Sorry, I wasn't maybe precise in my question. What I am looking for is
to use custom cadences (as defined in indications.conf) for ring tone
generated by 'r' option in a Dial command. Just found this patch:
https://issues.asterisk.org/view.php?id=14504
which does exactly this:
Hi Giorgio
i've tried to upgrade to 1.4.25.1 but still have the same problem.
It seems that the problem is that the call is hangup by asterisk
before the connection with the transferred peer, but i haven't already
been able to reproduce it in a clear debug environment.
Is someone can helps...
Hi,
I have reinstalled Asterisk 1.4 on a machine which was previously running
Asterisk 1.2 on Cent OS 4.7. It had a 24 port digium analog card TDM2400. I
am using this machine only for making outgoing calls. Everything was working
fine with Asterisk 1.2. But after I did the reinstallation with
Ironhide*CLI dialplan show phones
[ Context 'phones' created by 'pbx_config' ]
Include ='internal'[pbx_config]
-= 0 extensions (0 priorities) in 1 context. =-
Ironhide*CLI dialplan show internal
[ Context 'internal' created by 'pbx_config' ]
'1000'
Hi Guys,
Any suggestions on any open source soft phones that has IAX and
SIP support.
I would also like to some programming over it and interface it with
address book or LDAP in order to make the call making easier for the
users.
Thanks
Manoj
What are these dns queries for? I'd like to disable them but I cant
find any obvious reference to them in the asterisk source.
I'm running Asterisk 1.4.21.2
I call voicemail and immediately hang up:
I called from a sip client called line1, but I have no idea where
08c5b9e0 is coming from...
Hello,
I am revising this issue again because I am looking to upgrade to the latest
1.4 Asterisk. When I recompiled, I again, lost my conference rooms because
the chan_zap was not available in the make menuconfig.
Can anyone help me so I can easily upgrade this PBX? I'd hate to have to
