FreePBX Setup for sip2sip.info

[jcusick]

I tried that, too, before I checked this forum, and just tried it again. Same results, tcpdump shows "udp port sip unreachable".

My other trunks are working fine, and have been for a couple of years. This is frustrating. I guess I'll try a trouble ticket with sip2sip.info.

Thanks,

JC

 

[vcallaway]

allow=h323 ????

I though H323 was a protocol not a codec.

 

[jcusick]

allow=h323 ????

I though H323 was a protocol not a codec.

And you thought right.

My mistake, I was thinking H323 and typed that instead of h263

However, even after changing that, using tcpdump I still get "udp port sip unreachable" reply on registration.

I'm using #"tcpdump src or dst sip2sip.info" and the reply is coming from redirect.dns-hosting.info

I find that confusing in and of itself, but I don't know how they are setup at their end and my PEER Details [now] matches that of others in this thread.

My other providers are working just fine (sipgate and voip.ms) so it would seem that it's not a firewall issue on my end.

I also checked their wiki for support, but they give no direct asterisk setup info.

Thanks for pointing out the protocol vs codec mixup here.

JC

 

[warraich]

JC Try your sip2sip account from a sip phone such as xlite and see if it is working!
Also as per sip2sip if your router has SIP ALG, Disable SIP ALG in some cases that can cause sip2sip not register. Also try nat=no.
sip2sip service is running fine I have 5 of my remote extensions registered with them and the are all working fine including my PBXinaFLASH.

 

[jcusick]

JC Try your sip2sip account from a sip phone such as xlite and see if it is working!
Also as per sip2sip if your router has SIP ALG, Disable SIP ALG in some cases that can cause sip2sip not register. Also try nat=no.
sip2sip service is running fine I have 5 of my remote extensions registered with them and the are all working fine including my PBXinaFLASH.

Definitely a puzzle

I downloaded Blink and set that up on my Ubuntu laptop behind the same router. Works fine as far as the incoming and echo tests are concerned. The only differences I see between the Blink setup and Asterisk/PIAF is that Blink uses an Outbound Proxy setting.

Interestingly I also tried their suggested software SIP Communicator and that would not connect either with the following error... which means nothing to me: We failed to create your account due to the following error: "Index: 0, Size: 0"

I added what I believe to be the equivalent to the PEER Details, "outboundproxy=proxy.sipthor.net", but unfortunately it changed nothing, so I dumped that addition. I also tried your suggestion of "nat=no", but no change there either.

I have used both host= sip2sip.info and proxy.sipthor.net. (as well as proxy.sipthor.net in the reg string). The only thing I can think of is that my eyes are shot all to h*** and I'm missing something very obvious, which is why I posted my setup last night... lucky for that in one respect, the codc error was caught.

Oh well, something will strike someone and the lightbulb will light... maybe even here! In the meantime I will hold off before contacting sip2sip for suggestions... I hate to bug them unnecessarily considering it's a free service.

Ultimately, though, what I suspect is that my dns server is not returning the proper SRV record since I have sip2sip.info in the registration string, but tcpdump is showing the reply from redirect.dns-hosting.info... That ain't right.

I'm gonna play with that piece next. That may be the lightbulb I've been waiting for!

Thanks again.

JC

 

[jcusick]

Success!

Ultimately, though, what I suspect is that my dns server is not returning the proper SRV record since I have sip2sip.info in the registration string, but tcpdump is showing the reply from redirect.dns-hosting.info... That ain't right.

I'm gonna play with that piece next. That may be the lightbulb I've been waiting for!

Well, that was it.

Apparently, not knowing how sip2sip's dns records are setup, I guessing my service provider's DNS Server was returning the A record of the proxy server instead of the registration server. On the PIAF box, I deleted the isp dns resolver and instead put in Google's DNS Server so that PIAF's requests would go straight to Google instead of my service provider (Google's is 8.8.8.8) restarted asterisk/amportal and viola! Registration occurred.

Thanks everyone, for being a great, and patient, sounding board while I worked my way through this issue.

Regards,

JC

 

[[email protected]]

How do you change the PIAF DNS reolver?

 

[randy7376]

It's in the same place it's always been..

See /etc/resolv.conf for DNS. If you want to set a IP/name lookup local to PIAF only, see /etc/hosts .

Unless you meant dnsmgr in Asterisk. Completely different thing...

 

[dallas]

For anyone struggling with sip2sip, take a look at this page. It is not straightforward and I don't know what can be in FreePBX and what has to be in sip_registration_custom.conf.

http://wiki.sip2sip.info/wiki/SipDevicesAsterisk#AsteriskPBX

 

[dallas]

Well it's certainly not straightforward... I still can't get it going. I get "Failed to authenticate on INVITE to '"80XXXXXX" <sip:[email protected]>" when I try to call out.Sip show registry is empty but the peers are listed with status OK.

 

[vcallaway]

One thing I just discovered is my ISP, CenturyLink hijacks the DNS queries.

This is becoming a common practice for some of these clowns. They do redirects to their search/spam page if a domain does not resolve.

Does not matter what you set your nameservers to. They redirect based on port.

Easy way to test is type in a bogus domain in a web page and see if your ISP search page comes up.