I have a user who is working remote and needs all calls dialing her extension to immediately forward to her external cell phone. Sorry to be such a newbie here but all the searches I do end up with very complicated instructions and such. Here is her entry in sip.additional.conf. I just need all calls to her extension to forward to her cell phone number (xxx) xxx-xxxx.
chan_sip is obsolete, unsupported and completely missing from Asterisk 21, whose release is imminent.
There is no documented dial= option, and I wouldn’t expect one.
Except for the context= setting, the incoming channel configuration is irrelevant, and context tends to work the same way for all drivers. The relevant file here is extensions.conf (or one of its functional equivalents).
Thanks David. I edited the line
dial=SIP/xxxxxxxxxx ; her cell phone number
Restarted asterisk and all is well.
Probably should have included that we’re using an older version of Asterisk. The entire PBX system is soon to be replaced so more to come.
I’ve checked the source of 1.6.2, 11, and 20.2, and none of then recognize a dial= option. I think you may be using a locally patched version.
I get this when I’m checking from the Elastix GUI with core show version:
Asterisk 11.13.0 built by palosanto @ rpmbuild32-2.elastix.palosanto.com on a i686 running Linux on 2014-10-02 17:41:51 UTC
That is so ancient that github doesn’t have the individual versions, but neither 11, nor 12, will recognize any option beginning with “dial”. It’s almost certainly part of Elastix, rather than Asterisk. Either they are using patched chan_sip.c, or they have code that interprets the configuration file during initialisation.
It looks like the Elastix web site is dead, so I guess it is abandonware.
dial ist Elastix specific. And Elastix was bought by 3CX late 2016.
Good morning all and thanks again for the updates and news. We are soon to replace this aged Elastix server with a more modern option. But for the next few weeks we’re stuck with what we have. And in it’s defense I can say that it has performed almost flawlessly with a very high uptime.
Can someone convince me I’m not crazy. On two different occasions I did the above edits in the sip_additional.conf file. Both times I tested the change by dialing the user’s internal extension #6338 and each time the call was forwarded to the user’s cell phone. Okay, that’s cool. But a day or so later it stopped working and I checked the sip_additional.conf and the changes I had made and saved to the file were gone. I edited the file again and just now, less than one week later I rechecked the file and sure enough the edits I made were gone.
Is this overwriting of edited files a feature of Asterisk?
No. That would be your GUI overwriting the file.
“Is this overwriting of edited files a feature of Asterisk?”
No, it is not.
I would ask “where did this file ‘sip_additional.conf’ come from and what is
causing it to be included in what Asterisk reads and parses as 'sip.conf?”
Is this a pure Asterisk system, or are you using something like FreePBX,
Elastix, Issabel, etc?
Yes, it’s an old Elastix v2.4.0-19. There is no doubt, I can edit the file in the Elastix gui and then reload Asterisk and all is well. I can even open up the file in the gui again and see the changes as well as using putty.exe to ssh into the Linux server and then cat the sip_additional.conf file and the changes are there. But sooner or later I look into it and the changes are gone and the fowarding I needed for the users is no longer working. BUMMER!
I believe that FreePBX and its children create that file from their database every time that they are started or a configuration reload is forced. I believe the files that users are allowed to change generally have custom in their names.
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