I thought to something like:
[...]
or
[...]
Or make the script place a call file [0].
-nik
[0]: http://www.voip-info.org/wiki/view/Asterisk+Call+Files
--
burny Ein Jabber-Account, sie alle zu finden; ins Dunkel zu treiben
und ewig zu binden; im NaturalNet, wo die Schatten
Hi,
Is there some SIP magic that can trigger MusicOnHold on my end?
obviously, putting a call on hold will trigger music on hold.
Mayber your gateway does that when all outbound channels are busy or
something?
-nik
--
Wer den Grünkohl nicht ehrt, ist der Mettwurst nicht wert!
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YosLogan ss.xs49...@gmail.com schrieb:
Hello,all
Is the text chat such as LINE not possible in Asterisk?
Yes, SIP/SIMPLE is possible with some workarounds, but only on a connected
channel IIRC. Google is your friend.
- -nik
- --
Diese
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s m sam.gh1...@gmail.com schrieb:
hello all,
i have problem in using g729 codec. my asterisk version is 1.8.22. when
i
run core show codecs in asterisk, there is a g729 codec in the list
so i
assume that i can use it for my channels. but
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Hi,
about g729, you mean if it get free g729 and all my systems (PBXs and
routers) use g729 codec for setting a call, call is set without any
problem?
Yes, if all systems use g729 directly, you are ready to go.
- -nik
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Hi,
for a call routing setup with my mobile phone, I'd like to set the CDO feature
on an outgoing SIP call. I know my SIP proxy provider will pass the call to a
Teles SS7 gateway, keeping most of it intact.
The goal is to forward all calls from the mobile phone number to Asterisk,
which will
