G'day folks
Struggling to work out whether Kamailio can help me or not in this situation, and I hope someone can tell me whether I'm barking up the wrong tree.
I have hundreds of users which register to an old legacy Asterisk box that needs retiring. The "users" are customers consisting of VoIP ATAs, PBXes, softphones, and other hardware phones.
I've been told I need to retire the Asterisk system in the next month and move customers to our new provider, who provide the SBC themselves (it's a Broadsoft-based system). Obviously a customer once ported to the new provider requires new SIP proxy and SIP domain settings configured in their CPE.
For obvious reasons, I cannot require hundreds of customers to change their configuration overnight. However, there is not enough time to assist hundreds of customers changing their SIP proxy/registrar/domain details.
So in the meantime I'm trying to work out a transition solution. I cannot simply change my SIP registrar DNS to be a CNAME for the new provider's registrar, as the domain is different, and the new provider doesn't allow domains other than their own to register with them (even if the username/password portion is correct).
My new provider has previously told me they are unwilling to accept my old domain whilst authenticating (they are much bigger than me, and I have no power to change them).
So I was hoping to use Kamailio and possibly the Path or Dispatch modules to proxy SIP registration to the new provider, rewriting the domain part of the SIP register string as I go
But I can't for the life of me work out whether that's even possible. I have spent weeks poring over various Kamailio configurations posted to the interwebs, but nobody seems to have done this before.
Specifically the "translation proxy" -- which I am hoping to make Kamailio -- will need to be stateless, and perform no local authentication. Individual registrations will need to be passed through to the new provider (including authentication!), while still translating the domain in the Request-URI, To, From, and various other SIP headers.
Because the current legacy Asterisk box has nat=yes set for all users, I will need to proxy media, because if I start proxying SIP traffic to the new provider, I will be confusing their own NAT hacks.
I don't understand the documentation for either the Path or Dispatch modules, and the information on the wiki is confusing and poorly written. I am very familiar with SIP, and have no trouble configuring Asterisk, but trying to work with Kamailio is leaving me feeling like an idiot.
However, Kamailio's config is very flexible, so despite this I suspect what I'm trying to achieve may be possible.
Any pointers in the right direction?
Cheers, Jeremy.
On Monday 07 July 2014 13:04:26 Jeremy Visser wrote:
I am very familiar with SIP, and have no trouble configuring Asterisk, but trying to work with Kamailio is leaving me feeling like an idiot.
I know exactly how you feeling, the excellent Advanced Kamailio Training managed to fix that :)
But back to your actual problem, my guess is kamailio can solve your problem (I'm far from an expert). But considering your deadline and familiarity with asterisk: do it with asterisk instead. You already have the config for handling the customer side, you only need to implement the logic to/from your upstream (having asterisk register and bridge the traffic both ways).
Hi Jeremy!
You have some functions in the uac-module to change From and To, see here: http://kamailio.org/docs/modules/stable/modules/uac.html
I havn't tried this on REGISTER messages, just INVITES. But in the later case it works fine to normalize the number to the format the upstream provider wants. I also change the domain at the same time that seems to be what you would like to do.
/Johan Wilfer
2014-07-07 13:04, Jeremy Visser skrev:
G'day folks
Struggling to work out whether Kamailio can help me or not in this situation, and I hope someone can tell me whether I'm barking up the wrong tree.
I have hundreds of users which register to an old legacy Asterisk box that needs retiring. The "users" are customers consisting of VoIP ATAs, PBXes, softphones, and other hardware phones.
I've been told I need to retire the Asterisk system in the next month and move customers to our new provider, who provide the SBC themselves (it's a Broadsoft-based system). Obviously a customer once ported to the new provider requires new SIP proxy and SIP domain settings configured in their CPE.
For obvious reasons, I cannot require hundreds of customers to change their configuration overnight. However, there is not enough time to assist hundreds of customers changing their SIP proxy/registrar/domain details.
So in the meantime I'm trying to work out a transition solution. I cannot simply change my SIP registrar DNS to be a CNAME for the new provider's registrar, as the domain is different, and the new provider doesn't allow domains other than their own to register with them (even if the username/password portion is correct).
My new provider has previously told me they are unwilling to accept my old domain whilst authenticating (they are much bigger than me, and I have no power to change them).
So I was hoping to use Kamailio and possibly the Path or Dispatch modules to proxy SIP registration to the new provider, rewriting the domain part of the SIP register string as I go
But I can't for the life of me work out whether that's even possible. I have spent weeks poring over various Kamailio configurations posted to the interwebs, but nobody seems to have done this before.
Specifically the "translation proxy" -- which I am hoping to make Kamailio -- will need to be stateless, and perform no local authentication. Individual registrations will need to be passed through to the new provider (including authentication!), while still translating the domain in the Request-URI, To, From, and various other SIP headers.
Because the current legacy Asterisk box has nat=yes set for all users, I will need to proxy media, because if I start proxying SIP traffic to the new provider, I will be confusing their own NAT hacks.
I don't understand the documentation for either the Path or Dispatch modules, and the information on the wiki is confusing and poorly written. I am very familiar with SIP, and have no trouble configuring Asterisk, but trying to work with Kamailio is leaving me feeling like an idiot.
However, Kamailio's config is very flexible, so despite this I suspect what I'm trying to achieve may be possible.
Any pointers in the right direction?
Cheers, Jeremy.
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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Daniel Tryba -
Jeremy Visser -
Johan Wilfer