SOLVED Asterisk restarts when pjsip call comes into the server

Lee Alderman

New Member
Joined
Oct 22, 2015
Messages
9
Reaction score
0
Really confused. After about 3 days of troubleshooting and figuring out how to get inbound routes and other routes working. connecting extensions and trunks. We are able to dial out from a local phone in the new office. We were curious why asterisk wasn't registering phones at the local office. we set pjsip to port 5060, we set the phones to do the same, and the extensions accordingly. we finally got some of the phones to register with the new piaf build. the phones that have registered can dial all the way out using pjsip they hit our trunk, then get routed accordingly out of our network. trying to call them it would seem to "ring" from the callers perspective, but the physical phone doesn't ring. We had been learning a lot from watching the asterisk -r log. Mainly what we learned this time is everytime we try to call one of the pjsip phones the asterisk restarts. an error comes accross Asterisk and says Disconnected from Asterisk server Asterisk cleanly
 
In addition to this, the physical phones don't seem to associate under regular chan_sip. And they loose all ability to call out. I have tried to set the phones up to alter the ports so originally i had pjsip on port 5060, restart server, set up extension profile save config, set up physical extension via ip address. restart phone. phone was able to connect to server. but the server was rebooting during a call in to the phone, the outbound call from that phone goes all the way through. Because the phone server only reboots when we try to call pjsip i've tried to put everything on regular chan_sip. The phones are polycom501's . I changed the port on the server to 5060, altered password on both profile and phone to give the phone a reason to reboot. reboot the asterisk server as well. phone does not associate with the piaf server. Now when someone tries to call the phone it gets stuck at the users enabled voicemail instantly. (on multiple phones. ). We either need ( a way to get the physical phones to connect to the piaf server while using chan_sip, or need to figure out why the piaf server keeps rebooting when we call a pjsip device. ) help me obi-wan .... your my only hope!
 
I went through the steps and upgraded asterisk to the latest and as of right now it does seem to have helped, however, we still have not been able to get a handset to fully authorize as an endpoint. So We dont really know if the reason it's not rebooting the server is because the phone hasn't authenticated as pjsip. As of right now the softphones don't cause it to reboot. Which I'm pretty positive that they did yesterday. We were going through the asterisk logs and we've gotten rid of the username / password missmatches verified all the ports are correct and we don't see any of those random 1 off errors anymore, but the phones still dont register as pjsip clients. We can't verify that this worked until we get one of those phones to associate again (which at least one of them did associate yesterday). Though the settings looked the exact same on the rest of the phones and they weren't associating unfortunately).
 
can anyone tell me if piaf doesn't work with older sip versions? We cannot get phones to register with Asterisk... we've been trying for 3 days now.
 
Are we talking about Chan SIP connections on UDP 5061 or PJSIP connections on UDP 5060? Have you tried both? Do both fail? Have you tried using a softphone to identify whether it's a phone issue as opposed to an Asterisk problem? Please post the results of status so we all can see what your platform is.
 
Yesterday when i was able to get 1 pjsip connection working it was when we had set pjsip to utilize 5060. We had set the phone to 5060 (it already was). our phones are polycom 501's using ROM 4.1.0.0219 and are on SIP 3.1.3.0439 We have verified that the reg.1.address and reg.1.auth.userID are both the extension number We have then set the password to password11!! We have also tried the 32bit character string, and simply aa11. While making sure that it matches the extension set up the hardphone and the user link account (which is always linked to the extension number aka the user account name). We have successfully managed to call out from a sip line off of the computer using Zoiper and configuring them to use an extension created for them. The sip clients can usually pretty quick call out to anyone (which I believe means we've set up routes and trunks properly). Unfortunately we cannot call the sip extensions on the computer. I've gone ahead and installed oss extension manager and set it up. Linked the phones (which it finds really fast and easily). I've come to trust this program as it always shows RED for all ofthe phones, except twice. If it's RED icon is next to the extension then I know the phone hasn't properly registered. We have gotten 1 phone handset (polycom) to receive calls one time using PJSIP and 1 different phone to work using CHAN_SIP both times I knew to try the phone because it by chance had turned green in the oss extension manager. Both times only 1 phone had worked. and each time the perspective algorithm (chan_sip, or pjsip) was set to port 5060. I have tried enabling NAT under chan_sip, and I'm not sure if it was the original reason we were able to get chan_sip to work, because i could not get any other phones to work using that settings. and unfortunately the interface stopped interacting on the piaf so I rebooted the server, and the original chan_sip phone that was working had stopped. I tried multiple times again to get it to work, but it hasn't reconnected since. The one phone we got to work, was able to be dialed from anywhere, and dial out to anywhere and it was a polycom 501 with the same settings as above. As I said before I tried to ensure that the other phones had it's exact same settings. but none of them were able to connect. The one time we were able to get the pjsip phone to work, it was able to dial out, but not be dialed. The problem I believe is that the extensions are having a really hard time authenticating.
 
Ward I feel the original post that this was about the asterisk phone rebooting during a phone call was solved when you had me update the server so this really should be on a forum post of it's own. I know i'm not the only one thats had issues connecting polycom's to piaf. However I would like to post my phone settings and my server settings for phones and see what people say as to why it might not be working. I dont know if it would be worth my effort. But I'm going to try to do that maybe it will help some other poor schmuck including myself if i do that.
 
Please post the results of status so we all can see what your platform is.
I can't believe I just noticed this really sorry for not posting this sooner.
piaf%20settings.png
http://considertf.com/images/piaf settings.png
 

Members online

Forum statistics

Threads
26,687
Messages
174,410
Members
20,257
Latest member
Dempan
Get 3CX - Absolutely Free!

Link up your team and customers Phone System Live Chat Video Conferencing

Hosted or Self-managed. Up to 10 users free forever. No credit card. Try risk free.

3CX
A 3CX Account with that email already exists. You will be redirected to the Customer Portal to sign in or reset your password if you've forgotten it.
Back
Top