The external_media_address option does not currently support a dynamic IP address, and while a hostname is allowed it is not guaranteed that the remote side will tolerate it.
Is there any to provide an IP? dnsmgt resolves the name. Can it be used to generate a variable with the value of the current outside IP?
The only way currently is to provide it to the option. You can’t use a variable. The issue tracking support for this is ASTERISK-26458[1]. You can add yourself as a watcher if you have an account to be notified of any updates to the issue.
Thanks a lot for your answer.
It seems to work with direct_media = no as I was able to place a call and the audio was bidirectional.
BUT I think I’m going crazy…
If there’s no one answering the call it should go to the voicebox. The message is played but there is no hint in Asterisk CLI that the file is played but only a
Spawn extension (Telekom_in, 01234567891, 1) exited non-zero on ‘PJSIP/Telekom_in-00000008’
The called party just seems to hangup…
The full console output and dialplan would be needed.
Your wish is my command…
-- Executing [01234567890@Telekom_in:1] Dial("PJSIP/Telekom_in-00000008", "PJSIP/22&SCCP/11,30") in new stack > SCCP: sccp_requestChannel returned Line 11 not currently registered -> Try again later[Feb 8 13:14:28] WARNING[2723][C-00000005]: app_dial.c:2530 dial_exec_full: Unable to create channel of type ‘SCCP’ (cause 44 - Requested channel not available)
– Called PJSIP/22
– PJSIP/22-00000009 is ringing
– PJSIP/22-00000009 answered PJSIP/Telekom_in-00000008
– Channel PJSIP/22-00000009 joined ‘simple_bridge’ basic-bridge <017e57f0-1083-465d-bb35-d4139b2b789e>
– Channel PJSIP/Telekom_in-00000008 joined ‘simple_bridge’ basic-bridge <017e57f0-1083-465d-bb35-d4139b2b789e>
> 0x76419660 – Probation passed - setting RTP source address to 192.168.178.57:5004
– Channel PJSIP/22-00000009 left ‘simple_bridge’ basic-bridge <017e57f0-1083-465d-bb35-d4139b2b789e>
– Channel PJSIP/Telekom_in-00000008 left ‘simple_bridge’ basic-bridge <017e57f0-1083-465d-bb35-d4139b2b789e>
== Spawn extension (Telekom_in, 01234567890, 1) exited non-zero on ‘PJSIP/Telekom_in-00000008’
Dialplan:
exten => 01234567890,1,Dial(PJSIP/22&SCCP/11,30)
exten => 01234567890,n,VoiceMail(11@voicemail)
exten => 01234567890,n,Hangup()
The endpoint PJSIP/22 was called and it answered. Thus voicemail was not executed.
No, I heard the voicemail message but as you noticed there’s no hint in the CLI
I wrote that in one of my posts before…
Okay, voicemail message from what? The Voicemail application isn’t being executed in Asterisk on the log you posted, so it’s not from that. What is PJSIP/22?
Or better yet, if you remove PJSIP/22 from your Dial line do things behave as you expect? If so then the problem is with PJSIP/22
I have a suspicion what could cause that behaviour.
I have to check with my provider and I’´LL come back to you when I have more informations…
It looks that the provider switched on a voicemail which we disabled…
Thanks for pointing into this direction!
The problem with PJSIP/22 is solved.
I changed the plan with my provider and he re-activated the provider voicemail.
But I observed another voicemail/sound related phenomenon.
The voicemail soundfiles are played (as seen in the CLI) but there is no sound on the caller’s phone only the ringing tone on the caller’s phone.
Sometimes when I pick up a call no sound is heard.
This is what I see in CLI
– Executing [01234567890@Telekom_in:1] Dial(“PJSIP/Telekom_in-00000004”, “PJSIP/22&SCCP/11,30”) in new stack
– Called PJSIP/22
– SCCP: Asterisk request to call SCCP/11-00000011 (dest:11, timeout: 0)
– 11: Asterisk request to call SCCP/11-00000011
– SCCP/11-00000011: (sccp_pbx_call) Returning: 0
– Called SCCP/11
– SCCP/11-00000011 is ringing
– PJSIP/22-00000005 connected line has changed. Saving it until answer for PJSIP/Telekom_in-00000004
– PJSIP/22-00000005 is ringing
– Nobody picked up in 30000 ms
– Executing [01234567890@Telekom_in:2] Wait(“PJSIP/Telekom_in-00000004”, “1”) in new stack
– Executing [01234567890@Telekom_in:3] VoiceMail(“PJSIP/Telekom_in-00000004”, “11@voicemail”) in new stack
– <PJSIP/Telekom_in-00000004> Playing ‘vm-intro.alaw’ (language ‘de’)
> 0x75f24438 – Probation passed - setting RTP source address to 217.0.4.199:47014
– <PJSIP/Telekom_in-00000004> Playing ‘beep.alaw’ (language ‘de’)
– Recording the message
– x=0, open writing: /var/spool/asterisk/voicemail/voicemail/11/tmp/7OG8hy format: wav, 0x75a058fc
– User hung up
== Parsing ‘/var/spool/asterisk/voicemail/voicemail/11/INBOX/msg0000.txt’: Found
== Parsing ‘/var/spool/asterisk/voicemail/voicemail/11/INBOX/msg0000.txt’: Found
== Parsing ‘/var/spool/asterisk/voicemail/voicemail/11/INBOX/msg0000.txt’: Found
== Spawn extension (Telekom_in, 01234567890, 3) exited non-zero on ‘PJSIP/Telekom_in-00000004’
Do you need a debug log?
That would help to confirm that Asterisk is doing the right thing, yes.
And here we go…
What version of Asterisk is this? There appears to be a codec issue which might be solved in the latest release.
It’s Asterisk 14.2.1 on a armv7l running Linux on 2017-02-04 20:23:31 UTC (Raspberry Pi 2)
Is it possible to do an in situ upgrade or do I have to reinstall everything?
I have no idea how you’ve installed on that platform or what is involved, so I can’t answer that. Building the new Asterisk with the same config and installing it is what most people do.
I just compiled from the sources.
If I compile the 14.3 version and do a “make install” will the config files be overwritten?
No. The “make install” does not overwrite config files.
Installed 14.3 and first calls don’t show this behaviour. Will check this evening and report back tomorrow.
Thanks!