Hi all!
I have installed openser through CVS from the following link
http://openser.org/dokuwiki/doku.php?id=install_and_maintain_openser_from_c…
till the 8 th step in the link everything went well fine. in the 9 th step,
there is a link to make oepnser.init script work and i did all the steps as
described. I downloaded and installed the monit daemon to start openser thru
it. however, now if i type /etc/init.d/openser start, it says command monit
to start openser and then the command prompt appears back. i dont see the
tcp and udp ports (5060 ) and other information etc when the openser is
started.....and i cant figure out if it is running at all.....if i issue ps
command, i dont see any pid for openser.....if i open the web interface of
the monit daemon, i see that it is trying to start the openser but after
some time it shows failed to start......
please let me know what to do asap.....
Thanks
----- Original Message -----
From: <users-request(a)openser.org>
To: <users(a)openser.org>
Sent: Friday, May 26, 2006 3:30 PM
Subject: Users Digest, Vol 12, Issue 71
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> Today's Topics:
>
> 1. Re: How to extract sip-id when using alias-id?
> (Bogdan-Andrei Iancu)
> 2. Route Algorithm. (raviprakash sunkara)
> 3. Re: Forward SUBSCRIBER (Bogdan-Andrei Iancu)
> 4. RE: 2 Openser Instances (Sam Lee)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Fri, 26 May 2006 12:35:21 +0300
> From: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro>
> Subject: Re: [Users] How to extract sip-id when using alias-id?
> To: Frogger <froggerandbongo(a)yahoo.com>
> Cc: users(a)openser.org
> Message-ID: <4476CBD9.80002(a)voice-system.ro>
> Content-Type: text/plain; charset=us-ascii; format=flowed
>
> Hi,
>
> what you can do is to save into an AVP the alias before doing
> lookup(aliases). In failure route, before sending to voicemail, push the
> AVP into RURI to have the alias and not the sip id.
>
> regards,
> bogdan
>
> Frogger wrote:
>
>>Good point.
>>I need to clarify...
>>
>>The alias is working to route to the user however if
>>the timer expires, the route fails or if there is a
>>reject, the failure routes are applying their logic to
>>the alias-id not the sip-id.
>>
>>What I need to do is apply the failure route logic to
>>the sip-id so we can send the attempt to voicemail.
>>
>>Let say I have an alias of 1000(a)sip.com for sip-id
>>5551212(a)sip.comip.com.
>>
>>When 1000(a)sip.com is dialed by another user in the
>>domain (I am using "use_domain") the call is properly
>>routed to: 5551212(a)sip.com.
>>
>>However, if 5551212(a)sip.com does not answer, timer
>>expires, or a reject, then the failure logic is be
>>applied to the alias.
>>
>>Failure route says:
>>
>>prefix("66");
>>append_branch();
>>append_urihf("CC-Diversion: ", "\r\n");
>>t_relay();
>>
>>
>>Result of the failure logic:
>>
>>661000(a)sip.com
>>
>>Needs to be:
>>
>>665551212(a)sip.com
>>
>>
>>Thoughts?
>>
>>F
>>
>>
>>--- Bogdan-Andrei Iancu <bogdan(a)voice-system.ro>
>>wrote:
>>
>>
>>
>>>Hi,
>>>
>>>not sure I understand - you dial an alias and via
>>>db_aliases you replace
>>>the alias with the sip-id. So you have the sip-id ..
>>>where is the problem?
>>>
>>>regards,
>>>bogdan
>>>
>>>Frogger wrote:
>>>
>>>
>>>
>>>>I am using alias_db to store alias's for users.
>>>>
>>>>However when an alias is dialed I need to also be
>>>>
>>>>
>>>able
>>>
>>>
>>>>to restore the alias owners original sip-id.
>>>>
>>>>This is necessary to forward the call to voicemail
>>>>using sip-id.
>>>>
>>>>Any guidance on getting the sip-id out of the
>>>>
>>>>
>>>alias_db
>>>
>>>
>>>>and replacing the alias-id prior to forwarding to
>>>>vmail?
>>>>
>>>>Thanks in advance!
>>>>
>>>>F
>>>>
>>>>__________________________________________________
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>>>>
>>>protection around
>>>
>>>
>>>>http://mail.yahoo.com
>>>>
>>>>_______________________________________________
>>>>Users mailing list
>>>>Users(a)openser.org
>>>>http://openser.org/cgi-bin/mailman/listinfo/users
>>>>
>>>>
>>>>
>>>>
>>>>
>>>
>>>
>>
>>
>>__________________________________________________
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>>http://mail.yahoo.com
>>
>>
>
>
>
>
> ------------------------------
>
> Message: 2
> Date: Fri, 26 May 2006 05:38:23 -0400
> From: "raviprakash sunkara" <sunkara.raviprakash.feb14(a)gmail.com>
> Subject: [Users] Route Algorithm.
> To: Users(a)openser.org, "Klaus Darilion"
> <klaus.mailinglists(a)pernau.at>at>, daniel(a)voice-system.ro,
> nickm(a)kryptontech.com
> Message-ID:
> <f0f0f9340605260238l2e6ce419qd109b6f97e57c485(a)mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi listers and Users.
>
> Please help me in " how to make a call from one place to other place in
> route algorithm "
>
>
> I got Call not approved ? what is this and also $04 and $08 messages as
> not
> found and request timeout........
>
>
> It connect ( REgister) to server .
>
> server respond as 1 st " 200 ok " and then " 408 request time
out....
>
> plesse help me
>
> Bye..
> --
> Thanks and Regards with cheers
> Sunkara Ravi Prakash (Voip Developer)
> Hyperion Technology
>
www.hyperion-tech.com
> <b>
>