On 10/06/2010 12:55 PM, JR Richardson wrote:
So the first thing I notice with using 'uac_replace_from()' is the
caller ID number is replaced with the replaced value, so I guess this
is not what I need for my particular application. I'll fool around
with the 'append_hf' and some pattern matches
>> Hi Alex. I use a general context in asterisk to route calls already
>> and wanted to get away from pattern matches. I'm using asterisk
>> realtime and want to decrease the load. I'll play around with the
>> 'uac_replace_from()' and 'append_hf' to see what works best for me.
>
> Well, you don'
JR,
On 10/05/2010 04:00 PM, JR Richardson wrote:
Hi Alex. I use a general context in asterisk to route calls already
and wanted to get away from pattern matches. I'm using asterisk
realtime and want to decrease the load. I'll play around with the
'uac_replace_from()' and 'append_hf' to see w
>> In Kamailio, how would I go about receiving a sip request, append a
>> user "sipentry1" then forward it to Asterisk? I would be using some
>> sort of trunk prefix to identify which sip request to append the user
>> like:
>>
>> 552145551...@siprouter, strip 55, append user "sipentry1", dispatch
JR,
On 10/05/2010 02:00 PM, JR Richardson wrote:
In Kamailio, how would I go about receiving a sip request, append a
user "sipentry1" then forward it to Asterisk? I would be using some
sort of trunk prefix to identify which sip request to append the user
like:
552145551...@siprouter, strip 55
Hi All,
I am using Kamilio 3.0.2 as a load balancer in front of Asterisk
servers, using Dispatcher/PDT and such, working fine. I would like to
be able to bring sip calls into Asterisk at different entry points in
the dialp plan, so I want to setup sip users; [sipentry1] contex=blah,
[sipentry2] c
