[SR-Users] Upgrade to version 4.1

2014-05-20 Thread VOIP Tests
Hello, I understand that there are some bugs in ver 4.03. Can someone let
me know how I can update my running version of Kamailio ( 4.03 ) to version
4.1? Will I need to do a fresh  install of the new version or is there a
way I can do an upgrade/update to ver 4.03?

Thank you,
Arun
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Re: [SR-Users] Upgrade to version 4.1

2014-05-20 Thread VOIP Tests
Thanks Daniel.


On Tue, May 20, 2014 at 9:28 AM, Daniel Tryba dan...@pocos.nl wrote:

 On Tuesday 20 May 2014 15:35:15 VOIP Tests wrote:
  Hello, I understand that there are some bugs in ver 4.03. Can someone let
  me know how I can update my running version of Kamailio ( 4.03 ) to
 version
  4.1? Will I need to do a fresh  install of the new version or is there a
  way I can do an upgrade/update to ver 4.03?

 Seen the migration guide:

 http://www.kamailio.org/wiki/install/upgrade/4.0.x-to-4.1.0

 You need to update some sql tables if you use them, upgrading myiasm to
 innodb
 is optional. Can't remember running into any problems.

 --

 POCOS B.V. - Croy 9c - 5653 LC Eindhoven
 Telefoon: 040 293 8661 - Fax: 040 293 8658
 http://www.pocos.nl/   - http://www.sipo.nl/
 K.v.K. Eindhoven 17097024

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Re: [SR-Users] Realtime integration: Unregistered clients showing as registered?

2014-05-18 Thread VOIP Tests
Try updating your /etc/hosts file with the domain 'testers.com'.

Arun


On Sun, May 18, 2014 at 5:06 AM, Olli Heiskanen 
ohjelmistoarkkite...@gmail.com wrote:

 Hello,

 It took me a while to get forward on this, but I had progress. I've
 changed my sip.conf back and forth so I can't name the exact cause for my
 problem, but it may have been the fact that in my asterisk sippeers table
 the fields permit and deny may have been in the wrong order. And/or some
 configuration values in sip.conf.

 So now clients can register and asterisk 'sip show peer' shows the
 registered clients.

 However, there is still one thing that's probably not quite there yet. I'm
 using the domain 'testers.com' for my clients, but I can't register them
 using that domain. I was able to get clients to register and visible to
 asterisk only by using domain '127.0.0.1', if I try commenting that out,
 asterisk will say:
 chan_sip.c:28073 handle_request_register: Registration from '
 sip:660@127.0.0.1' failed for '1.1.1.1:5060' - Not a local domain
 (where 1.1.1.1 is the public ip of the asterisk+kamailio box)

 In my sip.conf I have domains defined like this:
 autodomain=no
 domain=127.0.0.1
 domain=testers.com

 I think this may be the cause for this behavior:
 In my kamailio.cfg I have asterisk and kamailio bindips defined like this:
 asterisk.bindip = 127.0.0.1 desc Asterisk IP Address
 asterisk.bindport = 5070 desc Asterisk Port
 kamailio.bindip = 127.0.0.1 desc Kamailio IP Address
 kamailio.bindport = 5060 desc Kamailio Port

 And this route forwards REGISTER messages to asterisk using the 127.0.0.1
 as domain:

 route[REGFWD] {
 if(!is_method(REGISTER))
 {
 return;
 }
 $var(rip) = $sel(cfg_get.asterisk.bindip);
 $uac_req(method)=REGISTER;

 $uac_req(ruri)=sip: + $var(rip) + : +
 $sel(cfg_get.asterisk.bindport);
 $uac_req(furi)=sip: + $au + @ + $var(rip);
 $uac_req(turi)=sip: + $au + @ + $var(rip);
 $uac_req(hdrs)=Contact: sip: + $au + @
 + $sel(cfg_get.kamailio.bindip)
 + : + $sel(cfg_get.kamailio.bindport) +
 \r\n;

 if($sel(contact.expires) != $null)
 $uac_req(hdrs)= $uac_req(hdrs) + Expires:  +
 $sel(contact.expires) + \r\n;
 else
 $uac_req(hdrs)= $uac_req(hdrs) + Expires:  +
 $hdr(Expires) + \r\n;

 uac_req_send();
 }


 So question is, what would be the good-practice way to fix my setup into
 using the client's domain? I thought about using the domain 'testers.com'
 in place of kamailio.bindip but was unable to build the sip message and
 send it to kamailio ip. uac_req_send() seems to send the message to what is
 defined in the request line of the message so replacing it with '
 testers.com' would not work.

 cheers,
 Olli






 2014-04-23 17:31 GMT+03:00 Pedro Niño nino.pe...@gmail.com:

 Ahhh also, don't forget to place the *rtcachefriends*=*yes* in your
 sip.conf (asterisk) to show the realtime peers
 El abr 23, 2014 8:29 AM, Olli Heiskanen ohjelmistoarkkite...@gmail.com
 escribió:

  Hello,

 Gracias Pedro, kiitos Mikko.

 It's good to know I have configured Kamailio correctly. I added the type
 into my table but so far no luck having asterisk see the clients
 registered, at least on cli. I do see that asterisk adds registration data
 into the table. I'll work on this for a bit and ask in the asterisk list on
 more tricks on asterisk side. I'll post back here if I find out what the
 problem was, in case someone is having similar issues.

 Thanks again,
 Olli



 2014-04-22 21:06 GMT+03:00 Pedro Niño nino.pe...@gmail.com:

 Don't forget to include peer type (friend), and The callbacknumber In
 The table.

 It happened to me and asterisk/kamailio behavior was wayyy to weird
 until made sure both parameters were there.

 -

 In this setup I have SIP peers in an asterisk table added like this:

 INSERT INTO sippeers (name, defaultuser, host, sippasswd, fromuser,
 fromdomain) VALUES ('660', '660', 'dynamic', 'password', '660', '
 testers.com');

 --
  El abr 19, 2014 1:17 PM, Olli Heiskanen 
 ohjelmistoarkkite...@gmail.com escribió:


 Hello,

 One of the tests I've been working with is Asterisk realtime
 integration according to Daniel's guide here:
 http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb

 Weird thing is the client looks registered but I'm not sure if it
 really is registered. If I'm not mistaken I should see the peers when I
 issue 'sip show peers' on asterisk cli. Instead I get this:

 *CLI sip show peers
 Name/username  Host  Dyn Forcerport Comedia  ACL Port
  Status  Description  Realtime
 0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0
 offline]


 Also, calling between clients will fail; in Asterisk cli I get:
 *CLI
   == Using SIP RTP TOS bits 184
   == Using SIP RTP CoS mark 5
 -- Executing [661@default:1] 

[SR-Users] Duplicate values in SDP

2014-05-15 Thread VOIP Tests
Hello,

Can some one let me know the reason that there are these duplicate values
in the SDP ( o, c, m, nortpproxy)? Our server was working with kamailio and
asterisk on the same machine and had no problem. When we separated kamailio
and asterisk on different servers and added the dispatcher module I see
this error because of which there is not audio from an incoming PSTN call.


2014/05/15 18:55:19.972332 172.10.30.8:5060 - 66.136.17.30:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 66.136.17.30:5060
;rport=5060;branch=z9hG4bK1sansay1225318698rdb6244.
Record-Route: sip:54.108.18.75;lr=on;ftag=sansay1225318698rdb6244.
Record-Route: sip:sansay1225318698rdb6244@66.136.17.30:5060
;lr;transport=udp.
From: sip:xx85342@66.136.17.30;tag=sansay1225318698rdb6244.
To: sip:12142349395@54.108.18.75;tag=as3414811d.
Call-ID: 537507353-0-1838374200@64.136.174.226.
CSeq: 1 INVITE.
Server: Asterisk PBX 1.8.17.0.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH.
Supported: replaces, timer.
Session-Expires: 1800;refresher=uac.
Contact: sip:xxx9395@172.10.30.5:5080.
Content-Type: application/sdp.
Content-Length: 385.
.
v=0.
o=root 1739301191 1739301191 IN IP4 54.108.18.7554.108.18.7554.108.18.75.
s=Asterisk PBX 1.8.17.0.
c=IN IP4 54.108.18.7554.108.18.7554.108.18.75.
t=0 0.
m=audio 388503885038850 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.
a=nortpproxy:yes.
a=nortpproxy:yes.
a=nortpproxy:yes.


Thank you for the help.

Arun
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[SR-Users] Starting Kamailio with init script

2014-05-08 Thread VOIP Tests
Hello, can someone let me know how I can start Kamailio with the init
script? In the /etc/default/kamailio file I have commented
out RUN_KAMAILIO=yes, this starts  Kamailio but my phones do not register.
If I stop the init script and start kamailio with kamctl then my phones
register, what am I doing wrong here?

Also, if my understanding is correct, if I start Kamailio with the init
script I should not start kamailio with kamctl, in that case how do I run
the kamctl commands, for eg like kamctl ul show ?

Thanks for the help.

Arun
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Re: [SR-Users] Starting Kamailio with init script

2014-05-08 Thread VOIP Tests
Moritz, thank your for your pointer. I changed some parementer in the
/etc/init.d/kamailio file and now it seems fine. Can you please let me know
if this output looks ok.

Output of 'ps aux | grep kamailio' with kamctl start:

*root  1498  0.6  0.2 255716  7980 ?S15:29   0:00
/usr/local/sbin/kamailio -P /var/run/kamailio.pid -f
/usr/local/etc/kamailio//kamailio.cfg*

Output of 'ps aux | grep kamailio' with init.d :
r*oot  1805  0.0  0.2 294816  8036 ?S15:44   0:00
/usr/local/sbin/kamailio -f /usr/local/etc/kamailio/kamaili .cfg -P
/var/run/kamailio/kamailio.pid -m 64 -M 4 -u root -g root*


Thank you,
Arun



On Thu, May 8, 2014 at 10:35 AM, Moritz Graf moritz.g...@g-fit.de wrote:

 Hi,

 start kamailio with initscript:
 /etc/init.d/kamailio start

 Different behaviour with different starting methods might be due to
 different configs that are used for starting.

 Start kamailio each way and do a 'ps aux | grep kamailio'. Then compare
 or post here.

 greetz

 Am 08.05.2014 14:46, schrieb VOIP Tests:
  Hello, can someone let me know how I can start Kamailio with the init
  script? In the /etc/default/kamailio file I have commented
  out RUN_KAMAILIO=yes, this starts  Kamailio but my phones do not
  register. If I stop the init script and start kamailio with kamctl then
  my phones register, what am I doing wrong here?
 
  Also, if my understanding is correct, if I start Kamailio with the init
  script I should not start kamailio with kamctl, in that case how do I
  run the kamctl commands, for eg like kamctl ul show ?
 
  Thanks for the help.
 
  Arun


 --

 Moritz Graf, M.Sc.
 Betrieb NGN-Plattform

 G-FIT GmbH  Co. KG
 Greflingerstr. 26, 93055 Regensburg

 Telefon +49 (9 41) 46299 - 1 86
 Telefax +49 (9 41) 46299 - 2 86
 mailto:moritz.g...@g-fit.de
 http://www.g-fit.de

 G-FIT Gesellschaft für innovative Telekommunikationsdienste mbH  Co.
 KG, Kommanditgesellschaft, Sitz Regensburg, Registergericht Regensburg,
 HRA 7626; Geschäftsführer: Dipl.Inf. (FH) Alfred Rauscher


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[SR-Users] Asterisk-Kamailio set up

2014-05-06 Thread VOIP Tests
Hello, I have a set up with Asterisk-Kamailio as explained in
http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb.

This set up has been working well for us for sometime now. We are now
moving Asterisk to  a stand alone server ( not installing it on the same
box as asterisk ) but we are running into some issues. I have just made
some changes to the Asterisk bind IP address and port and also set up
Kamalio as a peer on asterisk.

When I make an extension to extension call the calls are failing from the
FROMASTERISK route ( call is getting cancelled ). Can someone let me know
what changes I need to make to correct the issue? Some pointer will help.

Thank you,
Arun
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Re: [SR-Users] Asterisk-Kamailio set up

2014-05-06 Thread VOIP Tests
I was able to resolve the issue, I had the wrong IP in the hosts file.

Regards,
Arun


On Tue, May 6, 2014 at 10:18 PM, 5060 toaster...@gmail.com wrote:

 hi:
 you can check by kamctl online, maybe some users are not there due to bind
 ..has expired.


 2014-05-07 6:03 GMT+08:00 VOIP Tests kamailio...@gmail.com:

 Hello, I have a set up with Asterisk-Kamailio as explained in
 http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb
 .

 This set up has been working well for us for sometime now. We are now
 moving Asterisk to  a stand alone server ( not installing it on the same
 box as asterisk ) but we are running into some issues. I have just made
 some changes to the Asterisk bind IP address and port and also set up
 Kamalio as a peer on asterisk.

 When I make an extension to extension call the calls are failing from the
 FROMASTERISK route ( call is getting cancelled ). Can someone let me know
 what changes I need to make to correct the issue? Some pointer will help.

 Thank you,
 Arun

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Re: [SR-Users] Kamailio and mysql connection error: db_mysql [km_my_con.c:109]: driver error:

2014-05-04 Thread VOIP Tests
Have you granted DB access to Kamailio...?

GRANT ALL ON kamailio.* TO DB_USERNAME@'KAMAIALIO_IP' IDENTIFIED BY
'DBPASSWORD';


On Sun, May 4, 2014 at 2:32 AM, toasterisk toaster...@gmail.com wrote:

 hello, all of users:
 I am using this link for kamalio+asterisk+mysql:
 http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb
 i do everything based on the link, only different thing is I installed
 Kamalio+mysql at same server, and asterisk in another server.
 currently, my asterisk can connect with the remove MYSQL server, but
 kamailio can not connect with that.
 when start up the kamailio, i got the error message:
 --
 0(3739) DEBUG: core [sruid.c:100]: root for sruid is
 [uloc-5365e604-e9b-] (0 / 18)
  0(3739) DEBUG: core [sr_module.c:892]: DEBUG: init_mod_child (1): ctl
  0(3739) DEBUG: core [sr_module.c:892]: DEBUG: init_mod_child (1): mi_rpc
  0(3739) DEBUG: core [sr_module.c:892]: DEBUG: init_mod_child (1): acc
  0(3739) DEBUG: core [sr_module.c:892]: DEBUG: init_mod_child (1):
 auth_db
  0(3739) DEBUG: core [db.c:316]: connection 0xb71aab90 not found in pool
  0(3739) DEBUG: db_mysql [km_my_con.c:94]: opening connection:
 mysql://:@localhost/asterisk
  0(3739) ERROR: db_mysql [km_my_con.c:109]: driver error: Access denied
 for user 'asterisk'@'localhost' (using password: YES)
  0(3739) ERROR: core [db.c:320]: could not add connection to the pool
 0(3739) ERROR: auth_db [authdb_mod.c:174]: unable to connect to the database
  0(3739) ERROR: core [sr_module.c:896]: init_mod_child(): Error while
 initializing module auth_db (/usr/local/lib/kamailio/modules_k/auth_db.so)
  0(3739) ERROR: core [main.c:1433]: main_dontfork: init_child failed
  4(3744) INFO: core [main.c:851]: INFO: signal 15 received
  3(3743) INFO: core [main.c:851]: INFO: signal 15 received
  0(3739) INFO: core [main.c:797]: INFO: dont_fork turned on, living on
  0(3739) ALERT: core [main.c:782]: child process 3743 exited normally,
 status=0
  0(3739) ALERT: core [main.c:782]: child process 3744 exited normally,
 status=0
  0(3739) INFO: core [main.c:797]: INFO: dont_fork turned on, living on
  0(3739) DEBUG: core [db_pool.c:97]: connection still kept in the pool
  0(3739) DEBUG: tm [t_funcs.c:122]: DEBUG: tm_shutdown : start
  0(3739) DEBUG: tm [t_funcs.c:125]: DEBUG: tm_shutdown : emptying hash
 table
  0(3739) DEBUG: tm [t_funcs.c:127]: DEBUG: tm_shutdown : removing
 semaphores
  0(3739) DEBUG: tm [t_funcs.c:129]: DEBUG: tm_shutdown : destroying tmcb
 lists
  0(3739) DEBUG: tm [t_funcs.c:132]: DEBUG: tm_shutdown : done
  0(3739) DEBUG: core [mem/shm_mem.c:242]: shm_mem_destroy
  0(3739) DEBUG: core [mem/shm_mem.c:245]: destroying the shared memory
 lock

 ---
 [MySQL-asterisk]
 Description = MySQL Asterisk database
 Trace = Off
 TraceFile = stderr
 Driver = MySQL
 SERVER = 192.168.1.116  // IP for kamailio and mysql
 USER = asterisk
 PASSWORD = 11
 PORT = 3306
 DATABASE = asterisk
 -use isql test---
 root@kamailio:/etc# isql -v ODBC
 [IM002][unixODBC][Driver Manager]Data source name not found, and no
 default driver specified
 [ISQL]ERROR: Could not SQLConnect
 root@kamailio:/etc#
 ---
 anyone knows the problem?
 thanks!


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Re: [SR-Users] Location table not getting updated

2014-05-04 Thread VOIP Tests
Once I reboot the phone the location table gets populated? Any idea how to
correct the issue? Thank you.

Arun


On Sat, May 3, 2014 at 10:27 PM, VOIP Tests kamailio...@gmail.com wrote:

 No, the table does not seem to get updated. The entries in kamctl ul show
 --brief does not match the entries in the location table. Any suggestions?

 Thank you,
 Arun




 On Sat, May 3, 2014 at 10:10 PM, Alex Balashov 
 abalas...@evaristesys.comwrote:

 On 05/03/2014 11:08 PM, VOIP Tests wrote:

 Hello Alex, my usrloc setting are as follows:

 #!ifdef WITH_USRLOCDB
 modparam(usrloc, db_url, DBURL)
 modparam(usrloc, db_mode, 2)
 modparam(usrloc, use_domain, MULTIDOMAIN)
 modparam(usrloc, matching_mode, 1)


 Then, I would say wait for timer_interval seconds (by default 60) and
 see if the content of the location table now matches up with that of the
 runtime state of the registrar (i.e. kamctl ul show).


 --
 Alex Balashov - Principal
 Evariste Systems LLC
 235 E Ponce de Leon Ave
 Suite 106
 Decatur, GA 30030
 United States
 Tel: +1-678-954-0670
 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

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[SR-Users] Location table not getting updated

2014-05-03 Thread VOIP Tests
Hello, can someone let me know what could be the reason that the location
table in the kamailiio db does not get updated?  There are some extensions
that I can see in the location table but not others, but I am still able to
make extension to extension calls to all extensions. This is a multi domain
set up.

Thank you,
Arun
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Re: [SR-Users] Location table not getting updated

2014-05-03 Thread VOIP Tests
Hello Alex, my usrloc setting are as follows:

#!ifdef WITH_USRLOCDB
modparam(usrloc, db_url, DBURL)
modparam(usrloc, db_mode, 2)
modparam(usrloc, use_domain, MULTIDOMAIN)
modparam(usrloc, matching_mode, 1)

Arun


On Sat, May 3, 2014 at 9:51 PM, Alex Balashov abalas...@evaristesys.comwrote:

 On 05/03/2014 10:50 PM, VOIP Tests wrote:

  Hello, can someone let me know what could be the reason that the
 location table in the kamailiio db does not get updated?  There are some
 extensions that I can see in the location table but not others, but I am
 still able to make extension to extension calls to all extensions. This
 is a multi domain set up.


 The first question is, what is your usrloc db_mode?

 http://kamailio.org/docs/modules/4.1.x/modules/usrloc.
 html#usrloc.p.db_mode

 --
 Alex Balashov - Principal
 Evariste Systems LLC
 235 E Ponce de Leon Ave
 Suite 106
 Decatur, GA 30030
 United States
 Tel: +1-678-954-0670
 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

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Re: [SR-Users] Location table not getting updated

2014-05-03 Thread VOIP Tests
No, the table does not seem to get updated. The entries in kamctl ul show
--brief does not match the entries in the location table. Any suggestions?

Thank you,
Arun




On Sat, May 3, 2014 at 10:10 PM, Alex Balashov abalas...@evaristesys.comwrote:

 On 05/03/2014 11:08 PM, VOIP Tests wrote:

 Hello Alex, my usrloc setting are as follows:

 #!ifdef WITH_USRLOCDB
 modparam(usrloc, db_url, DBURL)
 modparam(usrloc, db_mode, 2)
 modparam(usrloc, use_domain, MULTIDOMAIN)
 modparam(usrloc, matching_mode, 1)


 Then, I would say wait for timer_interval seconds (by default 60) and
 see if the content of the location table now matches up with that of the
 runtime state of the registrar (i.e. kamctl ul show).


 --
 Alex Balashov - Principal
 Evariste Systems LLC
 235 E Ponce de Leon Ave
 Suite 106
 Decatur, GA 30030
 United States
 Tel: +1-678-954-0670
 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

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Re: [SR-Users] What does this script do?.

2014-05-02 Thread VOIP Tests
Thank you Daniel.


On Fri, May 2, 2014 at 4:29 AM, Daniel-Constantin Mierla
mico...@gmail.comwrote:

  Loading all integer-named attributes from domain_preferences table (which
 must have same structure as usr_preference table), using only domain for
 selecting the records.

 Cheers,
 Daniel


 On 30/04/14 23:38, VOIP Tests wrote:

  Can someone let me know what this script does?

 avp_db_load($ru/domain,i:/domain_preferences);



 Thank you,
 Arun



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


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[SR-Users] What does this script do?.

2014-04-30 Thread VOIP Tests
Can someone let me know what this script does?

avp_db_load($ru/domain,i:/domain_preferences);



Thank you,
Arun
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