We have used jmar71n/FreePBX docker and deployed a FreePBX server, but we are not able to set up bandwidth sip and not able to make or receive a call.We are getting the above mentioned error while an outbound call from Zoiper(softphone). Thank you in advance.
Kindly help this is very critical for us
I might be wrong, but I’m pretty sure that error is not related with your issue.
Post a complete log of a failed call attempt so we can help you.
DTMF[2463][C-0000000e]: channel.c:4030 __ast_read: DTMF begin ‘1’ received on SIP/2000-0000000c
[2018-04-04 12:39:21] DTMF[2463][C-0000000e]: channel.c:4034 __ast_read: DTMF begin ignored ‘1’ on SIP/2000-0000000c
[2018-04-04 12:39:22] DTMF[2463][C-0000000e]: channel.c:3944 __ast_read: DTMF end ‘1’ received on SIP/2000-0000000c, duration 120 ms
[2018-04-04 12:39:22] DTMF[2463][C-0000000e]: channel.c:4014 __ast_read: DTMF end passthrough ‘1’ on SIP/2000-0000000c
[2018-04-04 12:39:22] DTMF[2463][C-0000000e]: channel.c:4030 __ast_read: DTMF begin ‘2’ received on SIP/2000-0000000c
[2018-04-04 12:39:22] DTMF[2463][C-0000000e]: channel.c:4034 __ast_read: DTMF begin ignored ‘2’ on SIP/2000-0000000c
[2018-04-04 12:39:22] DTMF[2463][C-0000000e]: channel.c:3944 __ast_read: DTMF end ‘2’ received on SIP/2000-0000000c, duration 120 ms
[2018-04-04 12:39:22] DTMF[2463][C-0000000e]: channel.c:4014 __ast_read: DTMF end passthrough ‘2’ on SIP/2000-0000000c
[2018-04-04 12:39:22] DTMF[2463][C-0000000e]: channel.c:4030 __ast_read: DTMF begin ‘3’ received on SIP/2000-0000000c
[2018-04-04 12:39:22] DTMF[2463][C-0000000e]: channel.c:4034 __ast_read: DTMF begin ignored ‘3’ on SIP/2000-0000000c
[2018-04-04 12:39:22] DTMF[2463][C-0000000e]: channel.c:3944 __ast_read: DTMF end ‘3’ received on SIP/2000-0000000c, duration 120 ms
[2018-04-04 12:39:22] DTMF[2463][C-0000000e]: channel.c:4014 __ast_read: DTMF end passthrough ‘3’ on SIP/2000-0000000c
[2018-04-04 12:39:22] DTMF[2463][C-0000000e]: channel.c:4030 __ast_read: DTMF begin ‘#’ received on SIP/2000-0000000c
[2018-04-04 12:39:22] DTMF[2463][C-0000000e]: channel.c:4034 __ast_read: DTMF begin ignored ‘#’ on SIP/2000-0000000c
[2018-04-04 12:39:22] DTMF[2463][C-0000000e]: channel.c:3944 __ast_read: DTMF end ‘#’ received on SIP/2000-0000000c, duration 120 ms
[2018-04-04 12:39:22] DTMF[2463][C-0000000e]: channel.c:4014 __ast_read: DTMF end passthrough ‘#’ on SIP/2000-0000000c
[2018-04-04 12:39:23] WARNING[460]: func_cdr.c:377 cdr_write_callback: CDR requires a value (CDR(variable)=value)
)[2018-04-04 12:39:23] WARNING[460]: func_cdr.c:377 cdr_write_callback: CDR requires a value (CDR(variable)=value)
)[2018-04-04 12:39:24] NOTICE[488][C-0000000e]: chan_sip.c:23808 handle_response_invite: Failed to authenticate on INVITE to ‘sip:[email protected];tag=as21c38503’
[2018-04-04 12:39:24] NOTICE[488][C-0000000e]: chan_sip.c:23808 handle_response_invite: Failed to authenticate on INVITE to ‘sip:[email protected];tag=as21c38503’
[2018-04-04 12:39:24] WARNING[2463][C-0000000e]: app_macro.c:310 _macro_exec: No such context ‘macro-outisbusy’ for macro ‘outisbusy’. Was called by 7042566755@from-internal
[2018-04-04 12:39:24] WARNING[2463][C-0000000e]: app_macro.c:310 _macro_exec: No such context ‘macro-outisbusy’ for macro ‘outisbusy’. Was called by 7042566755@from-internal
That is not the complete log of the call, but anyway, this is probably why your call is failing:
what could be the reason ? username, secret is given correctly still it is saying ‘Failed to authenticate’
Re-check the username and password. If you are using the default auto-generated password, it can sometimes be too large, set it to a shorter password.
The account(asterisk extension) is registered in softphone
When I call someone, first it asks the outbound route password and says thank you for entering the correct password, later it gets disconnected.
Then the above logs appear.
Why don’t you first try an outbound route without a password?
I tried, it got disconnected as soon as I called a number
[image removed for sensitive info, time to change your trunk sip secret! - mod]
can you help me out with this
what should be in incoming and what should be in outgoing?
and how to connect an extension with the trunk?
I don’t see any context(from-internal,from-phones etc) here.I am totally confused
I hope you don’t feel offended, but I think that at this point, you need to go to the wiki and read the manual, it is explained there how to create an extension and how to create a trunk. Once you are done with that, come back with a specific description of your issues and provide a full log of the failed call, not just an extract. To do that, log into asterisk with this command: asterisk -R -vvv and try to make a call, and copy the full output that you get from the moment you try the call.
Also, take down the picture with your trunk details, you haven’t greyed out your username and password, that means anyone can see them and use your trunk.
Thanks will try!!
This topic was automatically closed 365 days after the last reply. New replies are no longer allowed.
