Hi Jon
Thanks for that. Trying to translate between the txt version and the xml
version, most of the settings seem to be about right.
I think the problem might be with the way we’re referring to the sip server in
the xml. The previous advice I got was to do it using the call manager section;
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<sipPort>5060</sipPort>
<securedSipPort>5061</securedSipPort>
</ports>
<processNodeName>1.2.3.4</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
and then later in the config
<sipLines>
<line button="1">
<featureID>9</featureID>
<featureLabel>SIPACCOUNT</featureLabel>
<name>SIPACCOUNT</name>
<displayName>SIPACCOUNT</displayName>
<contact>SIPACCOUNT</contact>
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<authName>SIPACCOUNT</authName>
<authPassword>MYPASSWORD</authPassword>
<sharedLine>false</sharedLine>
</line>
</sipLines>
It seems to be seeing the line name etc but never trying to contact the server.
Putting the SIP servers IP in the <proxy> field doesn’t seem to make any
difference.
Cheers
Mark
On 19 Jun 2014, at 11:51, Jonathan Hunter <hunter...@hotmail.com> wrote:
> Hi Mark,
>
> Yes we had exactly the same issue with a 7940.
>
> It is with the .cnf xml file.
>
> You need to make sure that the line name is populated;
>
> line1_name: "<<USERNAME>>"
>
> line1_authname: "<<USERNAME>>"
>
> line1_password: "supersecret"
>
> See an example file attached, hopefully will give you an idea.
>
> Thanks
>
> Jon
>
>
>
>
> From: m...@darkorigins.com
> Date: Thu, 19 Jun 2014 11:03:25 +0100
> To: sr-users@lists.sip-router.org
> Subject: Re: [SR-Users] Cisco 79XX Series Phones and TLS integration with
> kamailio
>
> Hi Jon
>
> I’m working with a variety of routers here at the moment (testing / lab
> environment :-) ranging from Mikrotik, Draytek to Firebricks.
>
> So far;
>
> - Phone on test is a 7970
>
> - Variety of routers here at the moment (testing / lab environment :-)
> ranging from Mikrotik, Draytek to Firebricks. All configured to NAT, no
> firewall filtering
>
> - SIP v9.3(1)SR4 firmware on the phone
>
> - Provisioning side ‘appears’ to be working. In that it requests the files,
> says ‘registering’
>
> - Doesn’t appear to generate any SIP traffic during registration (tcpdump on
> router/kamailio server)
>
>
> Wondering if I’ve got something wrong in the .cnf xml …
>
> Cheers
>
>
> Mark
>
>
> On 19 Jun 2014, at 10:48, Jonathan Hunter <hunter...@hotmail.com> wrote:
>
> Hi Mark,
>
> Sure of course, they are some what painful to get working due to their
> asymmetric NAT behaviour.
>
> What handset models are you working with, and what firewall devices do you
> have on site, as I have them working with a Cisco ASA on the network edge.
>
> Then I can give you some more details on at least them trying to register to
> kamailio.
>
> Thanks
>
> Jon
>
> From: m...@darkorigins.com
> Date: Thu, 19 Jun 2014 10:37:42 +0100
> To: sr-users@lists.sip-router.org
> Subject: Re: [SR-Users] Cisco 79XX Series Phones and TLS integration with
> kamailio
>
> Hi Jon
>
> It sounds like you’re a few steps in front of me. I’m just starting to try
> and get a working set of config files etc for provisioning to the phones.
> Which at the moment is going through the motions of registering but not
> actually sending anything.
>
> Would you mind sharing your experiences so far?
>
> Thanks
>
> Mark
>
>
> On 19 Jun 2014, at 09:50, Jonathan Hunter <hunter...@hotmail.com> wrote:
>
> Hi All,
>
> As you guys might remember I have been doing alot of work with these legacy
> handsets recently, in particular the Cisco IP phones 7945G and 7965G.
>
> They work well now with kamailio, using UDP or TCP as the transport protocol.
>
> I am now looking to implement SIP over TLS with them, and wondered if anyone
> had completed the same, as it appears they were designed very much to support
> SIP over TLS and SRTP but with Cisco call manager and not other SIP devices.
>
> I am trying to understand if its possible to integrate with kamailio, as from
> Cisco documentation it appears a CTL file (client trust list) is required
> which seems to be generated by cisco software, and the handset needs it
> before it will try to initiate a SIP connection over TLS.
>
> I have implemented TLS /SRTP with Cisco SPA's, Bria etc on kamailio fine, so
> its more a question around the handset and if anyone has achieved this.
>
> Thanks
>
> Jon
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> <7940examplefile.txt>_______________________________________________
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