[OpenSIPS-Users] Caller Name and P-Assterted

Bogdan-Andrei Iancu bogdan at opensips.org
Mon Apr 29 21:33:14 CEST 2013


Hello Sukanya,

Hard to tell - 1.6.2 is a real ancient release, do not know what kind of 
hanging you experience ( could be I/Os, not code issues).

Anyhow I do advice to upgrade to 1.8 and 1.9.

Regards,

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


On 04/29/2013 05:55 PM, Sukanya R wrote:
> 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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