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Provisioning with server behind NAT


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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>

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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.

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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.

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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 ...

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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.

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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 :-)

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  • 2 weeks later...
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

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