Hello,
Yes, actually my call flow for outbound calls is, Kamailio => Asterisk =>
Kamailio (route(PSTN)). SIP trunk is available at Kamailio server.
So, when final outbound INVITE goes to the SIP trunk provider it expects a
particular Contact header IP, instead of Asterisk server IP.
On Mon, Aug 1, 2022 at 10:20 PM Alex Balashov <abalashov(a)evaristesys.com>
wrote:
Before you do this, I would ask yourself why you’re
doing this, and
whether it’s a good idea from a technical point of view.
On Aug 1, 2022, at 8:36 AM, Ankit Jayswal
<ankit.jayswal(a)engagely.ai>
wrote:
Hello Team,
I am in need to modify the contact header in Kamailio before sending an
INVITE
request to the SIP trunk provider. Basically the IP address I need
to modify in the contact header. Please suggest a better way to do this.
Below is an example.
OLD (Existing):
Contact: <sip:009169240xxx@45.118.162.x:5080;alias=10.52.26.102~5080~1>
NEW (Needed):
Contact: <sip:00912269240xxx@10.52.26.98:5060>
I tried various functions from the textops module but no hope.
I am looking for a function that can replace the contact header's value.
Please help me with this.
--
Thanks & Regards,
Ankit Jayswal | Specialist - Software Development
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--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web:
http://www.evaristesys.com/,
http://www.csrpswitch.com/
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