Re: [OpenSIPS-Users] force one AOR

2017-01-12 Thread Søren Andersen
Hello,

I've some trouble with forceing 1 AOR per contact.

I'm using the config:

if (!save("location", "fc1")) {
sl_reply_error();
}

But then I reboot my SIP client opensips allows 2 AOR for the phonenumber? - 
But if I reboot the sip client again it'll delete all the AORs and insert the 
new one?

opensipsctl fifo ul_show_contact location 88XX
AOR:: 88175263
 Contact:: sip: 88XX@10.112.130.202 Q=
  Expires:: 897
  Callid:: 
25180a4e14bcef242fc01d1d1f5da9af@10.112.130.202
  Cseq:: 103
  User-agent:: ICOTERA 
IGW3000 (2.5.1)
  State:: CS_SYNC
  Flags:: 0
  Cflags::
  Socket:: 
udp:85.XXX.XXX.120:5060
  Methods:: 4294967295
 Contact:: sip: 88XX@10.112.130.202 Q=
  Expires:: 9
  Callid:: 
447b19de71f7ae6e305211c97f4574a9@10.112.130.202
  Cseq:: 107
  User-agent:: ICOTERA 
IGW3000 (2.5.1)
  State:: CS_SYNC
  Flags:: 0
  Cflags::
  Socket:: 
udp:85.XXX.XXX.120:5060
  Methods:: 4294967295

/Søren
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Re: [OpenSIPS-Users] drop_replies and drop_requests statistics counters

2016-07-14 Thread Søren Andersen
Hello OpenSIPS users,

I'm starting to see opensips statistics counters increase for drop_requests and 
drop_replies and my b2b server is not able to make any new connections to my 
proxy server.
I can't figure out why my proxy start to drops the connection? - Where is no 
errors in my log file. Do you guys have an idea why I'm experiencing this?


/Søren

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[OpenSIPS-Users] ERROR:tm:t_newtran: EoH not parsed on ACK?

2016-02-06 Thread Søren Andersen
I'm playing with sipp and trying to make some simple calls. - But then the call 
Is answered with 200 OK, I get "ERROR:tm:t_newtran: EoH not parsed" on the ACK? 
- And I can't figure out why?

My ACK looks like:

ACK sip:serv...@xx.xx.xx.120:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.118:8836;branch=z9hG4bK-24864-1-5
From: "sipp" ;tag=1
To: ;tag=1
Call-ID: 1-24...@xx.xx.xx.118
CSeq: 1 ACK
Contact: 
Max-Forwards: 100
User-Agent: SIPp/sippy_cup
Content-Length: 0


/Søren

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Re: [OpenSIPS-Users] recommended settings for performance

2016-02-01 Thread Søren Andersen
Hello there,

I'm looking for some reference configs on opensips performance.


-  How many cpu cores are you using and how many CPS are you peaking at?

-  Which startoptions in opensipsctlrc are you using?

-  How many opensips process children are you running?

Currently I'm planning to use STARTOPTIONS="-m 2048 -M 8" on a 2 core VM with 
children=8. - I hope I'll be able to run 600-1500 currently call on that setup. 
But that are your experience?

/Søren

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Re: [OpenSIPS-Users] B2B BYEs

2016-01-20 Thread Søren Andersen
Hi Bogdan,

My pleasure :)  - And it's working!!! :)

/Søren

Fra: Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
Sendt: 19. januar 2016 13:55
Til: Søren Andersen <s...@stofa.dk>; users@lists.opensips.org
Emne: Re: SV: SV: SV: SV: [OpenSIPS-Users] B2B BYEs

Hi Søren,

Thanks to all your support and info I found the problem. The fix is now 
available on GIT repo on all maintained versions.

Please update and confirm.

Regards,


Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 19.01.2016 12:38, Søren Andersen wrote:
Hi Bogdan-Andrei,

Hmm.. Actually it look likes "route_se" is missing in the LEGS:: after a 
restart?

# Before restart #
dlg:: 227897 param=774.0 state=5 last_invite_cseq=29551 last_method=4 
last_reply_code=200 db_flag=2
 callid:: B2B.348.227897
 from::  
uri=sip:xxx...@xx.xx.xx.36;user=phone<mailto:uri=sip:xxx...@xx.xx.xx.36;user=phone>
 tag=e33f629b154d16f5f57a07bf5d17124e-7644
 to::  
uri=sip:xxx...@xx.xx.xx.120<mailto:uri=sip:xxx...@xx.xx.xx.120> tag=2075842316
 cseq::  caller=29551 callee=1
 route_set::  
caller=<sip:XX.xx.XX.120;lr;ftag=e33f629b154d16f5f57a07bf5d17124e-7644;did=834.31f18683>
 contact::  caller=sip:xx.xx.xx.117:5060 
callee=sip:xx04@10.101.16.179:5060;transport=udp<mailto:callee=sip:xx04@10.101.16.179:5060;transport=udp>
 send_sock:: xx.xx.xx.117
 LEGS::
  leg:: 0 
tag=2075842316 cseq=29551 
contact=sip:xx04@10.101.16.179:5060;transport=udp<mailto:contact=sip:xx04@10.101.16.179:5060;transport=udp>
 
route_se=<sip:xx.xx.xx.120;lr;ftag=e33f629b154d16f5f57a07bf5d17124e-7644;did=834.31f18683>

# After restart #
dlg:: 227897 param=774.0 state=5 last_invite_cseq=29551 last_method=0 db_flag=0
 callid:: B2B.348.227897
 from::  
uri=sip:x...@xx.xx.xx.36;user=phone<mailto:uri=sip:x...@xx.xx.xx.36;user=phone>
 tag=e33f629b154d16f5f57a07bf5d17124e-7644
 to::  
uri=sip:x...@xx.xx.xxx.120<mailto:uri=sip:x...@xx.xx.xxx.120> 
tag=2075842316
 cseq::  caller=29551 callee=1
 route_set::  
caller=<sip:xx.xx.xx.120;lr;ftag=e33f629b154d16f5f57a07bf5d17124e-7644;did=834.31f18683>
 contact::  caller=sip:xx.xx.xx.117:5060 
callee=sip:xx04@10.101.16.179:5060;transport=udp<mailto:callee=sip:xx04@10.101.16.179:5060;transport=udp>
 send_sock:: xx.xx.xx.117
 LEGS::
  leg:: 0 
tag=2075842316 cseq=29551 
contact=sip:xx04@10.101.16.179:5060;transport=udp<mailto:contact=sip:xx04@10.101.16.179:5060;transport=udp>


/Søren
Fra: Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
Sendt: 19. januar 2016 11:13
Til: Søren Andersen <s...@stofa.dk><mailto:s...@stofa.dk>; 
users@lists.opensips.org<mailto:users@lists.opensips.org>
Emne: Re: SV: SV: SV: [OpenSIPS-Users] B2B BYEs

Hi Søren,

It seems that the callee route set is stored in db (see the b2b_entities table, 
the route1 field 
="<sip:XX.XX.XX.120;lr;ftag=a9ff31f61f2601c916f6044c2e20e16a-3c47;did=895.92039fc>")

Can you run the b2be_list MI command before and after the restart, to see if 
indeed, the callee route set is not properly restored ?

Regards,



Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 15.01.2016 12:21, Søren Andersen wrote:
Yup.

SELECT * FROM b2b.b2b_logic;
# id, si_key, scenario, sstate, next_sstate, sparam0, sparam1, sparam2, 
sparam3, sparam4, sdp, lifetime, e1_type, e1_sid, e1_from, e1_to, e1_key, 
e2_type, e2_sid, e2_from, e2_to, e2_key, e3_type, e3_sid, e3_from, e3_to, e3_key
'226', '360.0', 'StofaB2B', '1', '1', 
'sip:...@xx.xx.xx.120<mailto:sip:...@xx.xx.xx.120>', '', '', '', '', 
'', '1452896043', '0', 'server1', 
'sip:...@oxd3-sxw.xx.txxx.xxx;user=phone<mailto:sip:...@oxd3-sxw.xx.txxx.xxx;user=phone>',
 'sip:x...@xx.xx.xx.117:5060<mailto:sip:x...@xx.xx.xx.117:5060>', 
'B2B.39.160', '1', 'client1', 
'sip:x...@oxx3-sx.xx.xxx.xxx;user=phone<mailto:sip:x...@oxx3-sx.xx.xxx.xxx;user=phone>',
 'sip:...@xx.xx.xx.120<mailto:sip:...@xx.xx.xx.120>', 
'B2B.448.7695796', NULL, NULL, NULL, NULL, NULL

SELECT * FROM b2b.b2b_entities;
# id, type, state, ruri, from_uri, to_uri, from_dname, to_dname, tag0, tag1, 
callid, cseq0, cseq1, contact0, contact1, route0, route1, sockinfo_srv, param, 
lm, lrc, lic, leg_cseq, leg_route, leg_tag, leg_contact, leg_sockinfo
'316', '0', 

Re: [OpenSIPS-Users] B2B BYEs

2016-01-19 Thread Søren Andersen
Hi Bogdan-Andrei,

Hmm.. Actually it look likes "route_se" is missing in the LEGS:: after a 
restart?

# Before restart #
dlg:: 227897 param=774.0 state=5 last_invite_cseq=29551 last_method=4 
last_reply_code=200 db_flag=2
 callid:: B2B.348.227897
 from::  uri=sip:xxx...@xx.xx.xx.36;user=phone 
tag=e33f629b154d16f5f57a07bf5d17124e-7644
 to::  uri=sip:xxx...@xx.xx.xx.120 tag=2075842316
 cseq::  caller=29551 callee=1
 route_set::  
caller=<sip:XX.xx.XX.120;lr;ftag=e33f629b154d16f5f57a07bf5d17124e-7644;did=834.31f18683>
 contact::  caller=sip:xx.xx.xx.117:5060 
callee=sip:xx04@10.101.16.179:5060;transport=udp
 send_sock:: xx.xx.xx.117
 LEGS::
  leg:: 0 
tag=2075842316 cseq=29551 contact=sip:xx04@10.101.16.179:5060;transport=udp 
route_se=<sip:xx.xx.xx.120;lr;ftag=e33f629b154d16f5f57a07bf5d17124e-7644;did=834.31f18683>

# After restart #
dlg:: 227897 param=774.0 state=5 last_invite_cseq=29551 last_method=0 db_flag=0
 callid:: B2B.348.227897
 from::  uri=sip:x...@xx.xx.xx.36;user=phone 
tag=e33f629b154d16f5f57a07bf5d17124e-7644
 to::  uri=sip:x...@xx.xx.xxx.120 tag=2075842316
 cseq::  caller=29551 callee=1
 route_set::  
caller=<sip:xx.xx.xx.120;lr;ftag=e33f629b154d16f5f57a07bf5d17124e-7644;did=834.31f18683>
 contact::  caller=sip:xx.xx.xx.117:5060 
callee=sip:xx04@10.101.16.179:5060;transport=udp
 send_sock:: xx.xx.xx.117
 LEGS::
  leg:: 0 
tag=2075842316 cseq=29551 contact=sip:xx04@10.101.16.179:5060;transport=udp


/Søren
Fra: Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
Sendt: 19. januar 2016 11:13
Til: Søren Andersen <s...@stofa.dk>; users@lists.opensips.org
Emne: Re: SV: SV: SV: [OpenSIPS-Users] B2B BYEs

Hi Søren,

It seems that the callee route set is stored in db (see the b2b_entities table, 
the route1 field 
="<sip:XX.XX.XX.120;lr;ftag=a9ff31f61f2601c916f6044c2e20e16a-3c47;did=895.92039fc>")

Can you run the b2be_list MI command before and after the restart, to see if 
indeed, the callee route set is not properly restored ?

Regards,


Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 15.01.2016 12:21, Søren Andersen wrote:
Yup.

SELECT * FROM b2b.b2b_logic;
# id, si_key, scenario, sstate, next_sstate, sparam0, sparam1, sparam2, 
sparam3, sparam4, sdp, lifetime, e1_type, e1_sid, e1_from, e1_to, e1_key, 
e2_type, e2_sid, e2_from, e2_to, e2_key, e3_type, e3_sid, e3_from, e3_to, e3_key
'226', '360.0', 'StofaB2B', '1', '1', 
'sip:...@xx.xx.xx.120<mailto:sip:...@xx.xx.xx.120>', '', '', '', '', 
'', '1452896043', '0', 'server1', 
'sip:...@oxd3-sxw.xx.txxx.xxx;user=phone<mailto:sip:...@oxd3-sxw.xx.txxx.xxx;user=phone>',
 'sip:x...@xx.xx.xx.117:5060<mailto:sip:x...@xx.xx.xx.117:5060>', 
'B2B.39.160', '1', 'client1', 
'sip:x...@oxx3-sx.xx.xxx.xxx;user=phone<mailto:sip:x...@oxx3-sx.xx.xxx.xxx;user=phone>',
 'sip:...@xx.xx.xx.120<mailto:sip:...@xx.xx.xx.120>', 
'B2B.448.7695796', NULL, NULL, NULL, NULL, NULL

SELECT * FROM b2b.b2b_entities;
# id, type, state, ruri, from_uri, to_uri, from_dname, to_dname, tag0, tag1, 
callid, cseq0, cseq1, contact0, contact1, route0, route1, sockinfo_srv, param, 
lm, lrc, lic, leg_cseq, leg_route, leg_tag, leg_contact, leg_sockinfo
'316', '0', '5', 
'sip:...@xx.xx.xx.117:5060;user=phone<mailto:sip:...@xx.xx.xx.117:5060;user=phone>',
 
'sip:x...@osx3-sxw.xx..xxx;user=phone<mailto:sip:x...@osx3-sxw.xx..xxx;user=phone>',
 
'sip:X04@10.250.224.22;user=phone<mailto:sip:X04@10.250.224.22;user=phone>',
 '', '', 'mvgh6xvkp7z', 'B2B.39.160', 
'1q2rqu6-3232128...@osdx-ssw.xx.xxx.xx<mailto:1q2rqu6-3232128...@osdx-ssw.xx.xxx.xx>',
 '32736', '1', 
'sip:x...@xx.xx.xx.36:5060;transport=udp<mailto:sip:x...@xx.xx.xx.36:5060;transport=udp>',
 'sip:XX.XX.XX.117:5060', '<sip:XX.XX.XXX.117;lr>', '', 
'udp:XX.XX.XX.117:5060', '360.0', '0', '200', '32736', NULL, NULL, NULL, NULL, 
NULL
'317', '1', '5', '', 
'sip:x...@osdx-ssw.xx..xx;user=phone<mailto:sip:x...@osdx-ssw.xx..xx;user=phone>',
 'sip:x...@xx.xx.xx.120<mailto:sip:x...@xx.xx.xx.120>', '', '', 
'a9ff31f61f2601c916f6044c2e20e16a-3c47', '2761431845', 'B2B.448.7695796', 
'32737', '1', 'sip:XX.XXX.XXX.117:5060', 
'sip:XX04@10.101.16.179:5060;transport=udp<mailto:sip:XX04@10.101.16.179:5060

Re: [OpenSIPS-Users] B2B BYEs

2016-01-15 Thread Søren Andersen
Yup.

SELECT * FROM b2b.b2b_logic;
# id, si_key, scenario, sstate, next_sstate, sparam0, sparam1, sparam2, 
sparam3, sparam4, sdp, lifetime, e1_type, e1_sid, e1_from, e1_to, e1_key, 
e2_type, e2_sid, e2_from, e2_to, e2_key, e3_type, e3_sid, e3_from, e3_to, e3_key
'226', '360.0', 'StofaB2B', '1', '1', 'sip:...@xx.xx.xx.120', '', '', '', 
'', '', '1452896043', '0', 'server1', 
'sip:...@oxd3-sxw.xx.txxx.xxx;user=phone', 
'sip:x...@xx.xx.xx.117:5060', 'B2B.39.160', '1', 'client1', 
'sip:x...@oxx3-sx.xx.xxx.xxx;user=phone', 'sip:...@xx.xx.xx.120', 
'B2B.448.7695796', NULL, NULL, NULL, NULL, NULL

SELECT * FROM b2b.b2b_entities;
# id, type, state, ruri, from_uri, to_uri, from_dname, to_dname, tag0, tag1, 
callid, cseq0, cseq1, contact0, contact1, route0, route1, sockinfo_srv, param, 
lm, lrc, lic, leg_cseq, leg_route, leg_tag, leg_contact, leg_sockinfo
'316', '0', '5', 'sip:...@xx.xx.xx.117:5060;user=phone', 
'sip:x...@osx3-sxw.xx..xxx;user=phone', 
'sip:X04@10.250.224.22;user=phone', '', '', 'mvgh6xvkp7z', 'B2B.39.160', 
'1q2rqu6-3232128...@osdx-ssw.xx.xxx.xx', '32736', '1', 
'sip:x...@xx.xx.xx.36:5060;transport=udp', 'sip:XX.XX.XX.117:5060', 
'<sip:XX.XX.XXX.117;lr>', '', 'udp:XX.XX.XX.117:5060', '360.0', '0', '200', 
'32736', NULL, NULL, NULL, NULL, NULL
'317', '1', '5', '', 'sip:x...@osdx-ssw.xx..xx;user=phone', 
'sip:x...@xx.xx.xx.120', '', '', 'a9ff31f61f2601c916f6044c2e20e16a-3c47', 
'2761431845', 'B2B.448.7695796', '32737', '1', 'sip:XX.XXX.XXX.117:5060', 
'sip:XX04@10.101.16.179:5060;transport=udp', 
'<sip:XX.XX.XX.120;lr;ftag=a9ff31f61f2601c916f6044c2e20e16a-3c47;did=895.92039fc>',
 '', 'udp:XX.XX.XX.117:5060', '360.0', '4', '0', '32737', '32737', 
'<sip:XX.XX.XX120;lr;ftag=a9ff31f61f2601c916f6044c2e20e16a-3c47;did=895.92039fc>',
 '2761431845', 'sip:XX04@10.101.16.179:5060;transport=udp', NULL

/Søren

Fra: Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
Sendt: 15. januar 2016 11:07
Til: Søren Andersen <s...@stofa.dk>; users@lists.opensips.org
Emne: Re: SV: SV: [OpenSIPS-Users] B2B BYEs

Hi Søren,

In the scenario with the restart, after the restart, the sequential requests 
generated by b2b is missing the Route header - and it is bypassing the proxy - 
can you ppst the DB records for b2b you have during the restart?

Regards,


Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 15.01.2016 10:57, Søren Andersen wrote:
Hi Bogdan-Andrei,

Sure. I've attached the two files.
 (B2B server)
 (OpenSIPS Proxy)
 (SIP Client)

My scenario:



 
 

server1


client1
message

1


 
1
 


I'm doing this because my SIP provider can't handle multi dialogs, so if I 
first send a call to the sip client and if the call is unanswered I can't 
redirect the call to voicemail since this will generate a new to_tag. And this 
can't my SIP provider not handle. So therefore I need some B2B logic to make 
this happen for me.

/Søren


Fra: Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
Sendt: 14. januar 2016 16:41
Til: Søren Andersen <s...@stofa.dk><mailto:s...@stofa.dk>; OpenSIPS users 
mailling list <users@lists.opensips.org><mailto:users@lists.opensips.org>
Emne: Re: SV: [OpenSIPS-Users] B2B BYEs

Hi Søren,

Do you have an ngrep capture (taken from opensips b2b, covering all calls/legs 
involved in the scenario) showing the call with the B2B restarted ?

Regards,



Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 13.01.2016 14:39, Søren Andersen wrote:
Hi Bogdan-Andrei,

Only if I reload the B2B the BYEs is sent directly to the client. - But the 
funny thing is this only happens if the client is receives a call.  If the 
client initialize the call everything works fine.

/Søren

Fra: Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
Sendt: 13. januar 2016 11:39
Til: OpenSIPS users mailling list 
<users@lists.opensips.org><mailto:users@lists.opensips.org>; Søren Andersen 
<s...@stofa.dk><mailto:s...@stofa.dk>
Emne: Re: [OpenSIPS-Users] B2B BYEs

Hi Søren,

With or without restarting the B2B isn;t the BYE sent by client to the B2B 
instance ?

Regards,




Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 12.01.2016 16:32, Søren Andersen wrote:
Hello there,

I've some strange issues with the B2B module. - I use MySQL as backend, and if 
I reload OpenSIPS the BYE is sent directly to the client, and not my SIP proxy. 
My B2B and OpenSIPS is two difference servers.
If I take a look in b2b_entities table I notice the contact1 field contains the 
IP address of the client.  But if I don't reload the OpenSIPS the BYE will be 
sent correctly the my proxy, and then to the client.

Now, my question is how can I

Re: [OpenSIPS-Users] uac_replace_from multiple times?

2016-01-13 Thread Søren Andersen
Hi,

Thanks for your mail. - Actually I need to send the call out to my client 
without the prefix, and if the client don't answer the phone the call gets sent 
to another phone number via my ISP.
Do you have a smart solution for this?

/Søren
Fra: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] 
På vegne af Stefano Pisani
Sendt: 13. januar 2016 14:37
Til: OpenSIPS users mailling list <users@lists.opensips.org>
Emne: Re: [OpenSIPS-Users] uac_replace_from multiple times?

No you can't.
Use a variable to store the from and replace it once, just before to send out 
the message.



Il 13/01/2016 14.32, Søren Andersen ha scritto:
Hello,

I'm wondering if it's possible to use uac_replace_from multiple times? - fx. 
Inbound call gets changed by uac_replace_from and removed the +45 prefix. - But 
sometimes I need to forward the call back to my ISP, and they need to have +45 
in the from header. But if I try to use the function two times the sip headers 
gets invalid  like the below header:

From: 
<sip:12345...@xx.xx.xx.XXsip:+4512345678@1XX.2XX.2XX.3X;user=phone><mailto:sip:12345...@xx.xx.xx.XXsip:+4512345678@1XX.2XX.2XX.3X;user=phone>;tag=c675aa4668ba7a0c3150f682eea7a54b-7701

Perhaps there are a better option to do this?

/Søren





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[OpenSIPS-Users] uac_replace_from multiple times?

2016-01-13 Thread Søren Andersen
Hello,

I'm wondering if it's possible to use uac_replace_from multiple times? - fx. 
Inbound call gets changed by uac_replace_from and removed the +45 prefix. - But 
sometimes I need to forward the call back to my ISP, and they need to have +45 
in the from header. But if I try to use the function two times the sip headers 
gets invalid  like the below header:

From: 
;tag=c675aa4668ba7a0c3150f682eea7a54b-7701

Perhaps there are a better option to do this?

/Søren

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Re: [OpenSIPS-Users] B2B BYEs

2016-01-13 Thread Søren Andersen
Hi Bogdan-Andrei,

Only if I reload the B2B the BYEs is sent directly to the client. - But the 
funny thing is this only happens if the client is receives a call.  If the 
client initialize the call everything works fine.

/Søren

Fra: Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
Sendt: 13. januar 2016 11:39
Til: OpenSIPS users mailling list <users@lists.opensips.org>; Søren Andersen 
<s...@stofa.dk>
Emne: Re: [OpenSIPS-Users] B2B BYEs

Hi Søren,

With or without restarting the B2B isn;t the BYE sent by client to the B2B 
instance ?

Regards,


Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 12.01.2016 16:32, Søren Andersen wrote:
Hello there,

I've some strange issues with the B2B module. - I use MySQL as backend, and if 
I reload OpenSIPS the BYE is sent directly to the client, and not my SIP proxy. 
My B2B and OpenSIPS is two difference servers.
If I take a look in b2b_entities table I notice the contact1 field contains the 
IP address of the client.  But if I don't reload the OpenSIPS the BYE will be 
sent correctly the my proxy, and then to the client.

Now, my question is how can I make sure that B2B server will not try to 
communicate with the clients after a reload?

My topology:
ISP -> B2B -> Proxy -> Client

My config:

loadmodule "b2b_entities.so"
loadmodule "b2b_logic.so"
modparam("b2b_logic", "script_scenario", "/etc/opensips/b2b.xml")
modparam("b2b_entities", "db_url", 
"mysql://opensips:x...@xx.xx.xx.xx/b2b"<mailto:mysql://opensips:x...@xx.xx.xx.xx/b2b>)
modparam("b2b_entities", "db_mode", 1)
modparam("b2b_logic", "db_mode", 1)
modparam("b2b_logic", "db_url", 
"mysql://opensips:x...@xx.xx.xx.xx/b2b"<mailto:mysql://opensips:x...@xx.xx.xx.xx/b2b>)
modparam("b2b_entities", "replication_mode", 1)
modparam("b2b_logic", "init_callid_hdr", "Init-CallID")

if (is_method("INVITE") && !has_totag()) {
b2b_init_request("B2B","sip:$t...@xx.xx.xx.xx"<mailto:sip:$t...@xx.xx.xx.xx>);
exit;
}

/Søren




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Re: [OpenSIPS-Users] B2B BYEs

2016-01-12 Thread Søren Andersen
Hello there,

I've some strange issues with the B2B module. - I use MySQL as backend, and if 
I reload OpenSIPS the BYE is sent directly to the client, and not my SIP proxy. 
My B2B and OpenSIPS is two difference servers.
If I take a look in b2b_entities table I notice the contact1 field contains the 
IP address of the client.  But if I don't reload the OpenSIPS the BYE will be 
sent correctly the my proxy, and then to the client.

Now, my question is how can I make sure that B2B server will not try to 
communicate with the clients after a reload?

My topology:
ISP -> B2B -> Proxy -> Client

My config:

loadmodule "b2b_entities.so"
loadmodule "b2b_logic.so"
modparam("b2b_logic", "script_scenario", "/etc/opensips/b2b.xml")
modparam("b2b_entities", "db_url", "mysql://opensips:x...@xx.xx.xx.xx/b2b")
modparam("b2b_entities", "db_mode", 1)
modparam("b2b_logic", "db_mode", 1)
modparam("b2b_logic", "db_url", "mysql://opensips:x...@xx.xx.xx.xx/b2b")
modparam("b2b_entities", "replication_mode", 1)
modparam("b2b_logic", "init_callid_hdr", "Init-CallID")

if (is_method("INVITE") && !has_totag()) {
b2b_init_request("B2B","sip:$t...@xx.xx.xx.xx");
exit;
}

/Søren
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Re: [OpenSIPS-Users] hide to_tag from upstream provider

2015-11-30 Thread Søren Andersen
Hello,

Is it posibleto hide the to_tag from my upstream provider? - fx. If I rewrite 
the call to another user my upstream will only see one to_tag in the 1xx 
response?


/Søren

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[OpenSIPS-Users] Active/passive cluster with topology hiding?

2015-10-22 Thread Søren Andersen
Hello,

Would it be possible to have an active/passive cluster with topology hidden. - 
So if the cluster fails over to the second server it will recognize/match the 
already created dialogs from topology hiding?

/Søren

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Re: [OpenSIPS-Users] RFC1918 address in 180 ringing messages with topology hiding

2015-08-18 Thread Søren Andersen
Hello Guys,

Do you guys have an idea why I'm see some rfc1918 IP addresses in 180 ringing 
messages? - All other messages are topology hiding rewriteing with my server 
address.

180 ringing header:

2015/08/17 13:02:49.513033 XXX.XXX.XXX.117:5060 - XXX.XXX.XXX.36:5060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 
XXX.XXX.XXX.36:5060;rport=5060;received=XXX.XXX.XXX.36;branch=z9hG4bK9l6lqn002oahpqc8e6b1.1
Record-Route: sip:XXX.XXX.XXX.117;lr;ftag=z1h6i5xvtm;nat=yes
Contact: sip:XX35@192.168.250.129:64561;rinstance=413e0b8f9c7f9f99
To: XX35sip:+xxx...@xxx.xxx.xxx.117;user=phone;tag=cbf99c6e
From: sip:+xxx...@xxx.xxx.xxx.36;user=phone;tag=z1h6i5xvtm
Call-ID: 
pch8bubx-841640...@xxx3-ssw.xxx.XXXnetmailto:pch8bubx-841640...@xxx3-ssw.xxx.XXXnet
CSeq: 23370 INVITE
User-Agent: X-Lite release 4.8.4 stamp 76589
Allow-Events: talk, hold
Content-Length: 0

http://pastebin.com/GS9zuKNe - Opensips debug level 6 information.


/Søren
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Re: [OpenSIPS-Users] best way to append prefix (areacode)

2015-08-17 Thread Søren Andersen
Hello Guys,

What is the best way to append prefix (areacode) in OpenSIPS? - I need to 
append my areacode when I pass the call to my upstream provider, and then stip 
the prefix off when I receive a call.
Is the best way to use uac_replace then I receive/making a call?

/Søren
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Re: [OpenSIPS-Users] best way to append prefix (areacode)

2015-08-17 Thread Søren Andersen
Hi Bogdan,

Yup. Currently I'm using:

prefix(+areacode);

uac_replace_from(\+areacode$fU\,sip:+areacode$fU@$rd);
uac_replace_to($ruri);

But I don't know if that is the best way?

/Søren

Fra: Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
Sendt: 17. august 2015 15:27
Til: OpenSIPS users mailling list; Søren Andersen
Emne: Re: [OpenSIPS-Users] best way to append prefix (areacode)

Hi Søren,

Do you have to add this prefix to the RURI or to the TO URI too ?

For RURI is simple: $rU = my_prefix + $rU ;

Regards,


Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 17.08.2015 15:07, Søren Andersen wrote:
Hello Guys,

What is the best way to append prefix (areacode) in OpenSIPS? - I need to 
append my areacode when I pass the call to my upstream provider, and then stip 
the prefix off when I receive a call.
Is the best way to use uac_replace then I receive/making a call?

/Søren




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Re: [OpenSIPS-Users] best way to append prefix (areacode)

2015-08-17 Thread Søren Andersen
Cool, :)

I also have to append the prefix to the contact field..  If I use the subst the 
topology hiding won't replace my rfc1918 ip address with my server address. 
Should I use another function instead?

/Søren

Fra: Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
Sendt: 17. august 2015 16:40
Til: Søren Andersen s...@stofa.dk; OpenSIPS users mailling list 
users@lists.opensips.org
Emne: Re: SV: [OpenSIPS-Users] best way to append prefix (areacode)

Yes Søren, that is the correct way to do it.

Regards,


Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 17.08.2015 16:37, Søren Andersen wrote:
Hi Bogdan,

Yup. Currently I'm using:

prefix(+areacode);

uac_replace_from(\+areacode$fU\,sip:+areacode$fU@$rd);
uac_replace_to($ruri);

But I don't know if that is the best way?

/Søren

Fra: Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
Sendt: 17. august 2015 15:27
Til: OpenSIPS users mailling list; Søren Andersen
Emne: Re: [OpenSIPS-Users] best way to append prefix (areacode)

Hi Søren,

Do you have to add this prefix to the RURI or to the TO URI too ?

For RURI is simple: $rU = my_prefix + $rU ;

Regards,



Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 17.08.2015 15:07, Søren Andersen wrote:
Hello Guys,

What is the best way to append prefix (areacode) in OpenSIPS? - I need to 
append my areacode when I pass the call to my upstream provider, and then stip 
the prefix off when I receive a call.
Is the best way to use uac_replace then I receive/making a call?

/Søren





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