Hi
I have the following problem with asterisk 20/PJSIP
my server answer to the 180 Rining with 200 OK (on answer)
but I never receive an ACK, then after15 retries (30 seconds) the communication is closed
Any idéa? Suggestions?
qAsterisk 20 and PJSIP
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Liptonbrisk
- Technical Support
- Posts: 2773
- Joined: 04/26/2010
- SIP Device Name: Obihai 202/2182, Groundwire
- Firmware Version: various
- ISP Name: FTTH
- Computer OS: Windows 64 bit
- Router: Asuswrt-Merlin & others
Re: qAsterisk 20 and PJSIP
I'm not using Asterisk at this time, but lack of ACK with Freephoneline is often due to having SIP ALG/SIP Passthrough enabled in a router or the ISP's modem/router combo, gateway, or hub; not registering the trunk; or a NAT issue (NAT hole closing or association becoming corrupted).
So, that's steps 1, 2, 5 (check "SIP status" after logging in at https://www.freephoneline.ca/showSipSettings), and B from viewtopic.php?f=8&t=20199#p78976 (D from that link is likely not related, but you may want to look at it anyway).
If you have an AOR section, you may need to use qualify_frequency=20 to keep NAT association alive.
Expiration needs to be 3600 seconds. Using less than that can block the service from working.
viewtopic.php?f=8&t=20533
https://support.freephoneline.ca/hc/en- ... redentials (read that)
Beyond that, you're probably going to require help from someone who is actually using Asterisk.
I haven't tested Asterisk (was with an old version) in years, and when I did, it was before Freephoneline enforced the 3600 registration interval (and before their switch vendor implemented a 15 minute session timer).
You can also try asking at
https://community.asterisk.org/
https://www.reddit.com/r/Asterisk
https://www.dslreports.com/forum/voip
I would suggesting posting your settings and a SIP trace/log if you ask elsewhere.
So, that's steps 1, 2, 5 (check "SIP status" after logging in at https://www.freephoneline.ca/showSipSettings), and B from viewtopic.php?f=8&t=20199#p78976 (D from that link is likely not related, but you may want to look at it anyway).
If you have an AOR section, you may need to use qualify_frequency=20 to keep NAT association alive.
Expiration needs to be 3600 seconds. Using less than that can block the service from working.
viewtopic.php?f=8&t=20533
https://support.freephoneline.ca/hc/en- ... redentials (read that)
Beyond that, you're probably going to require help from someone who is actually using Asterisk.
I haven't tested Asterisk (was with an old version) in years, and when I did, it was before Freephoneline enforced the 3600 registration interval (and before their switch vendor implemented a 15 minute session timer).
You can also try asking at
https://community.asterisk.org/
https://www.reddit.com/r/Asterisk
https://www.dslreports.com/forum/voip
I would suggesting posting your settings and a SIP trace/log if you ask elsewhere.
Please do not send me emails; I do not work for nor represent Freephoneline or Fongo. Post questions on the forums so that others may learn from responses or assist you. Thank you. If you have an issue with your account or have a billing issue, submit a ticket here: https://support.fongo.com/hc/en-us/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at http://forum.fongo.com/viewforum.php?f=15.