Re: [OpenSIPS-Users] reason headers

2012-01-10 Thread Jan D.
I got feedback on the ticket i opened at NEC/Philips on the question why a
400 bad packet response is sent.

This is their answer:

This time SV8100 responded a 400 Bad because the received message did not
meet RFC. According to RFC, the text following after 'text=' should be
enclosed in double quotation marks.

Bad - SIP;cause=487;text=ORIGINATOR_CANCEL
Good - SIP;cause=487;text=ORIGINATOR_CANCEL

Can one of the maintainers respond to this. If this is true, how can i add
the quotation marks or can this be repaired in 1.8.x?

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Re: [OpenSIPS-Users] reason headers

2012-01-10 Thread Saúl Ibarra Corretgé
It's already fixed, at least in trunk. Not sure about 1.7 branch, but you can 
check it in the code ;-)

On Jan 10, 2012, at 10:25 AM, Jan D. wrote:

 I got feedback on the ticket i opened at NEC/Philips on the question why a
 400 bad packet response is sent.
 
 This is their answer:
 
 This time SV8100 responded a 400 Bad because the received message did not
 meet RFC. According to RFC, the text following after 'text=' should be
 enclosed in double quotation marks.
 
 Bad - SIP;cause=487;text=ORIGINATOR_CANCEL
 Good - SIP;cause=487;text=ORIGINATOR_CANCEL
 
 Can one of the maintainers respond to this. If this is true, how can i add
 the quotation marks or can this be repaired in 1.8.x?
 
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Re: [OpenSIPS-Users] reason headers

2012-01-10 Thread Jan D.
Hi,

Thanks for the quick response. I will test vrom trunk later and will let you
know if its fixed.

For now, we run a production server under 1.7.1 and have problems at this
moment.

So I have still the question, can I get rid of the Reason header or change
the text (add quotations?).


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Re: [OpenSIPS-Users] reason headers

2012-01-10 Thread Saúl Ibarra Corretgé

On Jan 10, 2012, at 10:52 AM, Jan D. wrote:

 Hi,
 
 Thanks for the quick response. I will test vrom trunk later and will let you
 know if its fixed.
 
 For now, we run a production server under 1.7.1 and have problems at this
 moment.
 
 So I have still the question, can I get rid of the Reason header or change
 the text (add quotations?).
 

Latest trunk does have quotations, so if you can you should use that. I f you 
want to get rid of it I'm afraid you'll need to manually edit the code.

Now, I do think that having a way to toggle adding the Reason header to CANCEL 
requests could be useful... perhaps a tm modparam would make sense.


Regards,

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[OpenSIPS-Users] Modify Via header

2012-01-10 Thread Maciej Bylica
Hello,

What is the best way to replace or modify Via header of incoming INVITE?
I need to change private ip address with $si.
Oryginal header is Via: SIP/2.0/UDP 10.10.10.128:5060;branch=z9hG4bK-680826

Is it subst? What is your advice?

Regards,
Maciej

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Re: [OpenSIPS-Users] [NEW] Cassandra Database Driver

2012-01-10 Thread Pete Kelly
That looks very good, do you know if there is scope in the future to store
SIP registration information in Cassandra (or KVP) so that a multi location
registration platform can be achieved?

On 5 January 2012 13:50, Vlad Paiu vladp...@opensips.org wrote:

 **
 Hello,

 A new *CACHEDB_CASSANDRA* module has been added to OpenSIPS trunk, and it
 will also be present in the future 1.8 release.
 The module implements the Key-Value Interface [1] to allow the script
 writer and also the module writer to connect and do Key-Value type
 operations ( store,fetch and remove ) on a Cassandra database.

 For a full documentation of the module and current limitations, visit [2] .
 Testing and feedback is very much welcomed.

 [1] http://www.opensips.org/Resources/DocsTutKvinterface
 [2] http://www.opensips.org/html/docs/modules/devel/cachedb_cassandra.html

 Regards,

 --
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 OpenSIPS Developer


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Re: [OpenSIPS-Users] [NEW] Cassandra Database Driver

2012-01-10 Thread Vlad Paiu

Hello Pete,

Yes, we are also planning to support distributed User Location, through 
the Key-Value Interface. After that, you will be able to use Cassandra 
or whatever other cachedb_* module in order to use your preferred 
back-end and support the functionality.


So far we have implemented distributed ratelimiting and also distributed 
dialog profiles are almost done.


Regards,

Vlad Paiu
OpenSIPS Developer


On 01/10/2012 02:48 PM, Pete Kelly wrote:
That looks very good, do you know if there is scope in the future to 
store SIP registration information in Cassandra (or KVP) so that a 
multi location registration platform can be achieved?


On 5 January 2012 13:50, Vlad Paiu vladp...@opensips.org 
mailto:vladp...@opensips.org wrote:


Hello,

A new *CACHEDB_CASSANDRA* module has been added to OpenSIPS trunk,
and it will also be present in the future 1.8 release.
The module implements the Key-Value Interface [1] to allow the
script writer and also the module writer to connect and do
Key-Value type operations ( store,fetch and remove ) on a
Cassandra database.

For a full documentation of the module and current limitations,
visit [2] .
Testing and feedback is very much welcomed.

[1] http://www.opensips.org/Resources/DocsTutKvinterface
[2]
http://www.opensips.org/html/docs/modules/devel/cachedb_cassandra.html

Regards,

-- 
Vlad Paiu

OpenSIPS Developer


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Re: [OpenSIPS-Users] [NEW] Cassandra Database Driver

2012-01-10 Thread Brett Nemeroff
I just wanted to chime in and say that I think this direction is great.
Love the distributed functionality! :)

Thanks Vlad!
-Brett


On Tue, Jan 10, 2012 at 8:05 AM, Vlad Paiu vladp...@opensips.org wrote:

 **
 Hello Pete,

 Yes, we are also planning to support distributed User Location, through
 the Key-Value Interface. After that, you will be able to use Cassandra or
 whatever other cachedb_* module in order to use your preferred back-end and
 support the functionality.

 So far we have implemented distributed ratelimiting and also distributed
 dialog profiles are almost done.

 Regards,

 Vlad Paiu
 OpenSIPS Developer


 On 01/10/2012 02:48 PM, Pete Kelly wrote:

 That looks very good, do you know if there is scope in the future to store
 SIP registration information in Cassandra (or KVP) so that a multi location
 registration platform can be achieved?

  On 5 January 2012 13:50, Vlad Paiu vladp...@opensips.org wrote:

  Hello,

 A new *CACHEDB_CASSANDRA* module has been added to OpenSIPS trunk, and
 it will also be present in the future 1.8 release.
 The module implements the Key-Value Interface [1] to allow the script
 writer and also the module writer to connect and do Key-Value type
 operations ( store,fetch and remove ) on a Cassandra database.

 For a full documentation of the module and current limitations, visit [2]
 .
 Testing and feedback is very much welcomed.

 [1] http://www.opensips.org/Resources/DocsTutKvinterface
 [2]
 http://www.opensips.org/html/docs/modules/devel/cachedb_cassandra.html

 Regards,

 --
 Vlad Paiu
 OpenSIPS Developer


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Re: [OpenSIPS-Users] Modify Via header

2012-01-10 Thread Dani Popa
none,
I think you want and need to use
topology_hiding()http://www.opensips.org/html/docs/modules/devel/dialog.html#id294845
from
dialog module.

Dani

On Tue, Jan 10, 2012 at 1:21 PM, Maciej Bylica mb...@gazeta.pl wrote:

 Hello,

 What is the best way to replace or modify Via header of incoming INVITE?
 I need to change private ip address with $si.
 Oryginal header is Via: SIP/2.0/UDP 10.10.10.128:5060
 ;branch=z9hG4bK-680826

 Is it subst? What is your advice?

 Regards,
 Maciej

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[OpenSIPS-Users] Problems with DB_TEXT

2012-01-10 Thread discodog62
I am trying to just use DB_TEXT.  When I start opensips I get this error.  I am 
using opensips 1.7 on archlinux 64bit.



ERROR:db_text:dbt_query: table does not exist!
ERROR:domain:reload_domain_table: Error while querying database
ERROR:db_text:dbt_query: table does not exist!
ERROR:drouting:dr_load_routing_info: DB query failed
ERROR:core:init_mod_child: failed to initializing module drouting, rank 1
ERROR:core:main_loop: init_child failed for UDP listener





Here is the top of my config.





loadmodule db_text.so
loadmodule signaling.so
loadmodule sl.so
loadmodule tm.so
loadmodule rr.so
loadmodule maxfwd.so
loadmodule usrloc.so
loadmodule dialog.so
loadmodule registrar.so
loadmodule textops.so
loadmodule mi_fifo.so
loadmodule uri.so
loadmodule acc.so
loadmodule alias_db.so
loadmodule domain.so
loadmodule permissions.so
loadmodule rtpproxy.so
loadmodule drouting.so


modparam(mi_fifo, fifo_name, /tmp/opensips_fifo)


modparam(rr, append_fromtag, 0)


modparam(uri, use_uri_table, 0)


modparam(alias_db, db_url,text:///zxa/opensipsdb)


modparam(domain, db_url,text:///zxa/opensipsdb)
modparam(domain, db_mode, 1)   # Use caching
modparam(domain, domain_table, domain)
modparam(domain, domain_col, domain)


modparam(dialog, db_mode, 1)
modparam(dialog, db_url, text:///zxa/opensipsdb)
modparam(dialog, default_timeout, 90)


modparam(drouting, use_domain, 1)
modparam(drouting,db_url, text:///zxa/opensipsdb)


modparam(permissions, db_url,text:///zxa/opensipsdb)



I set the file and folder permissions for  /zxa/opensipsdb to 777.  


I grabbed the database files from the opensips 1.7 media


Is there a step that I am missing




Thanks,


James




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[OpenSIPS-Users] OpenSIPS in a University Environment

2012-01-10 Thread Gabriel Kuri
Does anyone know of any Universities running OpenSIPS for local call
routing between handsets?

We're looking at replacing our old Avaya system or upgrade it, and the
forklift upgrade from Avaya is ridiculously expensive (no surprise).

We'd like to replace our Avaya system with a combination of OpenSIPS
and FreeSWITCH and some Cisco routers for external PSTN access, but
it's going to be a tough sell to our CIO, unless we can show someone
else has done it already.

Any pointers to other Universities would be great.

Cheers,
Gabe

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Re: [OpenSIPS-Users] OpenSIPS in a University Environment

2012-01-10 Thread Jorj Bauer
Hi Gabe,

Penn's serving about 9000 Polycom handsets with a combination of SERv3 (proxy), 
OpenSIPS (presence) and Asterisk (media) servers. We use Cisco 3845 gateways. 
Feel free to contact me off-list.

-- Jorj

-- 
Jorj Bauer
Manager of Engineering, Research and Development
Information Systems and Computing, University of Pennsylvania
215.746.3850
XMPP: j...@upenn.edu


On Jan 10, 2012, at 1:46 PM, Gabriel Kuri wrote:

 Does anyone know of any Universities running OpenSIPS for local call
 routing between handsets?
 
 We're looking at replacing our old Avaya system or upgrade it, and the
 forklift upgrade from Avaya is ridiculously expensive (no surprise).
 
 We'd like to replace our Avaya system with a combination of OpenSIPS
 and FreeSWITCH and some Cisco routers for external PSTN access, but
 it's going to be a tough sell to our CIO, unless we can show someone
 else has done it already.
 
 Any pointers to other Universities would be great.
 
 Cheers,
 Gabe
 
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Re: [OpenSIPS-Users] Problems with DB_TEXT

2012-01-10 Thread discodog62
I figured it out.  I needed to escape the :  like so \:





-Original Message-
From: discodog62 discodo...@aol.com
To: users users@lists.opensips.org
Sent: Tue, Jan 10, 2012 5:14 am
Subject: [OpenSIPS-Users] Problems with DB_TEXT


I am trying to just use DB_TEXT.  When I start opensips I get this error.  I am 
using opensips 1.7 on archlinux 64bit.



ERROR:db_text:dbt_query: table does not exist!
ERROR:domain:reload_domain_table: Error while querying database
ERROR:db_text:dbt_query: table does not exist!
ERROR:drouting:dr_load_routing_info: DB query failed
ERROR:core:init_mod_child: failed to initializing module drouting, rank 1
ERROR:core:main_loop: init_child failed for UDP listener





Here is the top of my config.





loadmodule db_text.so
loadmodule signaling.so
loadmodule sl.so
loadmodule tm.so
loadmodule rr.so
loadmodule maxfwd.so
loadmodule usrloc.so
loadmodule dialog.so
loadmodule registrar.so
loadmodule textops.so
loadmodule mi_fifo.so
loadmodule uri.so
loadmodule acc.so
loadmodule alias_db.so
loadmodule domain.so
loadmodule permissions.so
loadmodule rtpproxy.so
loadmodule drouting.so


modparam(mi_fifo, fifo_name, /tmp/opensips_fifo)


modparam(rr, append_fromtag, 0)


modparam(uri, use_uri_table, 0)


modparam(alias_db, db_url,text:///zxa/opensipsdb)


modparam(domain, db_url,text:///zxa/opensipsdb)
modparam(domain, db_mode, 1)   # Use caching
modparam(domain, domain_table, domain)
modparam(domain, domain_col, domain)


modparam(dialog, db_mode, 1)
modparam(dialog, db_url, text:///zxa/opensipsdb)
modparam(dialog, default_timeout, 90)


modparam(drouting, use_domain, 1)
modparam(drouting,db_url, text:///zxa/opensipsdb)


modparam(permissions, db_url,text:///zxa/opensipsdb)



I set the file and folder permissions for  /zxa/opensipsdb to 777.  


I grabbed the database files from the opensips 1.7 media


Is there a step that I am missing




Thanks,


James




 
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[OpenSIPS-Users] one way dispatcher

2012-01-10 Thread Matt Hamilton


Hi,

Is it possible for the dispatcher module just forward the message to the 
selected destination (an Asterisk server) without adding a Via field to the 
header? In other words, I would like Asterisk to get the message as if it's 
coming from the original source.

Thanks,
Matt
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Re: [OpenSIPS-Users] one way dispatcher

2012-01-10 Thread Brett Nemeroff
On Tue, Jan 10, 2012 at 2:51 PM, Matt Hamilton mistral9...@hotmail.comwrote:


 Hi,

 Is it possible for the dispatcher module just forward the message to the
 selected destination (an Asterisk server) without adding a Via field to the
 header? In other words, I would like Asterisk to get the message as if it's
 coming from the original source.


Not if your are proxying the call. As a redirect server you may be able to
make this work, but it might not meet your requirements.

Alternatively, you can use the topology hiding in B2BUA or dialog to
essentially provide a single Via of OpenSIPs rather than a collection of
Vias.
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Re: [OpenSIPS-Users] OpenSIPS in a University Environment

2012-01-10 Thread Andrew Pogrebennyk
You may want to get in contact with uniza.sk. I believe they are using
opensource voip as they have published some excellent tutorials like
http://nil.uniza.sk/sip/nat-fw/configuring-nat-traversal-using-kamailio-31-and-rtpproxy-server
even though it is about kamailio and not opensips

On 01/10/2012 07:46 PM, Gabriel Kuri wrote:
 Does anyone know of any Universities running OpenSIPS for local call
 routing between handsets?
 
 We're looking at replacing our old Avaya system or upgrade it, and the
 forklift upgrade from Avaya is ridiculously expensive (no surprise).
 
 We'd like to replace our Avaya system with a combination of OpenSIPS
 and FreeSWITCH and some Cisco routers for external PSTN access, but
 it's going to be a tough sell to our CIO, unless we can show someone
 else has done it already.
 
 Any pointers to other Universities would be great.
 
 Cheers,
 Gabe
 


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Re: [OpenSIPS-Users] one way dispatcher

2012-01-10 Thread Matt Hamilton

Is it possible for the dispatcher module just forward the message to the 
selected destination (an Asterisk server) without adding a 
Via field to the header? In other words, I would like Asterisk to get the 
message as if it's coming from the original source.

Not if your are proxying the call. As a redirect server you may be able 
to make this work, but it might not meet your requirements. 


No proxying, I don't want Asterisk reply back to Opensips (if that makes sense 
:)) 

I will look into the drouting module and see if that makes it possible.
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[OpenSIPS-Users] SIP re-register time

2012-01-10 Thread yasiru welandawe
Hi All,

Could someone help me with to find what is the maximum period of time that 
Opensips can keep a sip-registration.

I want to find out the greatest time period that I can set for phones for 
sending re-register requests.

Regards,

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[OpenSIPS-Users] B2BUA on OpenWRT router

2012-01-10 Thread Andrew Radke
Hi all,

I would like to setup a B2BUA SBC on our OpenWRT based router to work with our 
sipX PBX. Very simple requirements:
authenticate outgoing calls for our ITSP
preferably enforce negotiation of G729 due to limited bandwidth
no need to relay RTP, just let it get passed through from the phones to the 
ITSP directly
possibly register with our ITSP for inbound calls which would be sent to the 
PBX (unlikely to be used, but nice as an option)
An advantage would also be that it can be started and stopped, etc in line with 
the state of the Internet connection rather than waiting for a timeout if the 
Internet is down. The PBX would then automatically route calls via the PSTN SBC.

I've looked at the B2BUA documentation but it uses two modules that don't 
appear to be available in OpenWRT: b2b_entities and b2b_logic.

Is it possible to get opensips to act as a B2BUA with the available modules 
listed below, possibly with an external script. The version is given as 1.5.3-1 
if that affects anything.

Regards,
Andrew Radke
Yuruga Nursery Pty Ltd
Clonal Solutions Australia Pty Ltd
PO Box 220
Walkamin Qld 4872
Phone: (07) 4093 3826
Fax: (07) 4093 3869
Email: andrew.ra...@yuruga.com.au
Web: www.yuruga.com.au


opensips
opensips-example
opensips-mod-acc
opensips-mod-alias-db
opensips-mod-auth
opensips-mod-auth-db
opensips-mod-auth-radius
opensips-mod-avp-radius
opensips-mod-avpops
opensips-mod-benchmark
opensips-mod-call-control
opensips-mod-cfgutils
opensips-mod-closeddial
opensips-mod-cpl-c
opensips-mod-db-flatstore
opensips-mod-db-mysql
opensips-mod-db-postgres
opensips-mod-db-text
opensips-mod-dialog
opensips-mod-dialplan
opensips-mod-dispatcher
opensips-mod-diversion
opensips-mod-domain
opensips-mod-domainpolicy
opensips-mod-drouting
opensips-mod-enum
opensips-mod-exec
opensips-mod-gflags
opensips-mod-group
opensips-mod-group-radius
opensips-mod-h350
opensips-mod-identity
opensips-mod-imc
opensips-mod-jabber
opensips-mod-lcr
opensips-mod-ldap
opensips-mod-load_balancer
opensips-mod-localcache
opensips-mod-mangler
opensips-mod-maxfwd
opensips-mod-mi-datagram
opensips-mod-mi-fifo
opensips-mod-msilo
opensips-mod-nat_traversal
opensips-mod-nathelper
opensips-mod-options
opensips-mod-path
opensips-mod-pdt
opensips-mod-peering
opensips-mod-permissions
opensips-mod-pike
opensips-mod-presence
opensips-mod-presence-dialoginfo
opensips-mod-presence-mwi
opensips-mod-presence-xcapdiff
opensips-mod-presence-xml
opensips-mod-pua
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opensips-mod-qos
opensips-mod-ratelimit
opensips-mod-regex
opensips-mod-registrar
opensips-mod-rls
opensips-mod-rr
opensips-mod-seas
opensips-mod-signaling
opensips-mod-siptrace
opensips-mod-sl
opensips-mod-sms
opensips-mod-speeddial
opensips-mod-sst
opensips-mod-statistics
opensips-mod-textops
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opensips-mod-tm
opensips-mod-uac
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