[OpenSIPS-Users] Caller Name and P-Assterted

[Sukanya R]

Hey team,

Do you know if 1.6.2 release of opensip has issues of hanging often, will upgrading to 1.8 issue can fix it ? 


Warm regards,
Sukanya R

-----Original Message-----
From: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Monday, April 29, 2013 7:27 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Caller Name and P-Assterted

Use avp_db_load() or avp_db_query() from the avpops module.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 04/29/2013 04:21 PM, Nick Khamis wrote:
> Hello Bogdan,
>
> As usual. Thank you so much! In the past year I learned so much about 
> SIP then I did in the past five using the different media servers. The 
> gateway I sent the PAI hdr seems to  accept it, but I the CallID Name 
> part does not seem to be supported by the carrier. Ce la vie...
>
> On the question about rpid, I have:
> "Test User"<sip:15453387463 at test.server.com>;user=phone\r\n stored in 
> subscriber.rpid. In the branch and failure routes, I would like to 
> load this value into append_hf. What is the best way of doing this 
> since we do not perform any authentication for registration or invites 
> on our system, and thus no "consume_credentials"
>
>
> Kind Regards,
>
> Nick
>
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> Users at lists.opensips.org
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