New To using Asterisk PBX,,,,Issue regarding Sip account registration


(Zeenat Ullah) #1

Dear FreePBX Community member,
Hope you all will fine and doing well. I am new to Asterisk. Asterisk 11 is installed and its working but firstly as I goes to register sip account using softphone Application. it not getting registered and giving error of 408. everything is OK password username…your kind support will be appreciated, please.


(Croissantisokay) #2

Could you go into your CLI console and Copy/Paste logs when you try a connection ?

(To log into your CLI console you just have to log into the ssh of your server and enter “asterisk -rcvvvv”)


#3

Croissantisokay’s recommendation is good to try next to find the problem. It would also help to know which softphone you’re trying, and is it on the same local network as the pbx? If it is remote, you may need do forward some ports on your network.


(Zeenat Ullah) #4


(Zeenat Ullah) #5

Dear I am getting these error complete file is too big to upload facing limitation while uploading complete file


(Zeenat Ullah) #6

I have same LAN in which asterisk is,Asterisk is getting ping from my PC and i am using Eye beam counter path softphone


(Zeenat Ullah) #7

I have the same LAN in which asterisk…Asterisk is getting ping from my PC and i am using Eyebeam counter path softphone


#8

Paste the log at https://pastebin.freepbx.org and post the link here.


(Zeenat Ullah) #9

done sir


#10

Link, please?


(Zeenat Ullah) #11

https://pastebin.freepbx.org/view/933e50e0

this is link sir


#12

Asterisk 11 is too old to support pjsip. If this is a ‘new’ system, start over with FreePBX 15 and Asterisk 16. If you have some problem with the newest versions, FreePBX 14 and/or Asterisk 13 are also ok.

If this is an existing working system, please provide details.

If this is a system that was taken out of service, do you have a problem replacing it with something current?


(Zeenat Ullah) #13

Sir sorry by mistake i write that using asterisk 11
i am using Asterisk 17.2.0…
and I think it is quite newest


#14

Sorry, I missed that you were on Asterisk 17. I don’t understand why pieces of pjsip seem to be missing. How did you build this system? If your hardware permits, I recommend using the FreePBX Distro, which has everything in a consistent state.

If that’s not possible, next choice would be an install guide from the Wiki such as
https://wiki.freepbx.org/display/FDT/How+to+Install+FreePBX+15+on+Debian+10+with+Asterisk+16%2C+PHP+7.3

If you want to continue with your present system: The log shows no activity at all, after initialization. You could run tcpdump on the PBX and see whether registration attempts are reaching the PBX. If so, try turning off the software firewall. At the Asterisk command prompt, type
pjsip set logger on
and see whether registration attempts get logged there. If not, maybe give up on pjsip and set up a chan_sip extension instead (and configure the softphone to connect to port 5160, or change the bind ports so chan_sip is on 5060).


(Zeenat Ullah) #16

Sir
output for
pjsip set logger on …is below

=========================================================================
Connected to Asterisk 17.2.0 currently running on localhost (pid = 2062)
localhost*CLI> pjsip set logger on
No such command ‘pjsip set logger on’ (type ‘core show help pjsip set logger’ for other possible commands)


#17

So pjsip is not running at all. You should either:

  1. Use the FreePBX Distro or
  2. Build FreePBX from a well-tested script (FreePBX 15 or 14, Asterisk 16 or 13) or
  3. Find out why your build isn’t working or
  4. Try to get your box working with chan_sip extensions and trunks.

The tcpdump output was useless because it was just reporting the SSH traffic (that it was causing). Workarounds for this include writing the output to a file, attempting registration, stopping the capture and then viewing the file. Or, set a capture filter that looks only at UDP. Or, use the console (if your machine has one) instead of SSH. However, none of this is relevant until you have a working build.


(Zeenat Ullah) #18

localhost*CLI> sip reload
Reloading SIP
== SIP Listening on 0.0.0.0:5160
== Using SIP CoS mark 4
Dear Stewart, now i installed Asterisk 13.31.0
configured port 5160…but still not getting registerred sip soft phone??
Need you help in this plz.


#19

What did you change, and why?

I know nothing about Eyebeam but am guessing that it’s similar to X-Lite. If so, settings for the account should be:

Display Name: (as desired)
User name: same as your chan_sip extension number in FreePBX.
Password: same as the Secret for your extension.
Authorization user name: same as your chan_sip extension number in FreePBX.
Domain: 192.168.1.123:5160
(replace 192.168.1.123 with the local IP address of your Asterisk.)

Domain Proxy:
Register with domain …: checked.
Send outbound via: domain

If your settings don’t look like that, post a screenshot of the Eyebeam account settings.

If no luck, at the Asterisk command prompt, type
sip set debug on
then restart Eyebeam. Report what gets logged on the console. If nothing, use tcpdump to see whether registration requests are hitting the PBX.


(Zeenat Ullah) #20

My issue is resolved dear Stewart,
it was due to firewall of asterisk
thanks again.


(system) closed #21

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