Free PBX 2.2.1 + sip trunks (register command is wrong)

I was trying to get a voip provider working on my box and used freepbx (under trixbox) and when the trunk was created all was fine. I saw traffic going out but never a register command to the voip provider I decided o check the configs b y hand and noticed>

ther register command in sip_additional.conf was set to:

register=XXXXXXX:[email protected]:5065

I had to create the above in sip.conf as

register => XXXXXXX:[email protected]:5065

and then it worked. where in the freebpx code is this? Is this a bug that no body as found?

jim

Thanks for your help. I got my zingotel SIP working. Here’s what I had to do:

Open router ports:

5004-5082 ; This range might be too big. Zingotel recommends 5060-5063.
10000-20000

Add this to my sip-nat.conf:

externip=xxx.xxx.xxx.xxx ; My internet IP address.
localnet=192.168.1.0/255.255.255.0 ; My subnet & mask.

Add this to my sip.conf [general] section:

defaultexpirey=600 ; Required by Zingotel.
insecure=invite ; Asterisks challenges the SIP for “secret” until this is set.

No dice. Tried that, no luck. Is the register string you’re using really the following format?

myphonenumber:[email protected]/myphonenumber

No I am not using the phonenumber behind its a port my voip provider users 5065

so is your ZINGOTEL registration string as follows?

phonenumber:[email protected]/5065

Where did you get the 5065 port? I thought SIP standards use 5060. This is not how Zingotel specifies the register string.

I’m a bit confused.

I am not using zingtel I had to put the 5065 that then end as thats what my voip provider was using

If your voip provider is using 5060 then do not put anything at the end as it will default to 5060

jim

I just did some testing here is what me sip.conf looks like

; Note: If your SIP devices are behind a NAT and your Asterisk
; server isn’t, try adding “nat=1” to each peer definition to
; solve translation problems.

[general]

bindport = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
allow=gsm
; If you need to answer unauthenticated calls, you should change this
; next line to ‘from-trunk’, rather than ‘from-sip-external’.
; You’ll know this is happening if when you call in you get a message
; saying "The number you have dialed is not in service. Please check the
; number and try again."
context = from-trunk; Send unknown SIP callers to this context
callerid = Unknown
tos=0x68

#include sip_additional.conf (this used to be last in this file under sip_custom.conf)

; #, in this configuration file, is NOT A COMMENT. This is exactly
; how it should be.
#include sip_nat.conf
#include sip_custom.conf

since moving sip_additional.conf up it works fine. it could be something with sip_custom.conf as I do not have this file

Try that and I bet it will work.

I’m guessing something else may have been broken. If you have added any contexts or similar to sip.conf it would break all register commands. In the current implementation of 2.2.1 all the register commands are put at the top of sip_additional.conf, expecting it to still be in the general context and can easily be broken if you mess around with some of the files.

This is fixed in 2.2 trunk and will be out with the next maintance release. It puts registrations in a specifc sip_registrations.conf file so they can be sure to be included in the proper place.

As far as = vs. => asterisk really does not care, treats them the same.

I’m having problems getting freePBX to register with my SIP provider, Zingotel. I was hopeful the change from [quote]=[/quote] to [quote]=>[/quote] would make a difference, but no luck. Asterisk seems to behave exactly the same.

DiscusZ, did you do anything else to get your registration to work?

BTW, my log shows register commands that time out and are repeated every 20 seconds.

ya tyhe = and => work the same, I assumed it was the later but i will have to look at my sip.conf for other changes. I added the registration there as it was not working in sip_additional.conf so right now I have 2 registrations for the same sip provider. I have not reinvestigated it as i ahve not been home the last the couple of days.

Jim

Any help would be greatly appreciated. I wonder if the registration string Zingotel suggested we use is wrong, and you typed something different in the sip.conf file, and that’s the one that’s working.

I’ve been all around this problem with no luck. Here is some more info I just posted on the Asterisk board…

[quote]…What amazes me is that I have a Sipura SPA-1000 on the same IP switch right next to my Asterisk box. When I plug the net cable into the SPA, it registers in a heartbeat.

But whatever I try at the Asterisk box, it will not register with Zingotel. I don’t think it’s getting out to the WAN. Every 20 seconds I get this message in my log file:

Registration for ‘[email protected]’ timed out, trying again (Attempt #13)

I turned debug on, and it shows nothing when a call comes in. Probably because Zingotel never routes a call because it has no registration. When I place an outbound call, my console shows the debug info. Oddly though, my console doesn’t show any of the failed registration timeouts.

When I use the asterisk CLI command, sip show registry, I get the following output:

Host Username Refresh State
sip1.whistlertel.com:5060 MYNUMBER 120 Request Sent
Verbosity is at least 1

So asterisk it sent a registration request, but I don’t think Zingotel ever got it.

I don’t know if this is a clue, but my system info shows the following message:

File Line Command Message
common_functions.php 314 file_exists(/proc/scsi/scsi) the file does not exist on your machine

BTW, I don’t have a scsi interface on my system.

Normally I’m usually able to bang through problems like this thtough brute force and trial-and-error. But I’m at my wits end and am hoping someone here can point me in a direction I haven’t yet tried.[/quote]

Many thanks in advance.