I upgraded to the 2.5 Beta last night using the upgrade module, following the steps as prescribed. Everything seemed to be working fine last night. When I got to work today my Aastra 9133i says no service. I’ve checked the settings and rebooted the phone. The previously working SIP softphone on my laptop also does not register remotely now. I’ve double checked settings on both phones and in Asterisk. I’ve created new extensions to test unsuccessfully and checked the router settings on both ends, neither of which changed between yesterday and today. I can setup and connect an IAX connection with the softphone. I’m open to suggestions! Thanks.
my guess is you had edited sip.conf to add required settings like externip and localnet, sip.conf will regularly get overwritten any time a core module is upgraded. If that is the issue, you need to put those settings in sip_nat.conf (or sip_general_custom.conf).
OK - I checked the various sip.conf files vs. the ones backed up a few days ago and they are all the same including the externip and localnet settings.
Are you using externhost or externip in sip_nat.conf ?
If your using externhost and a dyndns name, try putting the ip in there or using a alternative to dyndns.
Could you in the CLI do sip set debug on and then try to register the phone. Then paste the sip trace here?
That would give us some more clue to what has happened
In sip_nat.conf I have:
externip = dynamic.host.com
localnet = 10.2.1.0/255.255.255.0
I tried changing it to the IP, using externhost with both IP and dynamic address and both externip and externhost. No change for the remote phone registering.
Here’s the trace. It appears it is receiving the request and trying to respond but not getting through now.
<— SIP read from 72.149.XXX.XXX:1028 —>
REGISTER sip:98.121.XXX.XXX:5060 SIP/2.0
Via: SIP/2.0/UDP 72.149.XXX.XXX:1028;branch=z9hG4bKea6676234
Max-Forwards: 70
Content-Length: 0
To: Will at Work sip:[email protected]:5060
From: Will at Work sip:[email protected]:5060;tag=5ac81cfd2d5773b
Call-ID: [email protected]
CSeq: 1564258181 REGISTER
Contact: Will at Work sip:[email protected]:1028;transport=udp
Allow-Events: talk,hold,conference
Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO
User-Agent: Aastra 9133i/1.4.2.3000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45
<— Transmitting (NAT) to 72.149.XXX.XXX:1028 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 72.149.XXX.XXX:1028;branch=z9hG4bKea6676234;received=72.149.XXX.XXX
From: Will at Work sip:[email protected]:5060;tag=5ac81cfd2d5773b
To: Will at Work sip:[email protected]:5060
Call-ID: [email protected]
CSeq: 1564258181 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:[email protected]
Content-Length: 0
<— Transmitting (NAT) to 72.149.XXX.XXX:1028 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 72.149.XXX.XXX:1028;branch=z9hG4bKea6676234;received=72.149.XXX.XXX
From: Will at Work sip:[email protected]:5060;tag=5ac81cfd2d5773b
To: Will at Work sip:[email protected]:5060;tag=as07b15708
Call-ID: [email protected]
CSeq: 1564258181 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="2347eec0"
Content-Length: 0
SIP/2.0 401 Unauthorized in the last message is interesting, try CLI sip show users - check the username and password.
And are that trace complete? I don’t see any ACK from the aastra. You could as well get a sip trace from the aastra as well. And to replace ip numbers are not really a good idea, put XXX on the last it you have to (you missed on spot though). As well as the username - stuff that makes debugging a lot easier.
Those are the three sets of information that go by each time the phone reboots. Username and password match. The phone also successfully accesses and uses the XML applications on the server it just won’t SIP register. Installing a syslog daemon on my work computer right now to capture the Aastra debug.
you should also have a nat=yes line…
The extension has nat=yes. I also added it to the sip_nat.conf.
Anyone have thoughts on where the 401/Unauthorized is coming from? Is that Asterisk responding or some other service? Could the firewall at work be blocking returns from Asterisk on port 1028?
I had the same problem with the aastra phones. For some reason the register would not work. Went to some other forums and they suggested removing any sip trunk that I had and it worked. Not sure why, and it kind of screwed me on my sip trunk, but it did fix the aastra problem.
My one external extension stopped registering after I upgraded to FreePBX 2.5. This was the case with both a Grandstream GXP-2000 and an Aastra 9133i.
I have the nat=yes, externip= and localnet options configured in /etc/asterisk/sip_general_custom.conf, and nothing has changed there.
Anyone else figure out the problem here???
After wasting a lot of time trying different configurations, settings, etc. I just gave up. Never could get the Aastra to register so I took it back home.