Provisioning with server behind NAT

[shopcomputer]

I am trying to provision a snom phone, where the pbx is behind NAT. The SIP IP replacement does not work for the htttp provisioning it seems. Is there anywhere I can set an ip replacement for the provisioning?

 

http://server_public_address/provisioning/...-macaddress.htm

returns the private ip

<?xml version="1.0" encoding="utf-8" ?>

- <setting-files>

<file url="http://192.168.10.2:8000/provisioning/snom_3xx_phone-000413290587.xml?model=snom360" />

<file url="http://192.168.10.2:8000/provisioning/snom_3xx_fkeys-000413290587.xml" />

<file url="http://192.168.10.2:8000/provisioning/snom_web_lang.xml" />

<file url="http://192.168.10.2:8000/provisioning/snom_gui_lang.xml" />

</setting-files>

[hosted]

the only way I know of to control the provisioning IP is the domain IP itself.. are all the phones behind the nat? if so just change the domain IP

[shopcomputer]

the only way I know of to control the provisioning IP is the domain IP itself.. are all the phones behind the nat? if so just change the domain IP

What do you mean by domain IP?

[hosted]

You know when you login and press the domain tab. you have your domain listed in there. that is the FQDN or IP that the provisioning system uses for the PnP.

 

so if all your phones are outside of the NAT change the domain IP to be outside the NAT so the PnP works properly.

 

if they are a mixture on inside/outside of the nat. I usually manually configure the remote IP phones.

[shopcomputer]

You know when you login and press the domain tab. you have your domain listed in there. that is the FQDN or IP that the provisioning system uses for the PnP.

 

so if all your phones are outside of the NAT change the domain IP to be outside the NAT so the PnP works properly.

 

if they are a mixture on inside/outside of the nat. I usually manually configure the remote IP phones.

I think you are referring to the domain name, that has seems to have no effect on what IP the provisiong provides. All users are remote in this case.

[hosted]

yes I always use IP's for my 'domain names' (to many bad experiences with flakey DNS systems that cause outages)

 

That IP is pulled into the PnP provisioning for me.

[shopcomputer]

yes I always use IP's for my 'domain names' (to many bad experiences with flakey DNS systems that cause outages)

 

That IP is pulled into the PnP provisioning for me.

I had localhost, changed to IP, did not change the PnP, although routing list, under ports, seems to change the PnP IP.

[hosted]

i though routing list didnt affect the PnP config, I will have to test this. Make sure the domain IP is the real IP and not an alias. you can use aliases but it doesnt do anything to the PnP.

[Vodia PBX]

I am trying to provision a snom phone, where the pbx is behind NAT. The SIP IP replacement does not work for the htttp provisioning it seems. Is there anywhere I can set an ip replacement for the provisioning?

 

That's still my favourite: http://wiki.pbxnsip.com/index.php/Office_w...ic_IP_addresses. The "Separation by Route" also applies to the generation of IP addresses for plug and play.

 

Of course, it is not recommended and very support unfriendly. But it should work ...

[shopcomputer]

That's still my favourite: http://wiki.pbxnsip.com/index.php/Office_w...ic_IP_addresses. The "Separation by Route" also applies to the generation of IP addresses for plug and play.

 

Of course, it is not recommended and very support unfriendly. But it should work ...

Yes I set my remote phone back to manual configuration, routing rules, were causing RTP no audio issues, I just was playing with it, this is in my demo system that is behind a CiSCO Pix in a data center. Their SIP ALG works good, when the phone is manually configured.

[Vodia PBX]

Yes I set my remote phone back to manual configuration, routing rules, were causing RTP no audio issues, I just was playing with it, this is in my demo system that is behind a CiSCO Pix in a data center. Their SIP ALG works good, when the phone is manually configured.

 

Oh yea, maybe Cisco should consider supporting PnP ALG as well :-)

[hosted]

Yes I set my remote phone back to manual configuration, routing rules, were causing RTP no audio issues, I just was playing with it, this is in my demo system that is behind a CiSCO Pix in a data center. Their SIP ALG works good, when the phone is manually configured.

 

Use Edgemarc, they have the best SIP ALG. it even gives you MOS/jitter scores for each call. very nice and pricey.

I just with they had TLS ALG. haha not