Re: [SR-Users] Forward UAC Registration with Authentication (like Upper Registration or Passthrough Registration)

2014-07-15 Thread Angelica Juarez


Anjelicajuarez desde mi iPhone

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[SR-Users] siremis/ser/xrcmds error help

2014-07-15 Thread MARY
sr-users,hello:
 
http://127.0.0.1:8890/siremis/ser/xrcmds 
error info:
[2014-07-14 08:49:30 (GMT)] An exception occurred while executing this script:
Error message: #2, require_once(XML/RPC.php): failed to open stream: No such 
file or directory
Script name and line number of error: 
//siremis-4.1.0/siremis/modules/ser/service/asipto/libs/cmds/serxr.php:2


why?
can not find the XML/RPC.php file。




2014-07-1417:56:16



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[SR-Users] have problem with iptel account

2014-07-15 Thread Samhu


hi,

This is Sam from Draytek Crop.
We have a IPPBX device with iptel for SIP account as below.


User B dial to IPPBX could heart the answer from Auto Attendant of IPPBX.
Then user B cannot heart anything (after input the extension of user A) 
when user A pickup his telephone .

At this momemt, user A could hear everything from user B.

This Network Architecture worked fine before.
And it still works fine now if we select the account of draytel.org.
Please show us how to solve the problem.


  Best Regards,

  Sam Hu





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Re: [SR-Users] siremis/ser/xrcmds error help

2014-07-15 Thread Daniel-Constantin Mierla

Hello,

you need to install XML_RPC package for php -- it can be done using pear 
tool.


Cheers,
Daniel

On 14/07/14 11:58, MARY wrote:

sr-users,hello:
 
http://127.0.0.1:8890/siremis/ser/xrcmds
error info:
[2014-07-14 08:49:30 (GMT)] An exception occurred while executing this 
script:
Error message: #2, require_once(XML/RPC.php): failed to open stream: 
No such file or directory
Script name and line number of error: 
//siremis-4.1.0/siremis/modules/ser/service/asipto/libs/cmds/serxr.php:2

why?
can not find the XML/RPC.php file。


2014-07-1417:56:16




MARY



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[SR-Users] please help with Msilo config

2014-07-15 Thread Miguel Rios
Hi list,

I'm a newbie when it comes to kamailio, although I have a fair amount of SIP 
experience.
I'm trying to setup a very basic kamailio install (4.1 on Wheezy) with Msilo 
support.

I just used the default kamailio.cfg file (changing obviously the relevant 
parameters for my setup) and have not touched the routing blocks.
I'm very confused about how routing works, and I don't have a background in C. 
I've read the wiki up and down and have looked at inumerous tutorials spread 
out on the internet (most of which just add to my confusion because they seem 
out of date), but I still don't get it.


I managed to setup a working kamailio server where local users can call each 
other fine. Now when I try to add the msilo module, and specially the routing 
example from MSILO Module is when things get tricky.

I'm sure my error is a fairly basic one and has to do with the routing logic 
and syntax. Could some kind soul please share a copy of the whole ### 
Routing Logic  on downwards for a basic no frills kamailio with msilo 
setup?

Thanks,
Miguel

  
          
MSILO Module
3. Parameters 3.1. db_url (string) Database URL. Default value is  
“mysql://kamailio:kamailiorw@localhost/kamailio”. Example 1.1. Set the “db_url” 
parameter   
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Re: [SR-Users] Routing calls to Asterisk using dispatch

2014-07-15 Thread Olli Heiskanen
A little progress on this; the double-AOR problem was fixed; I had
callbackextension field in the realtime table, which caused extra REGISTER
messages to be sent from Asterisk to Kamailio and that messed the AORs for
my clients.

Calling between the clients is a problem though, the INVITE gets routed
correctly but for some reason, when the INVITE is returning from Asterisk
to be routed to the called party, location lookup fails. I end up with 404
and no route to destination error. Any clues on to why location lookup
would fail?

kamctl ul show gives correct looking output for both registered clients.

cheers,
Olli







2014-07-12 17:27 GMT+03:00 Olli Heiskanen ohjelmistoarkkite...@gmail.com:


 Hello,

 I've started playing with an idea to add multiple asterisk servers and
 using dispatcher to balance the sip load between them. I added the code
 according to dispatcher module documentation (
 http://www.kamailio.org/docs/modules/4.2.x/modules/dispatcher.html), but
 I think there's something off in my setup:

 kamctl ul output shows 2 AORs for one client:

 AOR:: 7...@testers.com
 Contact:: sip:770@2.2.2.2:64340;rinstance=c634da314e12385f;transport=UDP
 Q=
  Expires:: 3221
 Callid:: ZTE1MWYwYzM3NGNjNjMxMmEzM2JjYWNmNzQyZTdiNGI.
 Cseq:: 2
  User-agent:: Z 3.2.21357 r21367
 State:: CS_SYNC
 Flags:: 0
  Cflag:: 0
 Socket:: udp:1.1.1.1:5060
 Methods:: 5087
  Ruid:: uloc-53bfe447-35ae-2a2
 Reg-Id:: 0
 Last-Keepalive:: 1405174150
  Last-Modified:: 1405174150
 AOR:: 770@1.1.1.1
 Contact:: sip:770@1.1.1.1:5070 Q=
  Expires:: 68
 Callid:: 327fcf07641f80006e962821112a6...@testers.com
  Cseq:: 754
 User-agent:: Asterisk PBX 11.10.2
 State:: CS_SYNC
  Flags:: 0
 Cflag:: 0
 Socket:: udp:1.1.1.1:5060
  Methods:: 4294967295
 Ruid:: uloc-53bfe447-35af-a82
 Reg-Id:: 0
  Last-Keepalive:: 1405174477
 Last-Modified:: 1405174477


 I don't think I should be seeing an AOR for 770 where Contact is the
 public address of my server (here 1.1.1.1) and User-Agent which is
 Asterisk.

 I'm using Asterisk Realtime integration, and by what I can tell the sip
 messages are going nicely, client authenticates with Kamailio and sends
 this message to Asterisk (which is on the same machine; Kamailio at 5060
 and Asterisk at 5070):

 1.1.1.1.sip  1.1.1.1.vtsas: SIP, length: 374
 REGISTER sip:1.1.1.1:5070 SIP/2.0
 Via: SIP/2.0/UDP
 1.1.1.1;branch=z9hG4bKbc8a.f4473947.0
 To: sip:770@1.1.1.1
 From: sip:770@1.1.1.1;tag=4a9c3f1c98b9f1c5704acfd1770d93d2-d0c1
 CSeq: 10 REGISTER
 Call-ID: 7ffa0191-13742@1.1.1.1
 Max-Forwards: 70
 Content-Length: 0
 Contact: sip:770@1.1.1.1:5060
 Expires: 3600


 Currently I can make calls from 770 to 123 which is an Asterisk extension
 that answers, plays hello world and hangs up. However I can't call another
 sip clients when I route calls through Asterisk, they do work fine if I
 don't use Asterisk for handling calls, but I'd like Kamailio to be in the
 role of proxy/loadbalancer and Asterisk to handle calls.

 My config is the simple default config, added with realtime stuff and then
 dispatcher according to the documentation. I wonder if there's something
 wrong in the REGISTER that Kamailio sends to Asterisk, or maybe something
 else going wrong?

 Has anyone seen results like this and do you spot something here that
 needs fixing?

 Thanks,
 Olli





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Re: [SR-Users] set_rtp_proxy_set behaviour

2014-07-15 Thread Igor Potjevlesch
Hi,

Just to complete my findings, I think that I got a working config with these
instructions:
modparam(rtpproxy, rtpproxy_sock, udp:127.0.0.1:7722) - keep the
default RTProxy for all the others use-cases.

modparam(rtpproxy, rtpproxy_sock, 2 == udp:127.0.0.1:7723) - add a
second instance into a set numbered 2 for example and call him when it's
necessary into the config file with set_rtp_proxy_set(2).

I have to do more tests but it looks that when set_rtp_proxy_set is not
called, there is only the first rtpproxy that is used.

Regards,

Igor.

-Message d'origine-
De : Igor Potjevlesch [mailto:igor.potjevle...@gmail.com] 
Envoyé : jeudi 10 juillet 2014 11:01
À : 'Kamailio (SER) - Users Mailing List'
Objet : RE: [SR-Users] set_rtp_proxy_set behaviour

Hi Alex,

Thank you for your answer.
But I'm not sure to understand.

If I have only one case with set_rtp_proxy_set, are the other calls of the
rtpproxy_manage()  (or similar) will be load-balanced between the two sets?
Is it possible to avoid load-balance and be sure that outside my
set_rtp_proxy_set(2), the first one will be used?

Regards,

Igor.

-Message d'origine-
De : sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] De la part de Alex Balashov
Envoyé : mardi 8 juillet 2014 19:55 À : sr-users@lists.sip-router.org
Objet : Re: [SR-Users] set_rtp_proxy_set behaviour

Hi Igor,


On 07/08/2014 01:44 PM, Igor Potjevlesch wrote:

 Hello,

 I set two entries of rtpproxy like :

 modparam(rtpproxy, rtpproxy_sock, 1 == udp:127.0.0.1:7722)

 modparam(rtpproxy, rtpproxy_sock, 2 == udp:127.0.0.1:7723)

 For a particular case, I want to use the second RTPProxy and the first 
 one as the default one.

 If in a conditional block I set set_rtp_proxy(“2”), all other call to 
 rtpproxy_manage will use the first RTPProxy? Or there is a 
 load-balancing introduce by default.

Load-balancing by default is within a set. Since you have two different sets
defined, forcing one or the other with set_rtp_proxy_set() will result in
only one of those proxies being used.

--
Alex Balashov - Principal
Evariste Systems LLC
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

Please be kind to the English language:

http://www.entrepreneur.com/article/232906

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Re: [SR-Users] Many instances of new URI shorter than old URI in log

2014-07-15 Thread Alex Villací­s Lasso

El 02/07/14 15:02, Alex Villací­s Lasso escribió:

El 02/07/14 10:51, Alex Villací­s Lasso escribió:

El 30/06/14 04:24, Daniel-Constantin Mierla escribió:

Hello,

iirc, this is a check when trying to convert from base64 to old uri, but I 
didn't have time to dig properly inside the sources.

I enhanced the log message to print the values of the uris, via commit:

- 
http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=6feeb886b58f4cd1cefeef132245c009bf8ca07f

If you have a sip trace for a message throwing the error, I can look at it. If 
not, try the patch from the above commit and send again the logs when they 
appear.

Jul  1 18:26:33 mail /usr/sbin/kamailio[32188]: ERROR: uac [replace.c:591]: 
restore_uri(): new URI [] shorter than old URI 
[sip:hgaibor@192.168.2.67:5060;transport=UDP]
Jul  1 18:26:33 mail /usr/sbin/kamailio[32189]: ERROR: uac [replace.c:591]: 
restore_uri(): new URI [] shorter than old URI 
[sip:hgaibor@192.168.2.67:5060;transport=UDP]
Jul  1 19:06:11 mail /usr/sbin/kamailio[32185]: ERROR: uac [replace.c:591]: 
restore_uri(): new URI [] shorter than old URI 
[sip:hgaibor@192.168.2.67:5060;transport=UDP]
Jul  1 19:06:11 mail /usr/sbin/kamailio[32185]: ERROR: uac [replace.c:591]: 
restore_uri(): new URI [] shorter than old URI 
[sip:hgaibor@192.168.2.67:5060;transport=UDP]
Jul  2 09:53:14 mail /usr/sbin/kamailio[32189]: ERROR: uac [replace.c:591]: 
restore_uri(): new URI [] shorter than old URI 
[sip:hgaibor@192.168.2.67:5060;transport=UDP]
Jul  2 09:53:15 mail /usr/sbin/kamailio[32188]: ERROR: uac [replace.c:591]: 
restore_uri(): new URI [] shorter than old URI 
[sip:hgaibor@192.168.2.67:5060;transport=UDP]
Jul  2 10:47:25 mail /usr/sbin/kamailio[32185]: ERROR: uac [replace.c:591]: 
restore_uri(): new URI [] shorter than old URI 
[sip:hgaibor@192.168.2.67:5060;transport=UDP]

More examples:

Jul  2 12:32:27 mail /usr/sbin/kamailio[3111]: ERROR: uac [replace.c:591]: 
restore_uri(): new URI [] shorter than old URI 
[sip:91632087@192.168.0.105:51526;alias=186.70.154.18~51526~1]
Jul  2 12:32:27 mail /usr/sbin/kamailio[3112]: ERROR: uac [replace.c:591]: 
restore_uri(): new URI [] shorter than old URI 
[sip:91632087@192.168.0.105:51526;alias=186.70.154.18~51526~1]
Jul  2 12:32:27 mail /usr/sbin/kamailio[3110]: ERROR: uac [replace.c:591]: 
restore_uri(): new URI [] shorter than old URI 
[sip:91632087@192.168.0.105:51526;alias=186.70.154.18~51526~1]
Jul  2 12:32:27 mail /usr/sbin/kamailio[3111]: ERROR: uac [replace.c:591]: 
restore_uri(): new URI [] shorter than old URI 
[sip:91632087@192.168.0.105:51526;alias=186.70.154.18~51526~1]
Jul  2 12:32:28 mail /usr/sbin/kamailio[3113]: ERROR: uac [replace.c:591]: 
restore_uri(): new URI [] shorter than old URI 
[sip:91632087@192.168.0.105:51526;alias=186.70.154.18~51526~1]
Jul  2 12:32:30 mail /usr/sbin/kamailio[3110]: ERROR: uac [replace.c:591]: 
restore_uri(): new URI [] shorter than old URI 
[sip:91632087@192.168.0.105:51526;alias=186.70.154.18~51526~1]
Jul  2 12:32:33 mail /usr/sbin/kamailio[3110]: ERROR: uac [replace.c:591]: 
restore_uri(): new URI [] shorter than old URI 
[sip:91632087@192.168.0.105:51526;alias=186.70.154.18~51526~1]


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Any comments on this?

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Re: [SR-Users] function to reset all flags or bflags?

2014-07-15 Thread Juha Heinanen
Alex Hermann writes:

 $mf is an R/W variable, you could assign 0 to it. Maybe $bf is R/W also, but 
 it isn't documented as such.

I tested, and, yes, it is r/w variable.  I added note about it to pv
wiki page.

-- Juha

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Re: [SR-Users] please help with Msilo config

2014-07-15 Thread Peter Villeneuve
Hi Miguel,

Welcome. Unfortunately I'm still fairly new to kamailio too, and the
learning curve is pretty steep as you've no doubt discovered.
I'm also having problems with getiing msilo running properly, and I'm sure
it's just bad routing on my part, so please share if you do happen to get
the routing down.
All the examples I found on the internet were dated or incomplete, so
hopefully someone can share a recent example of a working routing block
from start to finish.

Cheers,
Peter


On Tue, Jul 15, 2014 at 2:58 PM, Miguel Rios miguelrio...@yahoo.com wrote:

 Hi list,

 I'm a newbie when it comes to kamailio, although I have a fair amount of
 SIP experience.
 I'm trying to setup a very basic kamailio install (4.1 on Wheezy) with
 Msilo support.

 I just used the default kamailio.cfg file (changing obviously the relevant
 parameters for my setup) and have not touched the routing blocks.
 I'm very confused about how routing works, and I don't have a background
 in C. I've read the wiki up and down and have looked at inumerous tutorials
 spread out on the internet (most of which just add to my confusion because
 they seem out of date), but I still don't get it.

 I managed to setup a working kamailio server where local users can call
 each other fine. Now when I try to add the msilo module, and specially the
 routing example from MSILO Module
 http://kamailio.org/docs/modules/4.1.x/modules/msilo.html is when
 things get tricky.

 I'm sure my error is a fairly basic one and has to do with the routing
 logic and syntax. Could some kind soul please share a copy of the whole
 ### Routing Logic  on downwards for a basic no frills kamailio
 with msilo setup?

 Thanks,
 Miguel






 MSILO Module http://kamailio.org/docs/modules/4.1.x/modules/msilo.html
 3. Parameters 3.1. db_url (string) Database URL. Default value is
 “mysql://kamailio:kamailiorw@localhost/kamailio”. Example 1.1. Set the
 “db_url” parameter
 View on kamailio.org
 http://kamailio.org/docs/modules/4.1.x/modules/msilo.html
 Preview by Yahoo



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[SR-Users] http_query results question

2014-07-15 Thread Jonathan Hunter
Hi Guys,
I have used the http_query function in the past, in conjunction with the json 
module, on version 3.3 and unbuntu.
I am now using version 4.1 on Centos 6.5, and I am using the http_query, which 
I'm aware records in the results the first line of the response.
Can you select the result to be recorded depending on the line in the response, 
for example if a number is returned only in line 4, can this be extracted from 
a 200 ok response using this function?
Or is this then a need to use JSON module, and if so which repository for 
centos 6.5 contains it! :)
Sorry for the question!
Thanks
Jon

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Re: [SR-Users] SCA and PRESENCE

2014-07-15 Thread Jorj Bauer
Forgive me if this is daft - but what is your need for the Presence module?

It sounds like you're trying to enable two different technologies that do the 
same thing, since SCA already handles busy-lamps. I presume I'm just missing 
something.

-- Jorj



On Jul 14, 2014, at 3:57 PM, Joel White joelewh...@gmail.com wrote:

 I am enabling what was in the default config for WITH_PRESENCE 
 
 When I enable WITH_PRESENCE, the Polycom phones do not show BLF
 
 Sure, I am using SPIP 335, SPIP 601 and 650, as well as VVX 1500 and SPIP 5000
 
 For the 601 I am running Legacy firmware   3.1.8 combined with a Bootrom of 
 3.1.3 rev D
 
 For the 335, 650, and 5000 I am running firmware version  3.2.6 Combined .  I 
 believe the Bootrom is 4.2.0
 
 I have not started to work on auto provisioning with the VVX 1500 and have 
 been manually configuring them.  I believe that they will work with the 3.2.6 
 firmware as well although I have seen many using 4.0.3 firmware with the 
 newer VVX models
 
 
 
 Ask me anything and I will try to help :)
 
 
 On Thu, Jul 10, 2014 at 7:28 AM, João Vitor Arruda joao.arr...@gmail.com 
 wrote:
 Hi Joel,
 
 When you say: I enable presence modules what you mean? 
 Other than loading the module itself you may have included some code in 
 kamailio.cfg using to handle the SUBSCRIBE messages for presence right?
 
 Without knowing your config, one thing that crossed my mind is:
  - Since both SCA and PRESENCE modules need to handle SUBSCRIBE messages you 
 need to be sure that you do it first using the SCA function 
 sca_handle_subscribe() just for the ones with Event header == call-info 
 or == line-seize. Then, for all the other SUBSCRIBE messages you can use 
 the PRESENCE module function.
 
 There is an example of the code for filtering the ones to be handled by the 
 SCA module function here:
 http://www.kamailio.org/docs/modules/devel/modules/sca.html#idp27944
 
 You too mention that the SCA works fine on your Polycom phones, since I´m too 
 trying to use it but having some issues, I will appreciate if you can share 
 (either here or by pvt message) some info like: which Polycom phone models 
 and firmware version you are using.
 
 Regards,
 João Vitor Arruda
 
 
 2014-07-08 13:26 GMT-03:00 Joel White joelewh...@gmail.com:
 I am having an issue where I enable presence modules and the SCA module stops 
 working.  I disable presence and SCA works fine on Polycom phones.  Does 
 anyone know what I am missing or doing wrong?
 
 
 Thank you in advance
 
 Joel
 
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[SR-Users] http_query results question

2014-07-15 Thread Juha Heinanen
Jonathan Hunter writes:

 I am now using version 4.1 on Centos 6.5, and I am using the http_query, 
 which I'm aware records in the results the first line of the response.
 Can you select the result to be recorded depending on the line in the
 response, for example if a number is returned only in line 4, can this
 be extracted from a 200 ok response using this function?

that kind of functionality is not currently supported by http_query.

-- juha

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Re: [SR-Users] http_query results question

2014-07-15 Thread Jonathan Hunter
Hi Juha,
Thank you for the response.
So I presume there isn't any other option for this functionality at present?
Thanks
Jon

 Date: Tue, 15 Jul 2014 20:24:25 +0300
 To: sr-users@lists.sip-router.org
 From: j...@tutpro.com
 Subject: [SR-Users]  http_query results question
 
 Jonathan Hunter writes:
 
  I am now using version 4.1 on Centos 6.5, and I am using the http_query, 
  which I'm aware records in the results the first line of the response.
  Can you select the result to be recorded depending on the line in the
  response, for example if a number is returned only in line 4, can this
  be extracted from a 200 ok response using this function?
 
 that kind of functionality is not currently supported by http_query.
 
 -- juha
 
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Re: [SR-Users] http_query results question

2014-07-15 Thread Daniel-Constantin Mierla

Hello,

the patch for utils module might be trivial for such extension, however, 
my way to deal with http client needs was to use an embedded lua script. 
It was always fast and I had all the options I needed for setting the 
body of the request and handling the response. Lua has http lib with 
support for https as well. This can be an option for the moment. Perhaps 
other embedded interpreters can be used if you are more familiar with 
any of the alternatives (e.g., perl, python, C# ...).


Cheers,
Daniel

On 15/07/14 22:03, Jonathan Hunter wrote:

Hi Juha,

Thank you for the response.

So I presume there isn't any other option for this functionality at 
present?


Thanks

Jon

 Date: Tue, 15 Jul 2014 20:24:25 +0300
 To: sr-users@lists.sip-router.org
 From: j...@tutpro.com
 Subject: [SR-Users] http_query results question

 Jonathan Hunter writes:

  I am now using version 4.1 on Centos 6.5, and I am using the 
http_query, which I'm aware records in the results the first line of 
the response.

  Can you select the result to be recorded depending on the line in the
  response, for example if a number is returned only in line 4, can this
  be extracted from a 200 ok response using this function?

 that kind of functionality is not currently supported by http_query.

 -- juha

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--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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