Dial box not working
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You have to check your asterisk logs. Increase verbose and debug levels in the asterisk cli and check the full log when trying ( core set verbose10; core set debug 10;)
FOP2 is doing its job. It sends the correct command, is asterisk or your phone that is not accepting the origination.
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Now I originate a call by clicking on the dial box. This works - the call is originated successfully. Below you can see a portion of the full log
[Apr 8 12:22:09] DEBUG[29455] manager.c: Manager received command 'Originate'
[Apr 8 12:22:09] DEBUG[11496] chan_sip.c: Asked to create a SIP channel with formats: 0x40 (slin)
[Apr 8 12:22:09] VERBOSE[11496] netsock.c: == Using SIP RTP TOS bits 184
[Apr 8 12:22:09] VERBOSE[11496] netsock.c: == Using SIP RTP CoS mark 5
[Apr 8 12:22:09] VERBOSE[11496] netsock.c: == Using UDPTL TOS bits 184
[Apr 8 12:22:09] VERBOSE[11496] netsock.c: == Using UDPTL CoS mark 5
[Apr 8 12:22:09] DEBUG[11496] chan_sip.c: Allocating new SIP dialog for <!-- e --><a href="mailto:183e2ef558e7ac96476632d574f6e545@10.0.30.10">183e2ef558e7ac96476632d574f6e545@10.0.30.10</a><!-- e --> - INVITE (With RTP)
[Apr 8 12:22:09] DEBUG[11496] chan_sip.c: Setting NAT on RTP to On
[Apr 8 12:22:09] DEBUG[11496] chan_sip.c: Setting NAT on UDPTL to On
[Apr 8 12:22:09] DEBUG[11496] acl.c: Found IP address for this socket
[Apr 8 12:22:09] DEBUG[11496] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.0.30.10:5060
[Apr 8 12:22:09] DEBUG[11496] chan_sip.c: *** Our native formats are 0x4 (ulaw)
[Apr 8 12:22:09] DEBUG[11496] chan_sip.c: *** Joint capabilities are 0x0 (nothing)
[Apr 8 12:22:09] DEBUG[11496] chan_sip.c: *** Our capabilities are 0x4 (ulaw)
[Apr 8 12:22:09] DEBUG[11496] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw)
[Apr 8 12:22:09] DEBUG[11496] chan_sip.c: *** Our preferred formats from the incoming channel are 0x40 (slin)
[Apr 8 12:22:09] DEBUG[11496] chan_sip.c: This channel will not be able to handle video.
[Apr 8 12:22:09] DEBUG[11496] chan_sip.c: Outgoing Call for 14502
[Apr 8 12:22:09] DEBUG[11496] chan_sip.c: Updating call counter for outgoing call
[Apr 8 12:22:09] DEBUG[9984] devicestate.c: No provider found, checking channel drivers for SIP - 14502
[Apr 8 12:22:09] DEBUG[9984] chan_sip.c: Checking device state for peer 14502
[Apr 8 12:22:09] DEBUG[9984] devicestate.c: Changing state for SIP/14502 - state 6 (Ringing)Now, if I type the extension into the dialbox and hit ENTER, I cannot find anything in the full log relating to this request - no mention of and 'Originate' command or anything like that. The only output I see in the logs available to me are the dial msg tag being sent by the web client (seen in firebug), and the msg being received by fop2_server when running it at debuglevel 15.
So FOP2 succeeds in originating the call when the Dial button is clicked, but is unable to originate a call when the dialbox is used.
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I do not know your system, your configurations, your sip peers, your devices. So I cannot help you out, the full debug log will make no sense without that knowledge.
Look at the originate command that you first posted, does it seem ok? It does to me, but I am not sure if the originate channel is correct. As you see , the originate command was sent, and asterisk accepted it ok, after that all is done in asterisk-phone, not in fop2.
Enable sip debug, see if your phone is rejecting calls, look at the originate ami command and see if its correct or not. Do it with dial or typing and compare them. It is a task you have to do on YOUR box. It is not a general issue.
Best regards,
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Yes, the originate command is sent when the Dial button is clicked, but no originate command is sent by fop2_server to asterisk when the dialbox is used.
The originate command I first posted is a result of clicking on the Dial button, when the dialbox is used, nothing is sent to the asterisk server.
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Yes, we are running in device/user mode and a registered copy of FOP2 version 2.20 final.
I think we are getting somewhere now. This is the output from debuglevel 511:
** MAIN AMI event received...
** MAIN Processing command received from flash clients...192.168.180.106 <= <msg data="6|dial|4007|c4dbe7308d413f2c15365e7470249bde" />
-- PROCESS_FLASH_COMMAND origen 6 accion dial destino 4007
-- PROCESS_FLASH_COMMAND password c4dbe7308d413f2c15365e7470249bde
VALIDAR USUARIO 4006
VALIDAR USUARIO 4006 OK con clave regular (192.168.180.106)
Validation ok, have dial permissions
Not a reference at all
DIAL failed because origin channel SIP/14502 is not blessed
DIAL
The above is when signing into FOP2 as extension 4006, which is fixed to device id 14502.
Interestingly, if I try the same thing when logged into FOP as an extension that is not fixed to a device, but logged in to an adhoc device, it works:
** MAIN AMI event received...
** MAIN Processing command received from flash clients...192.168.180.106 <= <msg data="8|dial|4007|9bdefc597220a3c8710d6be15ac03809" />
-- PROCESS_FLASH_COMMAND origen 8 accion dial destino 4007
-- PROCESS_FLASH_COMMAND password 9bdefc597220a3c8710d6be15ac03809
VALIDAR USUARIO 4008
VALIDAR USUARIO 4008 OK con clave regular (192.168.180.106)
Validation ok, have dial permissions
It's blessed into class Extension
DIAL
Action: Originate
Channel: SIP/14501
Exten: 4007
Context: from-internal
Priority: 1 -
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Sure..
[root@pbx1 fop2]# md5sum fop2_server
b20267e789e235d02267b33c3e43f5fe fop2_server
[root@pbx1 fop2]# ./fop2_server -v
fop2_server version 2.20 -
Hmm, the fixed/adhoc device correlation may have just been coincidental - the dialbox is not working for all extensions now - fixed or adhoc.
Same error as before:
DIAL failed because origin channel SIP/14501 is not blessed
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We have a dedicated user defined with read=all, write=all.
I also checked fop2.cfg for an event_mask line - there is one, but it is commented out.
I tried restarting everything (fop2, asterisk, ctrl-F5 in browser) to make sure that nothing had got in a weird state, but it didn't help.
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I will have to set a system in user & device mode for testing this and see if I can reproduce the issue, or you can try to catch me on the live help and provide remote access to your server so we can work on there directly.
In the meantime, there is probably a way to fix this problem using the originatechannel configuration parameter. If you have the fop2admin module installed, go to Tools - FOP2 Buttons, and set the originatechannel of one of your users to Local/XXXX@from-internal where XXXX is the user/extension number. Then try again.
Best regards,
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Thanks Nicolás.
I tried setting the originate line, but it did not work. The error message is essentially the same:
DIAL failed because origin channel LOCAL/4006@FROM-INTERNAL is not blessed
Unfortunately I cannot provide you with remote access. As a developer, I understand how difficult it can be to diagnose a problem without access to the system with said problem. The system I am working on will become our live server soon, so has a lot of user information.
If there is anything I can send you directly - database dumps, config files, etc, I'd be happy to provide them. I can also try anything else you might suggest.
Thanks for your continued help.
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Hi,
Im also experiencing this with the latest version. Im using asterisk 1.4.22.
I just downloaded and purchased the basic version.When i hit enter on the Dial textbox, it does not react nor any debug messages to the asterisk cli is popping out
Appreciate any assistance on this matter
Thanks
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