[OpenSIPS-Users] How to force opensips add port field in via header?

2010-02-02 Thread Lei Tang
  Hi everyone, I'm using opensips 1.6.
  When I send sip msg by call forward function, opensips add Via header in
sips message like "Via: SIP/2.0/UDP
10.37.143.6;branch=z9hG4bKSS1vtZ6ZQ98rp."
   THE QUESTION IS: How can I make opensips add the port field to the VIA
header even my sip port is 5060? This mean, the VIA header should be
  "Via: SIP/2.0/UDP 10.37.143.6*:5060*;branch=z9hG4bKSS1vtZ6ZQ98rp."
   Does someone can give me some advice about this question? Thank a lot!

-- 
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[OpenSIPS-Users] One way audio - media port changed (opensips / mediaproxy)

2010-02-02 Thread Magnus Burman
Hi guys,

I've got a problem with "sporadic" one-way audio, using latest stable
Opensips 1.4 and Mediaproxy 2.

The problem occurs at re-invites, thou not every time. When it does happen,
the media port is changed towards the callee. Example:

First invite:
INVITE Trunk:40518 --> Proxy:50358
INVITE Proxy:58928 --> UA:5206

Re-invite:
INVITE Trunk:40518 --> Proxy:50358
INVITE Proxy:40518 --> UA:5206

Notice how the Proxy port on the UA side (callee local) changed to the same
port that Trunk uses (caller remote).

One way audio now occurs and naturally the call is soon hung up. Mediaproxy
now dumps some useful statistics showing:

'callee_remote': 'UA:5206'
'caller_remote': 'TRUNK:40518'
'callee_local':  'PROXY:58928'
'caller_local':  'PROXY:50358'

Notice here how the callee_local is the original port used.

Where should I start looking to be able to solve this problem? Is it most
likely my config, something weird with the re-invite or have I stumbled upon
a bug?

Thankful for every and all suggestions, as I lay sleepless at night thinking
about this! :-P

Cheers,
Magnus
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Re: [OpenSIPS-Users] How to force opensips add port field in via header?

2010-02-02 Thread Iñaki Baz Castillo
El Martes, 2 de Febrero de 2010, Lei Tang escribió:
>   Hi everyone, I'm using opensips 1.6.
>   When I send sip msg by call forward function, opensips add Via header in
> sips message like "Via: SIP/2.0/UDP
> 10.37.143.6;branch=z9hG4bKSS1vtZ6ZQ98rp."
>THE QUESTION IS: How can I make opensips add the port field to the VIA
> header even my sip port is 5060? This mean, the VIA header should be
>   "Via: SIP/2.0/UDP 10.37.143.6*:5060*;branch=z9hG4bKSS1vtZ6ZQ98rp."
>Does someone can give me some advice about this question? Thank a lot!
> 

There is a core parameter for that, I can not remember now the name but you 
can find it in the wiki (cookbook section).

-- 
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Re: [OpenSIPS-Users] One way audio - media port changed (opensips / mediaproxy)

2010-02-02 Thread Saúl Ibarra Corretgé
Hi,

El 02/02/10 11:54, Magnus Burman escribió:
> Hi guys,
>
> I've got a problem with "sporadic" one-way audio, using latest stable
> Opensips 1.4 and Mediaproxy 2.
>
> The problem occurs at re-invites, thou not every time. When it does
> happen, the media port is changed towards the callee. Example:
>
> First invite:
> INVITE Trunk:40518 --> Proxy:50358
> INVITE Proxy:58928 --> UA:5206
>
> Re-invite:
> INVITE Trunk:40518 --> Proxy:50358
> INVITE Proxy:40518 --> UA:5206
>
> Notice how the Proxy port on the UA side (callee local) changed to the
> same port that Trunk uses (caller remote).
>
> One way audio now occurs and naturally the call is soon hung up.
> Mediaproxy now dumps some useful statistics showing:
>
> 'callee_remote': 'UA:5206'
> 'caller_remote': 'TRUNK:40518'
> 'callee_local': 'PROXY:58928'
> 'caller_local': 'PROXY:50358'
>
> Notice here how the callee_local is the original port used.
>
> Where should I start looking to be able to solve this problem? Is it
> most likely my config, something weird with the re-invite or have I
> stumbled upon a bug?
>
> Thankful for every and all suggestions, as I lay sleepless at night
> thinking about this! :-P
>

It would be nice to have a full SIP trace and the syslog output of 
MediaProxy to check if it's doing something wrong. Also, which function 
are you using for starting it?



Regards,


-- 
Saúl Ibarra Corretgé
AG Projects

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Re: [OpenSIPS-Users] Error in module permission with db_text

2010-02-02 Thread Leonid Nasedkin
Hi, Bogdan.

I have got one more error with module db_text.

loadmodule "avpops.so"
modparam("avpops","db_url","text:///etc/opensips/dbtext/")
modparam("avpops","avp_table","usr_preferences")
modparam("avpops","use_domain",0)
modparam("avpops","uuid_column","uuid")
modparam("avpops","username_column","username")
modparam("avpops","domain_column","domain")
modparam("avpops","attribute_column","attribute")
modparam("avpops","type_column","type")
modparam("avpops","value_column","value")
...
avp_db_load("mediaproxy-RU","$avp(s:mediaproxy_ip)")
...

usr_preferences:
id(int,auto) uuid(str) username(str) domain(str) attribute(str) type(int)
value(str) last_modified(double,null)
10:mediaproxy-DEFAULT:mediaproxy-DEFAULT:domain:mediaproxy_ip:0:value:0
20:mediaproxy-RU:mediaproxy-RU:domain:mediaproxy_ip:0:value:0

And I see in log:
Feb  2 12:03:51 dev-sip /usr/sbin/opensips[19536]:
DBG:db_text:dbt_load_file: loading file
[/etc/opensips/dbtext//usr_preferences]
Feb  2 12:03:51 dev-sip /usr/sbin/opensips[19536]: ERROR:db_text:dbt_query:
table does not exist!
Feb  2 12:03:51 dev-sip /usr/sbin/opensips[19536]:
ERROR:avpops:ops_dbload_avps: db_load failed

2010/1/26 Bogdan-Andrei Iancu 

> Hi Leonid,
>
> Thanks for testing - I uploaded the fix on the 1.6 branch also.
>
>
> Regards,
> Bogdan
>
> Леонид Наседкин wrote:
>
>> Hi, Bogdan.
>> Its working now. Thanks.
>> 2010/1/26 Bogdan-Andrei Iancu > bog...@voice-system.ro>>
>>
>>
>>Hi Leonid,
>>
>>An official fix is available on SVN trunk (rev 6534). I would
>>really appreciate if you could give it a try and test - if ok, I
>>will do the backport.
>>
>>Thanks and regards,
>>Bogdan
>>
>>
>>Bogdan-Andrei Iancu wrote:
>>
>>Ok, I will investigate to come up with an official fix.
>>
>>Thanks and regards,
>>Bogdan
>>
>>Леонид Наседкин wrote:
>>
>>Hi Bogdan
>>Thank you. Its working now.
>>
>>2010/1/15 Bogdan-Andrei Iancu >
>>>>>
>>
>>   Hi Leonid,
>>
>>   Looks like there is a compatibility bug between
>>permission and
>>   db_text modules when comes to DB data typesGive me
>>couple of
>>   days to sort this out.
>>
>>   In the mean while, if you want to use db_text for
>>permissions,
>>   please use the attached patch.
>>
>>   Thanks and regards,
>>   Bogdan
>>
>>
>>   Леонид Наседкин wrote:
>>
>>   Hi there.
>>   I'm trying to use permission module with db_text,
>>and it's not
>>   working, and I can't understand what's wrong.
>>   Opensips 1.6.1 svnrevision: 2:6509
>>
>>   In opensips.cfg:
>>   loadmodule "db_text.so"
>>   modparam("db_text", "db_mode", 0)
>>   loadmodule "permissions.so"
>>   modparam("permissions","db_url",
>>"text:///etc/opensips/dbtext")
>>
>>   In /etc/opensips/dbtext/address:
>>   id(int,auto) grp(int) ip(str) mask(int) port(int)
>>proto(str)
>>   pattern(str,null) context_info(str,null)
>>   10:1:10.100.0.0:23:5060:udp::
>>   20:1:10.110.0.0:23:5060:udp::
>>   30:1:10.120.0.0:23:5060:udp::
>>
>>   LOG:
>>
>>   DBG:core:init_mod: initializing module permissions
>>   DBG:permissions:mod_init: initializing...
>>   WARNING:permissions:parse_config_file: file not found:
>>   /etc/opensips/permissions.allow
>>   WARNING:permissions:mod_init: default allow file
>>   (/etc/opensips/permissions.allow) not found =>
>>empty rule set
>>   WARNING:permissions:parse_config_file: file not found:
>>   /etc/opensips/permissions.deny
>>   WARNING:permissions:mod_init: default deny file
>>   (/etc/opensips/permissions.deny) not found => empty
>>rule set
>>   DBG:core:find_mod_export: found  in module
>>   db_text [/usr/lib/opensips/modules/]
>>   DBG:core:db_bind_mod: using db bind api for db_text
>>   INFO:db_text:dbt_init: using database at:
>>/etc/opensips/dbtext/
>>   DBG:db_text:dbt_cache_get_db: looking for db
>>   /etc/opensips/dbtext/!
>>   DBG:db_text:dbt_cache_get_db: new db!
>>   DBG:db_text:dbt_load_file: request for table [version]
>>   DBG:db_text:dbt_load_file: db is
>>[/etc/opensips/dbtext/]
>>   DBG:db_text:dbt_load_file: loading file
>>  

Re: [OpenSIPS-Users] How to force opensips add port field in via header?

2010-02-02 Thread Bogdan-Andrei Iancu
Hi Iñaki,

actually there is no param to force adding port to the local VIA when 
the port is the SIP default one (just checked the code).

Lei, if it really required to get such a via (even if not conforming to 
RFC), I can send you a small patch.

Regards,
Bogdan

Iñaki Baz Castillo wrote:
> El Martes, 2 de Febrero de 2010, Lei Tang escribió:
>   
>>   Hi everyone, I'm using opensips 1.6.
>>   When I send sip msg by call forward function, opensips add Via header in
>> sips message like "Via: SIP/2.0/UDP
>> 10.37.143.6;branch=z9hG4bKSS1vtZ6ZQ98rp."
>>THE QUESTION IS: How can I make opensips add the port field to the VIA
>> header even my sip port is 5060? This mean, the VIA header should be
>>   "Via: SIP/2.0/UDP 10.37.143.6*:5060*;branch=z9hG4bKSS1vtZ6ZQ98rp."
>>Does someone can give me some advice about this question? Thank a lot!
>>
>> 
>
> There is a core parameter for that, I can not remember now the name but you 
> can find it in the wiki (cookbook section).
>
>   


-- 
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www.voice-system.ro


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Re: [OpenSIPS-Users] How to force opensips add port field in via header?

2010-02-02 Thread Iñaki Baz Castillo
El Martes, 2 de Febrero de 2010, Bogdan-Andrei Iancu escribió:
> Hi Iñaki,
> 
> actually there is no param to force adding port to the local VIA when
> the port is the SIP default one (just checked the code).

Sure? what about "advertised_port=5060"? :)

Some years ago I used it in openser 1.2 to avoid a stupid bug in a stupid 
gateway (Nortel CS2K, the worst machine in the world).



-- 
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Re: [OpenSIPS-Users] high-availability - senario

2010-02-02 Thread Julien Chavanton
I deceided not to use a loadbalancer because, the IP address for outbound call 
would have been different and we want it to be completely transparent for 
existing Interco.
 
I went with heartbeat + mon 
I created sip.monitor
 
When the fail over take effect it is not listening on the virtual IP, how do I 
configure opensips to bind to 0.0.0.0:5060 ?



From: users-boun...@lists.opensips.org on behalf of Nigel Daniels
Sent: Tue 26/01/2010 11:45 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] high-availability - senario



have you considerd using keepalived with vrrp instead ?


On Tue, Jan 26, 2010 at 7:03 AM, Julien Chavanton  wrote:


I am configuring high-availability with heart-beat + LVS + ldirector
 
I do not want to load balance but mostly make sure it will fail over 
quickly, I have found situation where the in LVS (UDP/TCP) connection never 
timeout when the remote IP send periodic OPTIONS request for example.
 
I beleive I will have to set very low UDP/TCP time-out, however  with 
such a low time-out I can not load-balance so I will use weighted Round-Robin 
with very high priority 65535 on the active one and weith 1 on the passive 
server.
 
Any other suggested way to cluster without load-balancing ?
 

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-- 
Nigel Daniels
Network & Systems Administrator 
ConnectAndSell inc.
(650)-533-2542 


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Re: [OpenSIPS-Users] One way audio - media port changed (opensips / mediaproxy)

2010-02-02 Thread Magnus Burman
Hi,

I did a wireshark export, 2461 lines, so it's in the pastebin:

http://pastebin.ca/1775584

syslog below:

11.11.X.Y - Trunk subnet
22.22.X.Y - Proxy subnet
98400 - UA DID
X.voip.url.es - UA sip subscriber

Jan 11 22:28:07 sbc1 media-relay[4335]: debug: Got traffic information for
stream: (audio) 11.11.6.144:40518 (RTP: Unknown, RTCP: Unknown) <->
22.22.224.9:50358 <-> 22.22.224.9:58928 <-> Unknown (RTP: 22.22.198.159:5206,
RTCP: Unknown)
Jan 11 22:28:07 sbc1 media-relay[4335]: debug: Got traffic information for
stream: (audio) 11.11.6.144:40518 (RTP: Unknown, RTCP: Unknown) <->
22.22.224.9:50358 <-> 22.22.224.9:58928 <-> Unknown (RTP: 22.22.198.159:5206,
RTCP: 22.22.198.159:5207)
Jan 11 22:28:07 sbc1 media-relay[4335]: debug: Got initial answer from
callee for stream: (audio) 11.11.6.144:40518 (RTP: Unknown, RTCP: Unknown)
<-> 22.22.224.9:50358 <-> 22.22.224.9:58928 <-> 22.22.198.159:5206 (RTP:
22.22.198.159:5206, RTCP: 22.22.198.159:5207)
Jan 11 22:28:07 sbc1 media-relay[4335]: debug: Got traffic information for
stream: (audio) 11.11.6.144:40518 (RTP: 11.11.6.144:40518, RTCP: Unknown)
<-> 22.22.224.9:50358 <-> 22.22.224.9:58928 <-> 22.22.198.159:5206 (RTP:
22.22.198.159:5206, RTCP: 22.22.198.159:5207)
Jan 11 22:29:26 sbc1 media-relay[4335]: debug: Got traffic information for
stream: (audio) 11.11.6.144:40518 (RTP: 11.11.6.144:40518, RTCP: Unknown)
<-> 22.22.224.9:50358 <-> 22.22.224.9:58928 <-> 22.22.198.159:5206 (RTP:
22.22.198.159:5206, RTCP: 22.22.198.159:5207)
Jan 11 22:54:20 sbc1 media-dispatcher[4324]: debug: Got statistics:
{'from_tag': 'e-13c4-10b392b-75e19db4-10b392b', 'dialog_id':
'1305:480665684', 'start_time': 1263245287.181, 'timed_out': False,
'call_id': 'a27f94c89e3e13c410b392b13d753bdb84e00e2147f91b8-0266-6714',
'to_tag': 'ICF_197662497-514', 'streams': [{'status': 'closed',
'caller_codec': 'G711a', 'post_dial_delay': 10.76835417749,
'callee_codec': 'G711a', 'start_time': 0, 'caller_bytes': 15733000,
'callee_bytes': 15328600, 'caller_packets': 78665, 'end_time': 1573,
'callee_remote': '22.22.198.159:5206', 'caller_remote': '11.11.6.144:40518',
'media_type': 'audio', 'callee_local': '22.22.224.9:58928', 'timeout_wait':
0, 'caller_local': '22.22.224.9:50358', 'callee_packets': 76643}, {'status':
'rejected', 'caller_codec': 'Unknown', 'post_dial_delay': None,
'callee_codec': 'Unknown', 'start_time': 0, 'caller_bytes': 0,
'callee_bytes': 0, 'caller_packets': 0, 'end_time': 0, 'callee_remote':
'Unknown', 'caller_remote': 'Unknown', 'media_type': 'image',
'callee_local': '22.22.224.9:59676', 'timeout_wait': 0, 'caller_local': '
22.22.224.9:58930', 'callee_packets': 0}], 'duration': 1573, 'to_uri': '
984000...@22.22.224.9:5060', 'from_uri': '34963555...@11.11.0.144:5060',
'callee_ua': 'unknown agent', 'caller_ua': 'CS2000_NGSS/9.0'}

Cheers,
Magnus

2010/2/2 Saúl Ibarra Corretgé 

> Hi,
>
> El 02/02/10 11:54, Magnus Burman escribió:
> > Hi guys,
> >
> > I've got a problem with "sporadic" one-way audio, using latest stable
> > Opensips 1.4 and Mediaproxy 2.
> >
> > The problem occurs at re-invites, thou not every time. When it does
> > happen, the media port is changed towards the callee. Example:
> >
> > First invite:
> > INVITE Trunk:40518 --> Proxy:50358
> > INVITE Proxy:58928 --> UA:5206
> >
> > Re-invite:
> > INVITE Trunk:40518 --> Proxy:50358
> > INVITE Proxy:40518 --> UA:5206
> >
> > Notice how the Proxy port on the UA side (callee local) changed to the
> > same port that Trunk uses (caller remote).
> >
> > One way audio now occurs and naturally the call is soon hung up.
> > Mediaproxy now dumps some useful statistics showing:
> >
> > 'callee_remote': 'UA:5206'
> > 'caller_remote': 'TRUNK:40518'
> > 'callee_local': 'PROXY:58928'
> > 'caller_local': 'PROXY:50358'
> >
> > Notice here how the callee_local is the original port used.
> >
> > Where should I start looking to be able to solve this problem? Is it
> > most likely my config, something weird with the re-invite or have I
> > stumbled upon a bug?
> >
> > Thankful for every and all suggestions, as I lay sleepless at night
> > thinking about this! :-P
> >
>
> It would be nice to have a full SIP trace and the syslog output of
> MediaProxy to check if it's doing something wrong. Also, which function
> are you using for starting it?
>
>
>
> Regards,
>
>
> --
> Saúl Ibarra Corretgé
> AG Projects
>
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Re: [OpenSIPS-Users] One way audio - media port changed (opensips / mediaproxy)

2010-02-02 Thread Saúl Ibarra Corretgé
El 02/02/10 13:59, Magnus Burman escribió:
> Hi,
>
> I did a wireshark export, 2461 lines, so it's in the pastebin:
>
> http://pastebin.ca/1775584
>

It's really hard to follow a text wireshark cap, could you do a ngrep 
capture? ngrep -d any -W byline -T -P '' port 5060


-- 
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Re: [OpenSIPS-Users] Config Suggestions Request (SIP Trunks)

2010-02-02 Thread Jeff Pyle
Mike,

I have at least one customer on sipXecs.  It registers just fine.  Granted, 
there is a bunch of other stuff messed up, but I think that's more my 
customer's misunderstanding of how to configure it.


- Jeff


On Feb 2, 2010, at 1:51 AM, Mike O'Connor wrote:

> Hi All
> 
> I've a have a couple of customers who are all asking to use sipXecs,
> which from my investigations does not support registrations. Instead it
> excepts that the ITSP provide unauthenticated trunks for inbound and
> outbound calls.
> 
> So my question is what is the recommended way of supporting this ?
> 
> Thanks
> Mike
> 
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Re: [OpenSIPS-Users] One way audio - media port changed (opensips / mediaproxy)

2010-02-02 Thread Magnus Burman
I'll start capturing like that right away, less you want the pcap-file for
that specific call? Since the problem only occurs sporadically and at
re-invites (ie 20+ minutes into a call) it's a bit of a mess to track down
calls that fit the criteria. But I'll get right on it for sure, thanks.

Cheers,
Magnus

2010/2/2 Saúl Ibarra Corretgé 

> El 02/02/10 13:59, Magnus Burman escribió:
> > Hi,
> >
> > I did a wireshark export, 2461 lines, so it's in the pastebin:
> >
> > http://pastebin.ca/1775584
> >
>
> It's really hard to follow a text wireshark cap, could you do a ngrep
> capture? ngrep -d any -W byline -T -P '' port 5060
>
>
> --
> Saúl Ibarra Corretgé
> AG Projects
>
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Re: [OpenSIPS-Users] One way audio - media port changed (opensips / mediaproxy)

2010-02-02 Thread Iñaki Baz Castillo
El Martes, 2 de Febrero de 2010, Magnus Burman escribió:
> I'll start capturing like that right away, less you want the pcap-file for
> that specific call? Since the problem only occurs sporadically and at
> re-invites (ie 20+ minutes into a call) it's a bit of a mess to track down
> calls that fit the criteria. But I'll get right on it for sure, thanks.

Try this to filter the captured data:

  http://dev.sipdoc.net/projects/sip-stuff/wiki/Ngrep-SIP


-- 
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[OpenSIPS-Users] Opensips - using database views

2010-02-02 Thread Oleg Burlacu
Hi!
I'm planning to migrate from fixed database tables to views.
But I have a touble to generate the auto increment field - "id".
Can opensips operate with tables without the "id" filed?

Thanks,
Oleg
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[OpenSIPS-Users] Opensips - using database views

2010-02-02 Thread Oleg Burlacu
Hi!
I'm planning to migrate from fixed database tables to views.
But I have a touble to generate the auto increment field - "id".
Can opensips operate with tables without the "id" filed?

Thanks,
Oleg
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Re: [OpenSIPS-Users] high-availability - senario

2010-02-02 Thread Julien Chavanton
This was well documented
 
in opensips.cfg
 
listen=udp:0.0.0.0:5060
 
or for both TCP/UDP
 
listen=0.0.0.0:5060



From: users-boun...@lists.opensips.org on behalf of Julien Chavanton
Sent: Tue 02/02/2010 12:47 PM
To: OpenSIPS users mailling list; OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] high-availability - senario


I deceided not to use a loadbalancer because, the IP address for outbound call 
would have been different and we want it to be completely transparent for 
existing Interco.
 
I went with heartbeat + mon 
I created sip.monitor
 
When the fail over take effect it is not listening on the virtual IP, how do I 
configure opensips to bind to 0.0.0.0:5060 ?



From: users-boun...@lists.opensips.org on behalf of Nigel Daniels
Sent: Tue 26/01/2010 11:45 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] high-availability - senario



have you considerd using keepalived with vrrp instead ?


On Tue, Jan 26, 2010 at 7:03 AM, Julien Chavanton  wrote:


I am configuring high-availability with heart-beat + LVS + ldirector
 
I do not want to load balance but mostly make sure it will fail over 
quickly, I have found situation where the in LVS (UDP/TCP) connection never 
timeout when the remote IP send periodic OPTIONS request for example.
 
I beleive I will have to set very low UDP/TCP time-out, however  with 
such a low time-out I can not load-balance so I will use weighted Round-Robin 
with very high priority 65535 on the active one and weith 1 on the passive 
server.
 
Any other suggested way to cluster without load-balancing ?
 

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[OpenSIPS-Users] Bulding OpenSIPS v1.6.1-notls on 64 bit machine

2010-02-02 Thread Steven C. Blair

Hello,

 I'm trying to build a fresh copy of OpenSIPS v1.6.1-notls on a 64 bit RH 
system and experiencing some path issues when running opensipsdbctl create. It 
seems some scripts exist in /usr/local/lib64/opensips/ and some in 
/usr/local/src/apps/opensips-1.6.1-notls/scripts/. Is there a suggested way to 
resolve this different so the database build script will work?

Thanks
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Re: [OpenSIPS-Users] Opensips - using database views

2010-02-02 Thread Bogdan-Andrei Iancu
Hi Oleg,

most of the "id" columns are not used by opensips (neither written, nor 
read) - but they are used as primary keys.

So, you can remove the columns, but be sure in setting another primary 
key for the tables.

Regards,
Bogdan

Oleg Burlacu wrote:
> Hi!
> I'm planning to migrate from fixed database tables to views.
> But I have a touble to generate the auto increment field - "id".
> Can opensips operate with tables without the "id" filed?
>
> Thanks,
> Oleg
> 
>
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>   


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Re: [OpenSIPS-Users] high-availability - senario

2010-02-02 Thread Julien Chavanton
I still have a problem with binding all IP some headers do not have the IP set :
 
Record-Route: .
Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bKfc58.1427b764.0.
 
Is there a way to tell Opensip to use the routing source IP, or other 
alternative ?
 
 



From: users-boun...@lists.opensips.org on behalf of Julien Chavanton
Sent: Tue 02/02/2010 1:44 PM
To: OpenSIPS users mailling list; OpenSIPS users mailling list; OpenSIPS users 
mailling list
Subject: Re: [OpenSIPS-Users] high-availability - senario


This was well documented
 
in opensips.cfg
 
listen=udp:0.0.0.0:5060
 
or for both TCP/UDP
 
listen=0.0.0.0:5060



From: users-boun...@lists.opensips.org on behalf of Julien Chavanton
Sent: Tue 02/02/2010 12:47 PM
To: OpenSIPS users mailling list; OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] high-availability - senario


I deceided not to use a loadbalancer because, the IP address for outbound call 
would have been different and we want it to be completely transparent for 
existing Interco.
 
I went with heartbeat + mon 
I created sip.monitor
 
When the fail over take effect it is not listening on the virtual IP, how do I 
configure opensips to bind to 0.0.0.0:5060 ?



From: users-boun...@lists.opensips.org on behalf of Nigel Daniels
Sent: Tue 26/01/2010 11:45 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] high-availability - senario



have you considerd using keepalived with vrrp instead ?


On Tue, Jan 26, 2010 at 7:03 AM, Julien Chavanton  wrote:


I am configuring high-availability with heart-beat + LVS + ldirector
 
I do not want to load balance but mostly make sure it will fail over 
quickly, I have found situation where the in LVS (UDP/TCP) connection never 
timeout when the remote IP send periodic OPTIONS request for example.
 
I beleive I will have to set very low UDP/TCP time-out, however  with 
such a low time-out I can not load-balance so I will use weighted Round-Robin 
with very high priority 65535 on the active one and weith 1 on the passive 
server.
 
Any other suggested way to cluster without load-balancing ?
 

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Re: [OpenSIPS-Users] Opensips - using database views

2010-02-02 Thread Oleg Burlacu
Hi Bogdan,

Let's suppose that we have 2 tables, one for domains and other for users but
with the domain field the foreign key from domains table. Also I have in the
users table the bit type fields "local", "mobile"...
In this case our output from the created view "grp" will be

username  domaingrp last_modified
xxyy10sip1.abcd.efginternational1900-01-01 00:00:01
xxyy10sip1.abcd.efglocal1900-01-01 00:00:01
xxyy10sip1.abcd.efgmobile1900-01-01 00:00:01
xxyy10sip1.abcd.efgnational1900-01-01 00:00:01

So, each table has a primary key, but the "result" - not. Can I use this
result for group module? Or, I should generate also a primary key?

Best regards,
Oleg


On Tue, Feb 2, 2010 at 4:13 PM, Bogdan-Andrei Iancu
wrote:

> Hi Oleg,
>
> most of the "id" columns are not used by opensips (neither written, nor
> read) - but they are used as primary keys.
>
> So, you can remove the columns, but be sure in setting another primary
> key for the tables.
>
> Regards,
> Bogdan
>
> Oleg Burlacu wrote:
> > Hi!
> > I'm planning to migrate from fixed database tables to views.
> > But I have a touble to generate the auto increment field - "id".
> > Can opensips operate with tables without the "id" filed?
> >
> > Thanks,
> > Oleg
> > 
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
>
>
> --
> Bogdan-Andrei Iancu
> www.voice-system.ro
>
>
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Re: [OpenSIPS-Users] high-availability - senario

2010-02-02 Thread Stanisław Pitucha
On 02.02.2010 14:17, Julien Chavanton wrote:
> I still have a problem with binding all IP some headers do not have the IP 
> set :
>  
> Record-Route: .
> Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bKfc58.1427b764.0.
>  
> Is there a way to tell Opensip to use the routing source IP, or other 
> alternative ?

There are two options, (never used the first one though - it might work):

1) advertised_address="1.2.3.4" (your real ip)

2) Set ip_nonlocal_bind via sysctl, which will allow you to bind to an
interface even if it's not available on that host. You can have a hot
standby host this way (on the virtual address), because it will start
getting messages as soon as you update the routing. Just bind to the
address you want instead of 0.0.0.0.

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Re: [OpenSIPS-Users] priority differences between carrierroute and drouting

2010-02-02 Thread Gabriel Bermudez
Seems that a weighted load balancing is not directly posible with
drouting (http://lists.opensips.org/pipermail/users/2009-September/008106.html),
but you can trick the system to behave like this. It would be great if
the module could also support this feature natively, maybe by handling
a special format on the gwlist colum (GW1=0.4,GW2=0.2,GW3=0.4)

I was interested on use drouting, because it can be managed with
opensips control panel.  Is the carrierroute module on plans?

2010/2/1 Gabriel Bermudez 
>
> Hi everybody,
>
> I've read the documentation on carrierroute and drouting, about the algorithm 
> used to select a gateway.  I understand that on carrierroute the gateway 
> selected depends on the prefix dialed and the value of the prob field.  For 
> example for prefix 49, gateway x.x.x.x with prob=0.2 and gateway y.y.y.y with 
> prob=0.8 can exists in the carrierroute table.
>
> ++-+---+-+---++---+
> | id | carrier | domain | scan_prefix | flags | prob | rewrite_host  |
> ++-+---+-+---++---+
> | 1  |   1 |  0  | 49          | 0 |  0.2   | x.x.x.x     
>     |
> | 2  |   1 |  0  | 49          | 0 |  0.8   | 
> y.y.y.y     |
> ++-+---+-+---++---+
>
> I understand that this means 20% of the calls will be sent to x.x.x.x and 80% 
> to y.y.y.y
> Is this the same for drouting, or only the highest priority rule is selected?
>
> Thanks for your answers
>
> Regards,

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[OpenSIPS-Users] dns name and media_relay_avp.

2010-02-02 Thread Leonid Nasedkin
Hi there.
Is it posible to use dns-name at media_relay_avp?
Now I just get error:
media-dispatcher[356]: warning: user requested media_relay
relay-01.test.local is not available

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[OpenSIPS-Users] lb_reload issue

2010-02-02 Thread rajib deka
Hello all,

Is it possible to execute lb_reload command from remote machine using some
java/c/c# program. Or any other way to achieve this. Pls help me out.

-- Thanks
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Software Engineer
Servion Global Solution
Chennai, India

Mobile No: + 91 80157 09130
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Re: [OpenSIPS-Users] lb_reload issue

2010-02-02 Thread Iñaki Baz Castillo
El Martes, 2 de Febrero de 2010, rajib deka escribió:
> Hello all,
> 
> Is it possible to execute lb_reload command from remote machine using some
> java/c/c# program. Or any other way to achieve this. Pls help me out.

Sure, use the Manager Interface remotely by using the mi_xmlrpc or mi_datagram 
modules.

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[OpenSIPS-Users] RTP listener

2010-02-02 Thread Yannick LE COENT
Hello all,

 

I would to listen and record RTP streams in real-time.

 

RTP proxy seems to be able to record RTP streams in pcap format.

 

Is there a way to listen RTP streams in real-time?

 

Thanks for any help,

Yannick

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Re: [OpenSIPS-Users] Bulding OpenSIPS v1.6.1-notls on 64 bit machine

2010-02-02 Thread Andrew Pogrebennyk
On 02.02.2010 16:10, Steven C. Blair wrote:
>   I'm trying to build a fresh copy of OpenSIPS v1.6.1-notls on a 64 bit RH 
> system and experiencing some path issues when running opensipsdbctl create. 
> It seems some scripts exist in /usr/local/lib64/opensips/ and some in 
> /usr/local/src/apps/opensips-1.6.1-notls/scripts/. Is there a suggested way 
> to resolve this different so the database build script will work?

Steven,
You should provide the exact console output that shows what happens. 
Personally I and many people here are using 1.6.1 on 64 bit machines.

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Re: [OpenSIPS-Users] What is protocol/port mismatch?

2010-02-02 Thread opensipslist

Hello list,

The problem was that the voicemail server was respecting the host
and port of the routeset which OpenSIPS was rewriting to acommodate
the transport change from TLS to TCP. The voicemail server was
configured to send over TLS to the outbound proxy however, so
traffic was arriving at OpenSIPS on the port 5060 using TLS.

It's clear now why the warnings were appearing in the logs, but
even more it seems that OpenSIPS sometimes crashes when sending
TLS traffic to a TCP port which it is expecting clear text from.

To know about the solution to this problem, read below.

An ven., janv 29, 2010, opensipsl...@encambio.com schrieb:
>An ven., janv 29, 2010, Bogdan-Andrei Iancu schrieb:
>>opensipsl...@encambio.com wrote:
>>> ...and I see this in the log:
>>>
>>>opensips[2717]: WARNING:core:get_send_socket: protocol/port 
>>> mismatch
>>>
>>> some hundreds of times per day (about once every 20 minutes
>>> per registered UAC.)
>>>
>>> I have this in the route script:
>>>
>>>   listen = tls:name.host.tld:5061
>>>   [...]
>>>   t_relay("name.host.tld:5080");
>>>
First, although all UAs were exchanging SIP traffic with OpenSIPS
over TLS, t_relay was relaying the messages over UDP. I assume that
t_relay is either hardcoded to use UDP or OpenSIPS was routing the
messages to itself internally over UDP (to recursively resolve
aliases.) In this case maybe t_relay implicitly uses the transport
over which the message was last received (even internally.)

In any case, the call to t_relay should look like this if relaying
over the TCP transport is desired:

  t_relay("tcp:name.host.tld:5080");

The problem then is that the rr module will add two Record-Route
headers, expecting a symetrical fashion of SIP traffic. I turned
this off with modparam("rr", "enable_double_rr", 0) because the
voicemail server (Asterisk) is very tricky (maybe even buggy) to
configure to use TLS and outbound proxies.

At this point I saw that traffic was arriving at OpenSIPS port 5061
over TLS as expected, and as indicated by the routeset. The problem
is solved.

>Is the basic idea of this warning message that OpenSIPS
>exchanges a SIP message with a UA over a certain transport
>(TLS in this case) and port number (5061 in this case), but
>a t_relay in the route script forwards the message over a
>different transport or to a different port number?
>
The basic idea of this warning message is that OpenSIPS expects
a certain type (TLS or not TLS) of traffic at a certain TCP port
as configured by the 'listen' directives. If a UAC sends TLS
traffic to a port only configured to listen to plain text traffic
then this warning will appear. What happens to the mismatched
traffic is not clear, but I assume that it is ignored by OpenSIPS.

>If so, what is being compared and how is it compared?
>
The TCP port on which OpenSIPS listens to and the type of traffic,
either TLS encrypted or plain text.

Regards,
Brian

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Re: [OpenSIPS-Users] number of opensips children

2010-02-02 Thread opensipslist

Hello Bogdan,

An mer., déc  23, 2009, Bogdan-Andrei Iancu schrieb:
>opens...@encambio.com wrote:
>> An ven., déc 18, 2009, Bogdan-Andrei Iancu schrieb:
>>> To see what processes you have and what they are doing, do:
>>>
>>> opensipsctl fifo ps
>>>
>>   # opensipsctl fifo ps
>>   Process::  ID=0 PID=24975 Type=attendant
>>   Process::  ID=1 PID=24977 Type=SIP receiver udp:123.234.210.1:5060
>>   Process::  ID=2 PID=24978 Type=time_keeper
>>   Process::  ID=3 PID=24979 Type=timer
>>   Process::  ID=4 PID=24980 Type=MI FIFO
>>
>> [...]
>>
>> My gut feeling is that having four UDP listening processes and four
>> TCP listening processes is about right for us, because we only have
>> a handful of UACs participating infrequently (5 calls per day.)
>>   
>Actually that is more than needed - during some performance tests (only 
>simply call relaying) we managed to put 6K cps in a single process.
>
I have eight TCP listeners configured and about sixteen UACs are
connected. I get a ton of these warnings whenever REGISTER or INVITE
messages come in:

  Feb 02 18:17:22 name.host.tld  opensips[02126]: 
WARNING:core:send2child: no free tcp receiver, connection passed to the 
leastbusy one (1)
  Feb 02 18:17:25 name.host.tld  opensips[02126]: 
WARNING:core:send2child: no free tcp receiver, connection passed to the 
leastbusy one (1)

Because you mentioned that you benchmarked 6K CPS with a single
process (was it TCP?), I'd like to know if you got as many warnings
as well. One question is:

  What does 'free tcp receiver' mean? I assumed that listening
  TCP ports were free to accept as many connections as needed.

By the way, each of the 16 UACs registered to the 8 TCP listener
processes is avoiding NAT problems by keeping the TCP connection
open by setting the tcp_persistent_flag.

Is OpenSIPS expecting there to be at least one TCP listener process
which is not encumbered by the tcp_persistent_flag?

Regards,
Brian

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[OpenSIPS-Users] Quota system problem

2010-02-02 Thread Pedersen R.
Hi all,

I installed Opensips 1.6 + CDRTool version 6.9 and I'm not able to setup
quota for postpaid users.I added the *quota* column in
opensips.subscriber table. And after what else to setup?
What is the change on opensips.cfg ?
Is there any complete howto (tutorial) for quota/postpaid system?
Can anyone help me??


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Re: [OpenSIPS-Users] number of opensips children

2010-02-02 Thread opensipslist

Hello,

An mar., févr 02, 2010, opensipsl...@encambio.com schrieb:
>An mer., déc  23, 2009, Bogdan-Andrei Iancu schrieb:
>>opens...@encambio.com wrote:
>>> An ven., déc 18, 2009, Bogdan-Andrei Iancu schrieb:
>>> My gut feeling is that having four UDP listening processes and four
>>> TCP listening processes is about right for us, because we only have
>>> a handful of UACs participating infrequently (5 calls per day.)
>>>   
>>Actually that is more than needed - during some performance tests (only 
>>simply call relaying) we managed to put 6K cps in a single process.
>>
>I have eight TCP listeners configured and about sixteen UACs are
>connected. I get a ton of these warnings whenever REGISTER or INVITE
>messages come in:
>
>  Feb 02 18:17:22 name.host.tld  opensips[02126]: 
> WARNING:core:send2child: no free tcp receiver, connection passed to the 
> leastbusy one (1)
>  Feb 02 18:17:25 name.host.tld  opensips[02126]: 
> WARNING:core:send2child: no free tcp receiver, connection passed to the 
> leastbusy one (1)
>
>Because you mentioned that you benchmarked 6K CPS with a single
>process (was it TCP?), I'd like to know if you got as many warnings
>as well. One question is:
>
>  What does 'free tcp receiver' mean? I assumed that listening
>  TCP ports were free to accept as many connections as needed.
>
> [...]
>
>Is OpenSIPS expecting there to be at least one TCP listener process
>which is not encumbered by the tcp_persistent_flag?
>
At risk of answering my own question and questioning my own answer,
I'd like to suggest the following change:

--- tcp_main.c.orig 2010-01-18 12:33:49.151095000 +0100
+++ tcp_main.c  2010-02-02 20:07:15.263065567 +0100
@@ -911,7 +911,7 @@
tcp_children[idx].busy++;
tcp_children[idx].n_reqs++;
if (min_busy){
-   LM_WARN("no free tcp receiver, connection passed to the least"
+   LM_INFO("no free tcp receiver, connection passed to the least "
"busy one (%d)\n", min_busy);
}
LM_DBG("to tcp child %d %d(%d), %p\n", idx, tcp_children[idx].proc_no,

That would correct the defective english spelling 'leastbusy' as
well as ridding the log of a properly running OpenSIPS server of
false warnings. I'm assuming of course, that it's perfectly okay
for TCP listener processes to keep a TCP connection open by using
the tcp_persistent_flag and accept new SIP requests at the same
time.

Regards,
Brian

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Re: [OpenSIPS-Users] lb_reload issue

2010-02-02 Thread rajib deka
Thank you very much for your reply. I think this is what I was searching
for.
Let you know once I successfully apply it.

Cheers
Rajib

On Tue, Feb 2, 2010 at 10:00 PM, Iñaki Baz Castillo  wrote:

> El Martes, 2 de Febrero de 2010, rajib deka escribió:
> > Hello all,
> >
> > Is it possible to execute lb_reload command from remote machine using
> some
> > java/c/c# program. Or any other way to achieve this. Pls help me out.
>
> Sure, use the Manager Interface remotely by using the mi_xmlrpc or
> mi_datagram
> modules.
>
> --
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>
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Re: [OpenSIPS-Users] Quota system problem

2010-02-02 Thread osiris123d

Haven't worked with Quota yet, but perhaps this will get you started down the
right path

Look at the "bye_on_timeout_flag" modparam in Dialog module
http://www.opensips.org/html/docs/modules/devel/dialog.html#bye-on-timeout-flag-id


Heard about the "bye_on_timeout_flag" on this post
http://lists.opensips.org/pipermail/users/2008-October/001116.html
-- 
View this message in context: 
http://n2.nabble.com/Quota-system-problem-tp4502804p4503572.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.

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Re: [OpenSIPS-Users] Config Suggestions Request (SIP Trunks)

2010-02-02 Thread Mike O'Connor
I've never once been able to get sipXecs to send out a register packet,
the are very clear in the documentation that they expect a sip trunk. ie
static configuration.

Mike



On 2/02/10 11:44 PM, Jeff Pyle wrote:
> Mike,
>
> I have at least one customer on sipXecs.  It registers just fine.  Granted, 
> there is a bunch of other stuff messed up, but I think that's more my 
> customer's misunderstanding of how to configure it.
>
>
> - Jeff
>
>
> On Feb 2, 2010, at 1:51 AM, Mike O'Connor wrote:
>
>   
>> Hi All
>>
>> I've a have a couple of customers who are all asking to use sipXecs,
>> which from my investigations does not support registrations. Instead it
>> excepts that the ITSP provide unauthenticated trunks for inbound and
>> outbound calls.
>>
>> So my question is what is the recommended way of supporting this ?
>>
>> Thanks
>> Mike
>>
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>
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Re: [OpenSIPS-Users] How to force opensips add port field in via header?

2010-02-02 Thread Lei Tang
Hi Bogdan and Iñaki, Thank you every much.
Bogdan, Could you send me the patch?

2010/2/2 Iñaki Baz Castillo 

> El Martes, 2 de Febrero de 2010, Bogdan-Andrei Iancu escribió:
> > Hi Iñaki,
> >
> > actually there is no param to force adding port to the local VIA when
> > the port is the SIP default one (just checked the code).
>
> Sure? what about "advertised_port=5060"? :)
>
> Some years ago I used it in openser 1.2 to avoid a stupid bug in a stupid
> gateway (Nortel CS2K, the worst machine in the world).
>
>
>
> --
> Iñaki Baz Castillo 
>
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Re: [OpenSIPS-Users] Starting OpenXCAP without any logs

2010-02-02 Thread CheeWii
Havn't resloved the former problem,I got into the new trouble. The
errors shows as follows,could you give me some suggestions? Is it the fault
of python??

r...@ubuntu:/var/log/openxcap# /usr/bin/openxcap --no-fork
Starting OpenXCAP 1.1.2
xcap.server.HTTPFactory starting on 80
error: Traceback (most recent call last):
error:   File "/usr/lib/python2.5/threading.py", line 462, in __bootstrap
error: self.__bootstrap_inner()
error:   File "/usr/lib/python2.5/threading.py", line 486, in
__bootstrap_inner
error: self.run()
error:   File "/usr/lib/python2.5/threading.py", line 446, in run
error: self.__target(*self.__args, **self.__kwargs)
error: ---  ---
error:   File
"/usr/lib/python2.5/site-packages/twisted/python/threadpool.py", line 210,
in _worker
error: result = context.call(ctx, function, *args, **kwargs)
error:   File "/usr/lib/python2.5/site-packages/twisted/python/context.py",
line 59, in callWithContext
error: return self.currentContext().callWithContext(ctx, func, *args,
**kw)
error:   File "/usr/lib/python2.5/site-packages/twisted/python/context.py",
line 37, in callWithContext
error: return func(*args,**kw)
error:   File
"/usr/lib/python2.5/site-packages/twisted/enterprise/adbapi.py", line 426,
in _runInteraction
error: conn = self.connectionFactory(self)
error:   File
"/usr/lib/python2.5/site-packages/twisted/enterprise/adbapi.py", line 38, in
__init__
error: self.reconnect()
error:   File
"/usr/lib/python2.5/site-packages/twisted/enterprise/adbapi.py", line 75, in
reconnect
error: self._connection = self._pool.connect()
error:   File
"/usr/lib/python2.5/site-packages/twisted/enterprise/adbapi.py", line 395,
in connect
error: conn = self.dbapi.connect(*self.connargs, **self.connkw)


>
> Yes. As I see you're running a hybrid system, because twisted version in
> Lenny is 8.1.0 and you're running 9.0.0. This shouldn't matter, however
> I need to test OpenXCAP with latest Debian unstable to check if changes
> are needed because of updated library versions, I'll keep you posted.
>
>
> Regards,
>
> --
>  Saúl Ibarra Corretgé
> AG Projects
>
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Re: [OpenSIPS-Users] How to force opensips add port field in via header?

2010-02-02 Thread Lei Tang
Hi Bogdan,
I tried to modify msg_translator.c, change line 1933 from

  if ((send_sock->port_no!=SIP_PORT) || (port_str!=&send_sock->port_no_str))
to
  if (1)

It seem work now.


2010/2/3 Lei Tang 

> Hi Bogdan and Iñaki, Thank you every much.
> Bogdan, Could you send me the patch?
>
> 2010/2/2 Iñaki Baz Castillo 
>
> El Martes, 2 de Febrero de 2010, Bogdan-Andrei Iancu escribió:
>> > Hi Iñaki,
>> >
>> > actually there is no param to force adding port to the local VIA when
>> > the port is the SIP default one (just checked the code).
>>
>> Sure? what about "advertised_port=5060"? :)
>>
>> Some years ago I used it in openser 1.2 to avoid a stupid bug in a stupid
>> gateway (Nortel CS2K, the worst machine in the world).
>>
>>
>>
>> --
>> Iñaki Baz Castillo 
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
>
>
> --
> Lei.Tang
> lei.tl...@gmail.com
>



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Re: [OpenSIPS-Users] Config Suggestions Request (SIP Trunks)

2010-02-02 Thread David J.
Pardon my ignorance, but what exactly is the problem?

Your customer does not need to register to you, however, then you need 
to have a static ip address to route to.
and vice versa. When a customer sends an INVITE to your opensips, you 
need to authenticate by IP address.

The sipXecs does not need to regsiter.

Your customer will also have to enable ip authentication from you as well.

Again, pardon my ignorance.



Mike O'Connor wrote:
> I've never once been able to get sipXecs to send out a register packet,
> the are very clear in the documentation that they expect a sip trunk. ie
> static configuration.
>
> Mike
>
>
>
> On 2/02/10 11:44 PM, Jeff Pyle wrote:
>   
>> Mike,
>>
>> I have at least one customer on sipXecs.  It registers just fine.  Granted, 
>> there is a bunch of other stuff messed up, but I think that's more my 
>> customer's misunderstanding of how to configure it.
>>
>>
>> - Jeff
>>
>>
>> On Feb 2, 2010, at 1:51 AM, Mike O'Connor wrote:
>>
>>   
>> 
>>> Hi All
>>>
>>> I've a have a couple of customers who are all asking to use sipXecs,
>>> which from my investigations does not support registrations. Instead it
>>> excepts that the ITSP provide unauthenticated trunks for inbound and
>>> outbound calls.
>>>
>>> So my question is what is the recommended way of supporting this ?
>>>
>>> Thanks
>>> Mike
>>>
>>> ___
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>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>> 
>>>   
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>> 
>
>
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[OpenSIPS-Users] MediaProxy as bridge between Private and Public Interfaces

2010-02-02 Thread Daniel Worrad
Hi All,

 

I have MediaProxy working in a multi-homed setup where it is acting as a
relay between an interface on a public IP and one on a private IP (We
connect to our SIP provider over a private network via the private
interface using static routes in the route table, and a SNAT out the
private interface)

 

The system is functioning and calls are being relayed between public and
private interfaces, however with every call, I am receiving the
following in the logs (which is a bit unnerving):

 

Feb  3 15:19:32 SIPProxy1 media-relay[6806]: Traceback (most recent call
last):

Feb  3 15:19:32 SIPProxy1 media-relay[6806]:   File
"/usr/local/lib/python2.5/site-packages/twisted/internet/udp.py", line
126, in doRead

Feb  3 15:19:32 SIPProxy1 media-relay[6806]:
self.protocol.datagramReceived(data, addr)

Feb  3 15:19:32 SIPProxy1 media-relay[6806]:   File
"/usr/local/lib/python2.5/site-packages/mediaproxy/mediacontrol.py",
line 127, in datagramReceived

Feb  3 15:19:32 SIPProxy1 media-relay[6806]: self.cb_func(host,
port, data)

Feb  3 15:19:32 SIPProxy1 media-relay[6806]:   File
"/usr/local/lib/python2.5/site-packages/mediaproxy/mediacontrol.py",
line 203, in got_data

Feb  3 15:19:32 SIPProxy1 media-relay[6806]:
self.substream.check_create_conntrack()

Feb  3 15:19:32 SIPProxy1 media-relay[6806]:   File
"/usr/local/lib/python2.5/site-packages/mediaproxy/mediacontrol.py",
line 253, in check_create_conntrack

Feb  3 15:19:32 SIPProxy1 media-relay[6806]: self.forwarding_rule =
_conntrack.ForwardingRule(self.caller.remote, self.caller.local,
self.callee.remote, self.callee.local, self.stream.session.mark)

Feb  3 15:19:32 SIPProxy1 media-relay[6806]: Error: No such file or
directory

 

I have tried strace to determine what file may be missing or lacking
permissions, however there are references to hundreds of files and it is
proving difficult to track down.

 

Has anyone seen this before, or could suggest some other way of
troubleshooting the issue?

 

Many thanks,

 

Daniel Worrad

Ipera Communications

 

 

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Re: [OpenSIPS-Users] Config Suggestions Request (SIP Trunks)

2010-02-02 Thread Mike O'Connor
Authenticate by IP Address, yep but how ?

What is the recommend way of handling this and how do I route DID's to them.

That was my question :)
Mike

On 3/02/10 3:50 PM, David J. wrote:
> Pardon my ignorance, but what exactly is the problem?
>
> Your customer does not need to register to you, however, then you need 
> to have a static ip address to route to.
> and vice versa. When a customer sends an INVITE to your opensips, you 
> need to authenticate by IP address.
>
> The sipXecs does not need to regsiter.
>
> Your customer will also have to enable ip authentication from you as well.
>
> Again, pardon my ignorance.
>
>
>
> Mike O'Connor wrote:
>   
>> I've never once been able to get sipXecs to send out a register packet,
>> the are very clear in the documentation that they expect a sip trunk. ie
>> static configuration.
>>
>> Mike
>>
>>
>>
>> On 2/02/10 11:44 PM, Jeff Pyle wrote:
>>   
>> 
>>> Mike,
>>>
>>> I have at least one customer on sipXecs.  It registers just fine.  Granted, 
>>> there is a bunch of other stuff messed up, but I think that's more my 
>>> customer's misunderstanding of how to configure it.
>>>
>>>
>>> - Jeff
>>>
>>>
>>> On Feb 2, 2010, at 1:51 AM, Mike O'Connor wrote:
>>>
>>>   
>>> 
>>>   
 Hi All

 I've a have a couple of customers who are all asking to use sipXecs,
 which from my investigations does not support registrations. Instead it
 excepts that the ITSP provide unauthenticated trunks for inbound and
 outbound calls.

 So my question is what is the recommended way of supporting this ?

 Thanks
 Mike

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 Users mailing list
 Users@lists.opensips.org
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>>> 
>>>   
>>
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>> 
>
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