A few months ago I started using and had to abandon PJSIP because my
dialplan could not read the inbound signalling IP address, which I can
read now in Asterisk11 using CHANNEL(recvip). My app relies on this
information. The
question is, is it possible now access the signalling IP of an
incoming SI
On Wed, Jun 18, 2014 at 6:05 AM, CDR wrote:
> A few months ago I started using and had to abandon PJSIP because my
> dialplan could not read the inbound signalling IP address, which I can
> read now in Asterisk11 using CHANNEL(recvip). My app relies on this
> information. The
> question is, is it
Anyone using the French DID provider num2sip? Could you share the
sip.conf setup?
Thanks,
sean
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I am debugging an intermittent issue of missing audio on calls that come from a SIP provider into our asterisk-11.10 installation. Sometimes, incoming calls from this provider work correctly, with audio streaming in both directions. Other times, with the
same setup, the calling party is unable to
Hi guys
How can I record the confbridge only when after a marked user is logged in to
conference ? Is there any option on the confbridge to start recording when
marked user is logged in instead of when the first user logs in ? I tried
setting up "same => n,Set(CONFBRIDGE(bridge,record_conferen
Is this completely open source? And all it's dependencies?
https://github.com/trustmaster/SugarAsterisk
Is it free as in free speech and free as in free beer? I know there are
a few variations of SugarCRM.
We're currently using an Asterisk hosted PBX with Cisco hardphones.
Conceivably, we
http://www.voip-info.org/wiki/view/Asterisk+CRM+Integration
lists a few options. I'm looking for, literally, the simplest FOSS CRM
for "click to dial" functionality, but don't know where to start.
thanks,
Thufir
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