Here is the invite from the Linksys that worked
>>>>>
>>>>> U 64.69.40.120:5060 -> 68.233.222.9:5060
>>>>> INVITE sip:61 at 68.233.222.9:5060 SIP/2.0.
>>>>> Record-Route: <sip:64.69.40.120;lr=on>.
>>>>> Via: SIP/2.0/UDP 64.69.40.120;branch=z9hG4bK9857.800b0681.0.
>>>>> Via: SIP/2.0/UDP
>>>>>
>>>>> 192.168.1.104:5060;rport=5060;received=173.220.6.65;branch=z9hG4bK-4194567e.
>>>>> From: solhome5
>>>>> <sip:solhome5 at opensips.myserverip.com>;tag=833ac73613f3482o0.
>>>>> To: <sip:61 at opensips.myserverip.com>.
>>>>> Remote-Party-ID: solhome5
>>>>> <sip:solhome5 at opensips.myserverip.com>;screen=yes;party=calling.
>>>>> Call-ID: 78a92c07-62e399fe at 192.168.1.104.
>>>>> CSeq: 102 INVITE.
>>>>> Max-Forwards: 69.
>>>>> Contact: solhome5 <sip:solhome5 at 173.220.6.65:5060;nat=yes>.
>>>>> Expires: 240.
>>>>> User-Agent: Linksys/SPA2102-5.2.12.
>>>>> Content-Length: 446.
>>>>> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER.
>>>>> Supported: x-sipura, replaces.
>>>>> Content-Type: application/sdp.
>>>>>
>>>>> Here is the invite of the Aastra that did not work
>>>>>
>>>>> U 64.69.40.120:5060 -> 68.233.222.9:5060
>>>>> INVITE sip:61 at 68.233.222.9:5060;user=phone SIP/2.0.
>>>>> Record-Route: <sip:64.69.40.120;lr=on>.
>>>>> Via: SIP/2.0/UDP 64.69.40.120;branch=z9hG4bK847d.93e9e112.0.
>>>>> Via: SIP/2.0/UDP
>>>>>
>>>>> 192.168.1.102;rport=32857;received=173.220.6.65;branch=z9hG4bK7d9e53a542fc6ebe6.6e57a6889920eccf1.
>>>>> Max-Forwards: 69.
>>>>> From: "test2" <sip:test2 at opensips.myserverip.com:5060>;tag=ef646132b8.
>>>>> To: <sip:61 at opensips.myserverip.com:5060;user=phone>.
>>>>> Call-ID: f12b5324f31c0d30.
>>>>> CSeq: 20777 INVITE.
>>>>> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE,
>>>>> PRACK, SUBSCRIBE, INFO.
>>>>> Allow-Events: talk, hold, conference, LocalModeStatus.
>>>>> Contact: "test2"
>>>>>
>>>>> <sip:test2 at 173.220.6.65:32857;transport=udp;nat=yes>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D2A6C9E>".
>>>>> Supported: path, 100rel, replaces.
>>>>> User-Agent: Aastra 57iCT/3.2.2.56.
>>>>> Content-Type: application/sdp.
>>>>> Content-Length: 630.
>>>>>
>>>>> _______________________________________________
>>>>> Users mailing list
>>>>> Users at lists.opensips.org
>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>>
>>>>
>>>> --
>>>> --
>>>> *--*--*--*--*--*
>>>> Duane
>>>> *--*--*--*--*--*
>>>> --
>>>>
>>>> _______________________________________________
>>>> Users mailing list
>>>> Users at lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
>
>
>------------------------------
>
>Message: 3
>Date: Wed, 21 Sep 2011 17:09:07 -0500
>From: Brett Nemeroff <brett at nemeroff.com>
>Subject: Re: [OpenSIPS-Users] Sip invite sent,	not reaching dest from
>	certain phones
>To: OpenSIPS users mailling list <users at lists.opensips.org>
>Message-ID:
>	<CAPwC5ww8n11k+ucme14KkDWA2m-mODfPjAEG=ys8YOSk4KHa5A at mail.gmail.com>
>Content-Type: text/plain; charset="iso-8859-1"
>
>On Wed, Sep 21, 2011 at 5:03 PM, Vallimamod ABDULLAH <
>vallimamod.abdullah at imtelecom.fr> wrote:
>
>> Hi Schneur,
>>
>> What do you mean precisely by never hitting the asterisk server ?
>> As your ngrep trace shows, both packets are sent over the wire to the exact
>> same address (68.233.222.9:5060) so they should both reach Asterisk. But
>> it's possible that the latter doesn't treat them the same way, depending on
>> nat issues most of the time (Asterisk send replies to the contact header URI
>> by default if I recall correctly...)
>>
>
>I think asterisk does reply to the contact header and they are obviously
>different in the two traces. You'll see one is port 5060 and the other is
>based on some NAT translation. Need to find out why those are different..
>
>-Brett
>-------------- next part --------------
>An HTML attachment was scrubbed...
>URL: <http://lists.opensips.org/pipermail/users/attachments/20110921/b9232d06/attachment-0001.htm>
>
>------------------------------
>
>Message: 4
>Date: Thu, 22 Sep 2011 01:12:42 +0300
>From: Schneur Rosenberg <rosenberg11219 at gmail.com>
>Subject: Re: [OpenSIPS-Users] Sip invite sent,	not reaching dest from
>	certain phones
>To: OpenSIPS users mailling list <users at lists.opensips.org>
>Message-ID:
>	<CANvjR0XWxur1vRGXBWxwuLri6G1ZgZb_SXKwU2Z5sjaqx4GKuA at mail.gmail.com>
>Content-Type: text/plain; charset=ISO-8859-1
>
>If the packet would of reached asterisk then you might of been right,
>problem is a ngrep trace does not show a single packet reaching it.
>
>On Thu, Sep 22, 2011 at 1:09 AM, Brett Nemeroff <brett at nemeroff.com> wrote:
>> On Wed, Sep 21, 2011 at 5:03 PM, Vallimamod ABDULLAH
>> <vallimamod.abdullah at imtelecom.fr> wrote:
>>>
>>> Hi Schneur,
>>>
>>> What do you mean precisely by never hitting the asterisk server ?
>>> As your ngrep trace shows, both packets are sent over the wire to the
>>> exact same address (68.233.222.9:5060) so they should both reach Asterisk.
>>> But it's possible that the latter doesn't treat them the same way, depending
>>> on nat issues most of the time (Asterisk send replies to the contact header
>>> URI by default if I recall correctly...)
>>
>> I think asterisk does reply to the contact header and they are obviously
>> different in the two traces. You'll see one is port 5060 and the other is
>> based on some NAT translation. Need to find out why those are different..
>>
>> -Brett
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
>
>------------------------------
>
>Message: 5
>Date: Thu, 22 Sep 2011 00:24:15 +0200
>From: Vallimamod ABDULLAH <vallimamod.abdullah at imtelecom.fr>
>Subject: Re: [OpenSIPS-Users] Sip invite sent,	not reaching dest from
>	certain phones
>To: OpenSIPS users mailling list <users at lists.opensips.org>
>Message-ID: <E6BE8D8F-F96B-43AB-B5E9-F0432CCC754D at imtelecom.fr>
>Content-Type: text/plain; charset=windows-1252
>
>Then you have any intermediate device (known or unknown) that does filtering or mangling in some way?
>Try to trace the sip packet on every hop between the 2 servers to see how far it goes.
>
>Regards,
>- vma
>.
>
>On Sep 22, 2011, at 12:07 AM, Schneur Rosenberg wrote:
>
>> The packet does not reach asterisk, I did a ngrep on the asterisk
>> server and not a single packet arrives from the opensips when using
>> the Aastra phone, therefore its not sending back anything, the
>> asterisk CLI is also quiet nothing whatsoever :-(
>> 
>> On Thu, Sep 22, 2011 at 1:03 AM, Vallimamod ABDULLAH
>> <vallimamod.abdullah at imtelecom.fr> wrote:
>>> Hi Schneur,
>>> 
>>> What do you mean precisely by never hitting the asterisk server ?
>>> As your ngrep trace shows, both packets are sent over the wire to the exact same address (68.233.222.9:5060) so they should both reach Asterisk. But it's possible that the latter doesn't treat them the same way, depending on nat issues most of the time (Asterisk send replies to the contact header URI by default if I recall correctly...)
>>> 
>>> Try to make a ngrep trace on the iface attached to 68.233.222.9 and check if Asterisk does not send the answer to a different IP. Also enable the debug log on the asterisk console to spot any error / warning messages or sip retransmissions.
>>> 
>>> Hope this would help.
>
>
>
>
>------------------------------
>
>Message: 6
>Date: Thu, 22 Sep 2011 01:40:18 +0300
>From: Schneur Rosenberg <rosenberg11219 at gmail.com>
>Subject: Re: [OpenSIPS-Users] Sip invite sent,	not reaching dest from
>	certain phones
>To: OpenSIPS users mailling list <users at lists.opensips.org>
>Message-ID:
>	<CANvjR0UQCNUQJOYAvAwDn4AhZvzMaF9PXiVz793s7OYarQCHMg at mail.gmail.com>
>Content-Type: text/plain; charset=windows-1252
>
>both systems are on the open internet, I have no firewalls etc on any
>of the systems, I will try another 2 systems with same configurations
>and see what happens.
>
>On Thu, Sep 22, 2011 at 1:24 AM, Vallimamod ABDULLAH
><vallimamod.abdullah at imtelecom.fr> wrote:
>> Then you have any intermediate device (known or unknown) that does filtering or mangling in some way?
>> Try to trace the sip packet on every hop between the 2 servers to see how far it goes.
>>
>> Regards,
>> - vma
>> .
>>
>> On Sep 22, 2011, at 12:07 AM, Schneur Rosenberg wrote:
>>
>>> The packet does not reach asterisk, I did a ngrep on the asterisk
>>> server and not a single packet arrives from the opensips when using
>>> the Aastra phone, therefore its not sending back anything, the
>>> asterisk CLI is also quiet nothing whatsoever :-(
>>>
>>> On Thu, Sep 22, 2011 at 1:03 AM, Vallimamod ABDULLAH
>>> <vallimamod.abdullah at imtelecom.fr> wrote:
>>>> Hi Schneur,
>>>>
>>>> What do you mean precisely by never hitting the asterisk server ?
>>>> As your ngrep trace shows, both packets are sent over the wire to the exact same address (68.233.222.9:5060) so they should both reach Asterisk. But it's possible that the latter doesn't treat them the same way, depending on nat issues most of the time (Asterisk send replies to the contact header URI by default if I recall correctly...)
>>>>
>>>> Try to make a ngrep trace on the iface attached to 68.233.222.9 and check if Asterisk does not send the answer to a different IP. Also enable the debug log on the asterisk console to spot any error / warning messages or sip retransmissions.
>>>>
>>>> Hope this would help.
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
>
>
>------------------------------
>
>_______________________________________________
>Users mailing list
>Users at lists.opensips.org
>http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>End of Users Digest, Vol 38, Issue 65
>*************************************