Hello,
I am trying to switch latest centos and asterisk 18.4 , but issue to setup Sip Pri with pjsip.
Please guide. username - +9111412123XX@ims.airtel.in
PRI normally means Primary Rate Interface, which is fundamentally incompatible with VoIP, and only applies to circuit switched connections.
These are almost certainly wrong. They are workarounds for when the endpoint is inside NAT, but doesn’t compensate for it, and asterisk is outside. You have no configuration for asterisk inside NAT, so I assume both you and the ITSP are on public addresses. (I think the same applies to Rewrite Contact, but I’m not so sure on that.)
Are you sure the username is not just +9111412123XX? @'s in URI users get messy, and do you really need an authentication user?
You are missing a transport section.
You are going to need the outbound authentication un-commenting.
I’d be surprised if you didn’t need the From user. Including @domain, in that, is inconsistent with your client URI in the registration.
The basic framework for ITSPs as end points is given in Home - Asterisk Documentation but, as it says, details may vary, and only another user of your ITSP is likely to know the exact details.
What error messages do you get? If the REGISTER is sent, what does pjsip set logger on show as the response?
following command belong to my old version asterisk 1.8, where it is working fine.
register => +911141212300:xxxxxxx:+911141212300@ims.airtel.in@ims.airtel.in/+911141212300
[transport-udp-nat]
type = transport
protocol = udp
bind = 0.0.0.0
However, if they are doing silly things like including “@” in SIP userinfo fields, you should ask them what they actually want to see over the wire, rather than what has to be entered into your device. If they really say they need two @'s, there implementation is broken.
Did from user really work without the @ in chan_sip? Do they need from user at all?
Hello David, thanks for reply, I already implemented/tried same but result is same, not registered.
client_uri=sip:+911141212300%40ims.airtel.in@ims.airtel.in
[Jun 7 18:40:46] WARNING[2766]: res_pjsip_outbound_registration.c:894 schedule_retry: No response received from ‘sip:ims.airtel.in:5060’ on registration attempt to ‘sip:+911141212300%40ims.airtel.in@ims.airtel.in:5060’, retrying in ‘30’
[Jun 7 18:42:20] WARNING[2766]: res_pjsip_outbound_registration.c:894 schedule_retry: No response received from ‘sip:ims.airtel.in:5060’ on registration attempt to ‘sip:+911141212300%40ims.airtel.in@ims.airtel.in:5060’, retrying in ‘30’
This sounds like a connectivity problem (e.g. NAT or firewall), rather than an authentication one. I notice that your transport section mentions nat in its name but has no information that would allow it to work properly from behind NAT.
These are not public addresses. You are going to need to tell us a lot more about the structure of your network, as some ITSPs do use a dedicated private network for VoIP services, but we need to know more to know whether these can be treated as extensions of your private LAN, are better treated as public, or you have a fundamental conflict between real public addresses and pseudo ones.
Also, please use the forum </>, pre-formatted text, button, to prevent parts of your logs, etc., being treated as mark-up.
PS. There is no logging of a registration failure, so it may not have tried a re-registration, yet.
Dear David, Is that possible for you to spare 5 minutes to access my system through remote.
One more thing that in this unregistered situation, inbound calls are coming but disconnecting in few seconds.
this is my network provided by service provider - Airtel
P Address 10.204.55.22
Network Address: 10.204.55.20 / 30
gateway - 10.232.130.170
Can you pls guide to implement/restore Sip in asterisk 18.4 and make disabled PjSip to start using my Pri line functioning as working now fine with old version.
thanks