I have two clients with Asterisk 1.8.21 (was on 1.8.17) who are experiencing a very debilitating problem:
- They have several accounts with the same SIP provider
- Each account corresponds to a direct-dial number
- Each number is supposed to go to a SIP telephone thus using the SIP provider as a PBX.
Use more than three telephones internally, have an internal PBX, use Cisco SCCP phones.
Install Asterisk; add additional SCCP support, use existing SIP phones together with new Cisco ones;
>>> replace direct SIP connections to SIP provider with SIP trunks <<<
Having more than two ‘register’ commands in the SIP.conf file trigger
registration timeout messages.
The problem appears very similar to what this bug describes
However, I was never able to apply the patch - possibly because it is for an earlier version of Asterisk.
I need [at leaset three] registrations to my SIP provider so I can uniquely handle each incoming line accordingly:
[general] bindport=5060 bindaddr=0.0.0.0 context=default disallow=all allow=gsm allow=ilbc allow=ulaw allow=alaw srvlookup=yes register => 1111:firstname.lastname@example.org/1111 register => 2222:email@example.com/2222 register => 3333:firstname.lastname@example.org/3333  type=friend host=dynamic defaultip=192.168.1.12 username=1111 secret=pass1 context=default dtmfmode=rfc2833 nat=no  type=friend host=dynamic defaultip=192.168.1.12 username=2222 secret=pass2 context=default dtmfmode=rfc2833 nat=no  type=friend host=dynamic defaultip=192.168.1.12 username=3333 secret=pass3 context=default dtmfmode=rfc2833 nat=no [provider_line1] type=peer host=sip.provider.net username=1111 secret=pass1 context=testA fromuser=1111 fromdomain=sip.provider.net nat=no [provider_line2] type=peer host=sip.provider.net username=2222 secret=pass2 context=testB fromuser=2222 fromdomain=sip.provider.net nat=no [provider_line3] type=peer host=sip.provider.net username=3333 secret=pass3 context=testC fromuser=3333 fromdomain=sip.provider.net nat=no [provider-in] type=peer host=sip.provider.net context=default insecure=port,invite nat=no
If all three ‘register’ commands are active Asterisk starts yielding ‘connection timeout’ for each of them !
If I comment out ONE of these - all appears to work just fine!
In either case dial out is OK.
Of course, if any of the register commands is commented the relevant incoming number is no longer reachable from outside.
I have been struggling with this problem for over three months now and there seems some patchy information here and there of people experiencing similar issue, but no solution to the issue at all.
Any help would be much appreciated!