Hi All, i have the problem with connect my asterisk with OLD switch my client.
I cannot use PJSIP but only SIP.
When i send call the test for check the connection i have always NOTICE chan_sip.c handle_response_invite: Failed to authenticate on INVITE
This is my configuration
sip.conf
You mean server not client. Clients issue INVITE, and servers respond to them.
Please identify the endpoint and explain the technical reason why it is incompatible with chan_pjsip.
Your secret is wrong, or you are not providing the identification information required by the server user agent. Typically the identification information is either the user part of From: header URI, which in this case will be whatever the calling client user agent sent (probably 200, but there is insufficient information to be certain), or the IP address of Asterisk.
type=friend is wrong, although often included in copy and paste configurations.
Having both in the same context is questionable.
What is there about your network configuration that requires force_rport?
What does “quality” mean? There is no such option.
Hi David
If i missing type i have this answer
WARNING[4297]: chan_sip.c:33634 reload_config: Section ‘IP01’ lacks type
If i add the new SIP for softphone is all ok, if i call from 200 to 300 is ok.
But when i call to old switch i have the problem.
I rechack the parameter and are all ok.
== Using SIP RTP CoS mark 5
> 0x7feb740521e0 -- Strict RTP learning after remote address set to: 192.168.1.10:5062
-- Executing [45622222@tohost01:1] NoOp("SIP/200-00000054", "") in new stack
-- Executing [45622222@tohost01:2] Playback("SIP/200-00000054", "test") in new stack
-- <SIP/200-00000054> Playing 'test.gsm' (language 'en')
> 0x7feb740521e0 -- Strict RTP qualifying stream type: audio
> 0x7feb740521e0 -- Strict RTP switching source address to Y.Y.Y.Y:5062
-- Executing [45622222@tohost01:3] Dial("SIP/200-00000054", "SIP/45622222@IP01") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/45622222@IP01
[2023-12-30 17:57:35] NOTICE[4297][C-0000005e]: chan_sip.c:24413 handle_response_invite: Failed to authenticate on INVITE to '"200" <sip:200@X.X.X.X>;tag=as5ce0951d'
-- SIP/IP01-00000055 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [45622222@outcall:4] Hangup("SIP/200-00000054", "") in new stack
== Spawn extension (outcall, 45622222, 4) exited non-zero on 'SIP/200-00000054'
and if i check the sip status
Host dnsmgr Username Refresh State Reg.Time
A.A.A.A:5060 N IP01 3585 Registered Sat, 30 Dec 2023 17:57:12
and peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
200/200 Y.Y.Y.Y D Yes No 5060 OK (26 ms)
IP01 A.A.A.A Yes No 5060 OK (36 ms)
2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline]
This is 456, not something that matches 456X. It is two digits too short.
It’s asking for the password. That’s why you are getting a failure. I suppose it might be other reasons for 403. You need to use “sip set debug on” to see what the actual response is.
I don’t understand why you are saying it doesn’t work with chan_pjsip, when you are demonstrating that it doesn’t work with chan_sip.