Dial with customer send via SIP to 3rd party
Posted: Mon Nov 18, 2013 3:39 pm
Hello.
Vicidial
VERSION: 2.8-415a
BUILD: 131007-1234
Dedicated Server 24gig ram
I have a predictive campaign that currently is calling via lets say provider A using 9 as the dial prefix.
I need to be able to have my agents click the dial with customer option and have the transfer/conf call only sent via SIP to another provider.. Currently we are sending via our default carrier which sends via PSTN..
I figure I can setup the other server as a carrier B with a dial prefix of 5 but am at a loss for how I make the dial with customer option send via this route.
here are the directions provided
"Hostname: voip-xxx.xxxx.com
Port: 5070
They must include the 10 digit TPV phone number in the SIP INVITEs.
Please be sure to use the hostname, rather than the IP address, as the
address may change.
If using asterisk (or equivalent), be sure that "qualify=no" (which
is probably the asterisk default, but may be overridden), and
"dtmfmode=rfc2833"."
Or is there a way to send the call via SIP Uri without setting up a carrier??
If I am not explaining correctly please let me know what other information I need to provide.
Thanks
Vicidial
VERSION: 2.8-415a
BUILD: 131007-1234
Dedicated Server 24gig ram
I have a predictive campaign that currently is calling via lets say provider A using 9 as the dial prefix.
I need to be able to have my agents click the dial with customer option and have the transfer/conf call only sent via SIP to another provider.. Currently we are sending via our default carrier which sends via PSTN..
I figure I can setup the other server as a carrier B with a dial prefix of 5 but am at a loss for how I make the dial with customer option send via this route.
here are the directions provided
"Hostname: voip-xxx.xxxx.com
Port: 5070
They must include the 10 digit TPV phone number in the SIP INVITEs.
Please be sure to use the hostname, rather than the IP address, as the
address may change.
If using asterisk (or equivalent), be sure that "qualify=no" (which
is probably the asterisk default, but may be overridden), and
"dtmfmode=rfc2833"."
Or is there a way to send the call via SIP Uri without setting up a carrier??
If I am not explaining correctly please let me know what other information I need to provide.
Thanks