I have a base with one ip that works with asterisk server, this base have more than one user correctly registred and sometimes, the portable phone result busy
why ?
thank you
Let me propose a better description 
I have a Gigaset base with multiple SIP accounts configured. This base have more than one handset connected and each handset is assigned it’s own “connection” in Gigaset terms, so each handset has it’s own extension on Asterisk.
Sometimes Gigaset returns 486 Busy to Asterisk.
Everything is correct?
First thing to check - the number of simultaneous VoIP calls this particular Gigaset model supports, then the configured limit for simultaneous calls (somewhere in Advanced settings).
oh perfect thank you man, I think the simultaneous voip connection on gigaset setting is ok, but I don’t know how to check on asterisk, or better-- I don’t know what is going wrong
why the handset don’t call at any time ?
Please clarify on the call direction - are you placing a call from Gigaset or you’re sending a call from your Asterisk to Gigaset?
In either case the first step will be checking the configuration on both Gigaset and Asterisk then sip debug needs to be checked on Asterisk during the call attempt.
I have an asterisk server where sip is configure and even extension.conf; the 4 hanset for each base are correctly registred and configured ( I mean the handset registred on server asterisk, gigaset base allow more than 1 voip call, EACH HANDSET RECEIVE CALL FROM OUTSIDE OR OTHER NUMBER INSIDE THE SOCIETY ) but some handset don’t call, neither inside neither outside. What’s going wrong ? and what is sip debug ?
For SIP debug see: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
Without that information one cannot start on trying to answer your main question.
sorry but i need more help, I don’t understand what I have to do to obtain the debug file
? ? ? help !!!
There are various ways in Asterisk of obtaining different bits of debugging. The wiki page that @david551 linked gives you instructions.
I’m not so expert like you and so I don’t understand that wiki. Do u have a easy one please ?step by step
[Dec 6 10:21:46] WARNING[11232][C-0000000b]: chan_sip.c:16380 check_auth: username mismatch, have <1041>, digest has <1031>
[Dec 6 10:21:46] NOTICE[11232][C-0000000b]: chan_sip.c:25535 handle_request_invite: Failed to authenticate device “Lorenzo” sip:1031@192.168.2.1;tag=1258382285
[Dec 6 10:22:55] WARNING[11232][C-0000000c]: chan_sip.c:10088 process_sdp: Ignoring video stream offer because port number is zero
The first two messages are received when 2 number configured and I receive error 403. the third messagge it’s ok, I call
[Dec 6 10:27:39] NOTICE[11232]: chan_sip.c:23561 handle_response_peerpoke: Peer ‘1041’ is now Reachable. (38ms / 2000ms)
[Dec 6 10:28:43] WARNING[11232][C-0000000d]: chan_sip.c:10088 process_sdp: Ignoring video stream offer because port number is zero
[Dec 6 10:28:54] WARNING[11232][C-0000000e]: chan_sip.c:16380 check_auth: username mismatch, have <1041>, digest has <1031>
[Dec 6 10:28:54] NOTICE[11232][C-0000000e]: chan_sip.c:25535 handle_request_invite: Failed to authenticate device “Lorenzo” sip:1031@192.168.2.1;tag=3301536097
[Dec 6 10:29:24] WARNING[11232][C-0000000f]: chan_sip.c:16380 check_auth: username mismatch, have <1041>, digest has <1031>
[Dec 6 10:29:24] NOTICE[11232][C-0000000f]: chan_sip.c:25535 handle_request_invite: Failed to authenticate device “Lorenzo” sip:1031@192.168.2.1;tag=3921250829
What’s your evidence for this not being correct?
[Dec 6 10:21:46] WARNING[11232][C-0000000b]: chan_sip.c:16380 check_auth: username mismatch, have <1041>, digest has <1031>
[Dec 6 10:21:46] NOTICE[11232][C-0000000b]: chan_sip.c:25535 handle_request_invite: Failed to authenticate device “Lorenzo” ;tag=1258382285
both number (1031-1041) are correctly registred on gigaset and asterisk, but when i call from 1041 the debug say this (device lorenzo is 1031 device carlo is 1041)
how is ? it’s all strange
You probably need to use type=user.
In any case, you need to provide use with your sip.conf.
I try and let u know
sorry i need more help cause i don’t undestand;
i have set my external phone number like friend and my internal number are friend and peer
what you suggest me ?
Remove the peer entries. The peer entries will match on the IP address, but you cannot distinguish on IP address.
However, I thought Asterisk tried a user match before a peer one, so I’m not really sure I fully understand this.
As the device name doesn’t seem to be the same as the authentication user, do you have these correctly paired in the device and do you have distinct device names?
THANK YOU A LOT
I was missing a sample entries … TYPE=FRIEND
I was using the wrong one