Hi all,
I am trying to implement call redirection feature using cpl script.
I have complied cpl module that supports the cpl script.
I have uploaded the cpl script.
plz help me to solve the problem..
I have written this mail regarding the error that
> we are getting in testing the cpl script of Call redirection.
>
> We have three X-lite phones with SIP uris 2001(a)172.16.1.24,
> ,2002(a)172.16.1.28 and 2003(a)172.16.1.31 and all the three soft phones
> have been …
[View More]registered with SER proxy running on 10.2.3.123. All the
> three users have added in the SER database along with mail ids.
>
> user mail id
> 2001 2001(a)172.16.1.24
> 2002 2002(a)172.16.1.28
> 2003 2003(a)172.16.1.31
>
> The call scenario is as follows..
> 1.user 2001 will make a call to user 2002.
> 2. If the user 2002 is either not responding (No answer) or busy, then
> the call
> will be redirected to user 2003.
> 3. we able to make a call to 2002,and user 2002's X-lite phone is
> ringing but the call is
> not getting redirected.
>
> The cpl script for redirection has been deployed in SER proxy.
>
> I am sending the call log of sip phone:2001 as an attachment along
> with snapshot sip phone 2001.
>
> I am sending cpl script at the bottam of this mail.plz find the cpl script
at the end of this mail.
> As shown in the xlite screen "500 CPL script execution failed" can u
> please suggest me if there is
> any error in XML/CPL syntax.
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> ------------------------------------------------------------------------
>
> ------------------------------------------------------------------------
>
> <?xml version="1.0"?>
> <!DOCTYPE html cpl PUBLIC "-//IETF//DTD RFCxxx CPL 1.0//EN" "cpl.dtd">
> <cpl>
> <subaction id="pda">
> <location url="sip:2003@172.16.1.31 <sip:1003@172.16.1.31>">
> <proxy/>
> </location>
> </subaction>
> <incoming>
> <location url="sip:2002@172.16.1.28 <sip:1002@172.16.1.28>">
> <proxy timeout="8">
> <busy>
> <sub ref="pda"/>
> </busy>
> <noanswer>
> <sub ref="pda"/>
> </noanswer>
> </proxy>
> </location>
> </incoming>
> </cpl>
[View Less]
The more obvious question is:
How does it know there is no more bandwidth available? As a signalling proxy, surely it has no visibility of the RTP usages or requirements.
I guess if it counted all the complete INVITE-> OK dialogues and the SDP codec used it could estimate usage? But the RTP/media path could be different in each case of these so again the same question as above......
Neill...;o)
----- Original Message ----
From: Henning Westerholt <henning.westerholt(a)1und1.de>
To:…
[View More] users(a)openser.org; yanlin <yanlin(a)fortinet.com>
Sent: Tuesday, 3 April, 2007 10:41:31 AM
Subject: Re: [Users] does openser support QoS ?
On Tuesday 03 April 2007 11:15, yanlin wrote:
> Hi,
>
> does openser support some kind of QoS ?
> like, band width management, when there has no more usable bandwidth, then
> new INVITE should be rejected.
Hi Yanlin,
if the bandwith is exceeded, no new INVITEs will get to the server and will
timeout. :-) The "pike" module does some kind of DOS protection, but this is
also no real QoS.
There was a discussion two weeks ago about this topic on devel (the topic
was "rand function in config file"), Klaus suggested some kind of bandwith
management module there. But at the moment such a module don't exist in
OpenSER.
Cheers,
Henning
_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
[View Less]
Two new modules were just introduced to SVN -- both are related to
presence services.
1) pua_bla - implements Bridged Line Appearances according to the
specifications in draft-anil-sipping-bla-03.txt. BLA, aka Shared
Call/Line Appearance, comes from old telephony, where many devices share
same line. Translated to SIP, means that many SIP devices share same
AoR, but only one call can be established at a time, either addressed or
initiated from the line. The call can be put on hold, and …
[View More]the others
devices are notified so that they can pick up the call (the typical
example is that with the secretary answering the call and putting it on
hold to be picked up by the boss).
The SIP devices must have client-side support for BLA as well. In out
tests we used POLYCOM SoundPoint IP 430 SIP phones. The behaviour for
this specific device: when picking up the handset form one phone a red
light appears at the other phones. If attempting to call from a second
phone using the same line, you get the message "Resources full". If a
call is established, when putting it on hold, the light starts to
flicker at the other phones and it is possible to pick it up.
2) pua_xmpp - is a presence gateway between SIMPLE and XMPP. It comes to
complete the IM gateway, by enabling presence state information exchange
between SIP/SIMPLE and Jabber/XMPP clients. For testings we used Gaim
2.0.0beta5, Psi v0.10 for XMPP and Xlite v3.0 for SIP.
Some sample config file to implement above features will be published
soon on dokuwiki.
[View Less]
I am using SER ottendorf with TLS protocol and have
the following issues. Does anybody experience similar
problems?
SER cannot run with the following setup in the
configuration file: (I follow this link to setup key
and certificate:
http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/modules/tls/README…)
modparam("tls", "private_key", "cakey.pem")
modparam("tls", "certificate", "cacert.pem")
modparam("tls", "ca_list", "calist.pem")
modparam("tls", "cipher_list", "HIGH");
With the last …
[View More]line commented out:
#modparam("tls", "cipher_list", "HIGH");
SER can start, but the tls connection cannot be
established. Network trace shows SER does not responde
to ClientHello sent by client.
Thanks,
Joy
____________________________________________________________________________________
Food fight? Enjoy some healthy debate
in the Yahoo! Answers Food & Drink Q&A.
http://answers.yahoo.com/dir/?link=list&sid=396545367
[View Less]
Hi all!
Can somebody explain me how I can fill response with my useful headers?
For example, I need to send 305 "Use proxy" response with Contact
headers. Is it possible? If so, how can I populate it?
Thanks!
--
CU,
Victor Gamov
What happens if I disable rtpproxy but still load the nathelper?
Can I still solve NAT problems with that??
--
-------------------------------------------------------------------
Amanda Remes Mattiuz
amattiuz(a)gmail.com
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Hello,
I have a question about openser: Does openser support RFC 3428 (Instant
Messaging (IM))? If so, How do I have to configure openser to relay SIP
method MESSAGE ?
regards
helmut
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I experienced the same problem with the SNMPstats module. If you run openser in foreground mode (fork=no), the module doesn't initialize. Running in background mode fixes the problem.