[SR-Users] Howto send rtp stats infos to homer

2016-11-23 Thread Oliver Roth 
Hi there

We are using homer to analyce and monitor traffic.
I did not get a solution to send x-rtp or p-rtp stats to homer.

How can I find infos about jitter, sent/received packages, lost packages, etc.?

Our current infos are based on
Rtpstat provided by kamailio.

At the moment I do

   $var(xrtpstat) = 
$(rtpstat{s.striptail,1});

 # Work the stats
 $var(octetsent) = $(var(xrtpstat){s.select,1, 
}); #octetsent / OS
 $var(packetsent) = $(var(xrtpstat){s.select,3, 
}); #PS
 $var(octetrcv) = $(var(xrtpstat){s.select,8, 
}); #OR
 $var(packetrcv) = $(var(xrtpstat){s.select,10, 
}); #PR
 $var(errorsent) = $(var(xrtpstat){s.select,5, 
});

  $var(errorrcv) = 
$(var(xrtpstat){s.select,12, });
 if ($var(octetsent) != "" || $var(packetsent) 
!= "")
 {

  xlog(,"L_INFO", "WITHINDLG 
PS=$var(rtp0),PR=$var(rtp1),PL=$var(rtp3) \n");

  xlog(,"L_INFO", "WITHINDLG 
xrtpstats=$var(xrtpstat) \r\n");

  xlog(,"L_INFO", "WITHINDLG 
rtpstats=$rtpstat \r\n");

  xlog(,"L_INFO", "WITHINDLG 
X-RTP-Stat: 
PS=$var(packetsent),OS=$var(octetsent),PR=$var(packetrcv),OR=$var(octetrcv),PL=$var(errorsent)
 \r\n");
 #append_hf("P-RTP-Stat:  $rtpstat\r\n");

  append_hf("X-RTP-Stat: 
PS=$var(packetsent);OS=$var(octetsent);PR=$var(packetrcv);OR=$var(octetrcv);PL=$var(errorsent);JI=600;LA=40;\r\n");
 }


Other question:
Where has this to be placed? Only on bye messages?

KR,
Oli


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Re: [SR-Users] ***SPAM***Re: SDP Codec not removed with RTPengine - but with rtpproxy it worked

2016-11-23 Thread Oliver Roth 
That helped!
Now it works.

Thanks Daniel!

KR,
Oli

-Ursprüngliche Nachricht-
Von: sr-users [mailto:sr-users-boun...@lists.sip-router.org] Im Auftrag von 
Daniel-Constantin Mierla
Gesendet: Freitag, 18. November 2016 15:21
An: Kamailio (SER) - Users Mailing List 
Betreff: ***SPAM***Re: [SR-Users] SDP Codec not removed with RTPengine - but 
with rtpproxy it worked

Move the record_route() function to be executed somewhere after the 
msg_apply_changes().

Cheers,
Daniel


On 18/11/16 10:10, Oliver Roth wrote:
> Found the problem with msg_apply_changes:
> cannot apply msg changes after adding record-route header
>
> see log below:
>
> nxp-c4-gw03-pkzh1 /usr/sbin/kamailio[4456]: ERROR: *** 
> cfgtrace:request_route=[SDP] c=[/etc/kamailio/kamailio-gw.cfg] l=745 a=24 
> n=msg_apply_changes  x
> NoxNov 18 10:04:27 nxp-c4-gw03-pkzh1 /usr/sbin/kamailio[4456]: ERROR: 
> textopsx [textopsx.c:171]: msg_apply_changes_f(): cannot apply msg 
> changes after adding record-route header - it breaks conditional 2nd 
> header
>
> Any idea?
>
>
> -Ursprüngliche Nachricht-
> Von: sr-users [mailto:sr-users-boun...@lists.sip-router.org] Im 
> Auftrag von Carsten Bock
> Gesendet: Freitag, 18. November 2016 09:42
> An: Kamailio (SER) - Users Mailing List 
> 
> Betreff: Re: [SR-Users] SDP Codec not removed with RTPengine - but 
> with rtpproxy it worked
>
> Hi Oli,
>
> if you remove the codec before sending it to RTPEngine, you should call 
> msg_apply_changes() after removing the codec.
>
> Thanks,
> Carsten
>
> 2016-11-18 9:39 GMT+01:00 Oliver Roth :
>> Hi,
>>
>>
>>
>> The codec is removed before sending it to rtpengine …
>>
>> See the log below
>>
>>
>>
>>
>>
>>
>>
>> Nov 18 09:37:38 nxp-c4-gw03-pkzh1 /usr/sbin/kamailio[4229]: ERROR: 
>> *** cfgtrace:request_route=[RTPPROXY] 
>> c=[/etc/kamailio/kamailio-gw.cfg]
>> l=1078
>> a=2 n=return
>>
>> Nov 18 09:37:38 nxp-c4-gw03-pkzh1 /usr/sbin/kamailio[4229]: DEBUG: 
>> sdpops
>> [sdpops_mod.c:199]: sdp_remove_str_codec_id_attrs(): removing line:
>> a=rtpmap:125 CLEARMODE/8000
>>
>> Nov 18 09:37:38 nxp-c4-gw03-pkzh1 /usr/sbin/kamailio[4229]: ERROR: 
>> *** cfgtrace:branch_route=[MANAGE_BRANCH]
>> c=[/etc/kamailio/kamailio-gw.cfg]
>> l=1889 a=24 n=rtpengine_manage
>>
>> Nov 18 09:37:38 nxp-c4-gw03-pkzh1 /usr/sbin/kamailio[4229]: DEBUG: 
>> 
>> [mem/f_malloc.c:444]: fm_malloc(): fm_malloc(0x7fe376c30010, 536) 
>> called from rtpengine: bencode.c: __bencode_piece_new(79)
>>
>> Nov 18 09:37:38 nxp-c4-gw03-pkzh1 /usr/sbin/kamailio[4229]: DEBUG: 
>> rtpengine
>> [rtpengine_funcs.c:140]: check_content_type(): type  
>> found valid
>>
>>
>>
>> But in the outgoing sdp the codec is still listed
>>
>>
>>
>> v=0
>>
>> o=Dialogic_SDP 4043679 0 IN IP4 213.173.185.46
>>
>> s=Dialogic-SIP
>>
>> c=IN IP4 213.173.185.47
>>
>> t=0 0
>>
>> m=audio 9008 RTP/AVP 8 0 125 101
>>
>> a=rtpmap:0 PCMU/8000
>>
>> a=rtpmap:8 PCMA/8000
>>
>> a=rtpmap:125 CLEARMODE/8000
>>
>> a=rtpmap:101 telephone-event/8000
>>
>> a=fmtp:101 0-15
>>
>> a=silenceSupp:off - - - -
>>
>>
>>
>> Any ideas?
>>
>>
>>
>> Kr,
>>
>> Oli
>>
>>
>>
>> Von: Daniel-Constantin Mierla [mailto:mico...@gmail.com]
>> Gesendet: Freitag, 18. November 2016 09:27
>> An: Oliver Roth ; Kamailio (SER) - Users 
>> Mailing List 
>> Betreff: Re: AW: [SR-Users] SDP Codec not removed with RTPengine - 
>> but with rtpproxy it worked
>>
>>
>>
>> Hello,
>>
>>
>>
>> On 16/11/16 14:59, Oliver Roth wrote:
>>
>> Hi,
>>
>>
>>
>> I guess it is after executing rtpengine_maange() – but not really sure.
>>
>> How can I check that?
>>
>> load debugger module and enable cfgtrace option via modparam. Then 
>> you should see what functions are executed from config.
>>
>> Cheers,
>> Daniel
>>
>>
>>
>>
>> I get a different sdp header that is going out of the gateway
>>
>>
>>
>> v=0
>>
>> o=Dialogic_SDP 3975025 0 IN IP4 213.173.185.38
>>
>> s=Dialogic-SIP
>>
>> c=IN IP4 185.49.222.198
>>
>> t=0 0
>>
>> m=audio 20306 RTP/AVP 8 0 125 101
>>
>> a=rtpmap:0 PCMU/8000
>>
>> a=rtpmap:8 PCMA/8000
>>
>> a=rtpmap:125 CLEARMODE/8000
>>

Re: [SR-Users] SDP Codec not removed with RTPengine - but with rtpproxy it worked

2016-11-18 Thread Oliver Roth 
Found the problem with msg_apply_changes:
cannot apply msg changes after adding record-route header

see log below:

nxp-c4-gw03-pkzh1 /usr/sbin/kamailio[4456]: ERROR: *** 
cfgtrace:request_route=[SDP] c=[/etc/kamailio/kamailio-gw.cfg] l=745 a=24 
n=msg_apply_changes  x
NoxNov 18 10:04:27 nxp-c4-gw03-pkzh1 /usr/sbin/kamailio[4456]: ERROR: textopsx 
[textopsx.c:171]: msg_apply_changes_f(): cannot apply msg changes after adding 
record-route header - it breaks conditional 2nd header

Any idea?


-Ursprüngliche Nachricht-
Von: sr-users [mailto:sr-users-boun...@lists.sip-router.org] Im Auftrag von 
Carsten Bock
Gesendet: Freitag, 18. November 2016 09:42
An: Kamailio (SER) - Users Mailing List 
Betreff: Re: [SR-Users] SDP Codec not removed with RTPengine - but with 
rtpproxy it worked

Hi Oli,

if you remove the codec before sending it to RTPEngine, you should call 
msg_apply_changes() after removing the codec.

Thanks,
Carsten

2016-11-18 9:39 GMT+01:00 Oliver Roth :
> Hi,
>
>
>
> The codec is removed before sending it to rtpengine …
>
> See the log below
>
>
>
>
>
>
>
> Nov 18 09:37:38 nxp-c4-gw03-pkzh1 /usr/sbin/kamailio[4229]: ERROR: *** 
> cfgtrace:request_route=[RTPPROXY] c=[/etc/kamailio/kamailio-gw.cfg] 
> l=1078
> a=2 n=return
>
> Nov 18 09:37:38 nxp-c4-gw03-pkzh1 /usr/sbin/kamailio[4229]: DEBUG: 
> sdpops
> [sdpops_mod.c:199]: sdp_remove_str_codec_id_attrs(): removing line:
> a=rtpmap:125 CLEARMODE/8000
>
> Nov 18 09:37:38 nxp-c4-gw03-pkzh1 /usr/sbin/kamailio[4229]: ERROR: *** 
> cfgtrace:branch_route=[MANAGE_BRANCH] 
> c=[/etc/kamailio/kamailio-gw.cfg]
> l=1889 a=24 n=rtpengine_manage
>
> Nov 18 09:37:38 nxp-c4-gw03-pkzh1 /usr/sbin/kamailio[4229]: DEBUG: 
> 
> [mem/f_malloc.c:444]: fm_malloc(): fm_malloc(0x7fe376c30010, 536) 
> called from rtpengine: bencode.c: __bencode_piece_new(79)
>
> Nov 18 09:37:38 nxp-c4-gw03-pkzh1 /usr/sbin/kamailio[4229]: DEBUG: 
> rtpengine
> [rtpengine_funcs.c:140]: check_content_type(): type  
> found valid
>
>
>
> But in the outgoing sdp the codec is still listed
>
>
>
> v=0
>
> o=Dialogic_SDP 4043679 0 IN IP4 213.173.185.46
>
> s=Dialogic-SIP
>
> c=IN IP4 213.173.185.47
>
> t=0 0
>
> m=audio 9008 RTP/AVP 8 0 125 101
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:8 PCMA/8000
>
> a=rtpmap:125 CLEARMODE/8000
>
> a=rtpmap:101 telephone-event/8000
>
> a=fmtp:101 0-15
>
> a=silenceSupp:off - - - -
>
>
>
> Any ideas?
>
>
>
> Kr,
>
> Oli
>
>
>
> Von: Daniel-Constantin Mierla [mailto:mico...@gmail.com]
> Gesendet: Freitag, 18. November 2016 09:27
> An: Oliver Roth ; Kamailio (SER) - Users 
> Mailing List 
> Betreff: Re: AW: [SR-Users] SDP Codec not removed with RTPengine - but 
> with rtpproxy it worked
>
>
>
> Hello,
>
>
>
> On 16/11/16 14:59, Oliver Roth wrote:
>
> Hi,
>
>
>
> I guess it is after executing rtpengine_maange() – but not really sure.
>
> How can I check that?
>
> load debugger module and enable cfgtrace option via modparam. Then you 
> should see what functions are executed from config.
>
> Cheers,
> Daniel
>
>
>
>
> I get a different sdp header that is going out of the gateway
>
>
>
> v=0
>
> o=Dialogic_SDP 3975025 0 IN IP4 213.173.185.38
>
> s=Dialogic-SIP
>
> c=IN IP4 185.49.222.198
>
> t=0 0
>
> m=audio 20306 RTP/AVP 8 0 125 101
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:8 PCMA/8000
>
> a=rtpmap:125 CLEARMODE/8000
>
> a=rtpmap:101 telephone-event/8000
>
> a=fmtp:101 0-15
>
> a=silenceSupp:off - - - -
>
> a=sendrecv
>
> a=rtcp:20307
>
> a=ice-ufrag:UMTBynMy
>
> a=ice-pwd:63JdnvLS7cMyCQ978BA6syPmiI
>
> a=candidate:GUTMVOhP7VJyBkZg 1 UDP 2130706431 185.49.222.198 20306 typ 
> host
>
> a=candidate:GUTMVOhP7VJyBkZg 2 UDP 2130706430 185.49.222.198 20307 typ 
> host
>
>
>
> Incoming was
>
>
>
> v=0
>
> o=Dialogic_SDP 3975025 0 IN IP4 213.173.185.38
>
> s=Dialogic-SIP
>
> c=IN IP4 213.173.185.39
>
> t=0 0
>
> m=audio 9036 RTP/AVP 8 0 125 101
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:8 PCMA/8000
>
> a=rtpmap:125 CLEARMODE/8000
>
> a=rtpmap:101 telephone-event/8000
>
> a=fmtp:101 0-15
>
> a=silenceSupp:off - - - -
>
>
>
>
>
>
>
> Von: sr-users [mailto:sr-users-boun...@lists.sip-router.org] Im 
> Auftrag von Daniel-Constantin Mierla
> Gesendet: Mittwoch, 16. November 2016 10:34
> An: Kamailio (SER) - Users Mailing List 
> 
> Betreff: Re: [SR-Users] SDP Codec not removed with RTPengine - but 
> with rtpproxy it worked
>
>
>
> Hello,
>

Re: [SR-Users] SDP Codec not removed with RTPengine - but with rtpproxy it worked

2016-11-18 Thread Oliver Roth 
Hi,

The codec is removed before sending it to rtpengine …
See the log below



Nov 18 09:37:38 nxp-c4-gw03-pkzh1 /usr/sbin/kamailio[4229]: ERROR: *** 
cfgtrace:request_route=[RTPPROXY] c=[/etc/kamailio/kamailio-gw.cfg] l=1078 a=2 
n=return
Nov 18 09:37:38 nxp-c4-gw03-pkzh1 /usr/sbin/kamailio[4229]: DEBUG: sdpops 
[sdpops_mod.c:199]: sdp_remove_str_codec_id_attrs(): removing line: 
a=rtpmap:125 CLEARMODE/8000
Nov 18 09:37:38 nxp-c4-gw03-pkzh1 /usr/sbin/kamailio[4229]: ERROR: *** 
cfgtrace:branch_route=[MANAGE_BRANCH] c=[/etc/kamailio/kamailio-gw.cfg] l=1889 
a=24 n=rtpengine_manage
Nov 18 09:37:38 nxp-c4-gw03-pkzh1 /usr/sbin/kamailio[4229]: DEBUG:  
[mem/f_malloc.c:444]: fm_malloc(): fm_malloc(0x7fe376c30010, 536) called from 
rtpengine: bencode.c: __bencode_piece_new(79)
Nov 18 09:37:38 nxp-c4-gw03-pkzh1 /usr/sbin/kamailio[4229]: DEBUG: rtpengine 
[rtpengine_funcs.c:140]: check_content_type(): type  found 
valid

But in the outgoing sdp the codec is still listed

v=0
o=Dialogic_SDP 4043679 0 IN IP4 213.173.185.46
s=Dialogic-SIP
c=IN IP4 213.173.185.47
t=0 0
m=audio 9008 RTP/AVP 8 0 125 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:125 CLEARMODE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -

Any ideas?

Kr,
Oli

Von: Daniel-Constantin Mierla [mailto:mico...@gmail.com]
Gesendet: Freitag, 18. November 2016 09:27
An: Oliver Roth ; Kamailio (SER) - Users Mailing List 

Betreff: Re: AW: [SR-Users] SDP Codec not removed with RTPengine - but with 
rtpproxy it worked


Hello,

On 16/11/16 14:59, Oliver Roth wrote:
Hi,

I guess it is after executing rtpengine_maange() – but not really sure.
How can I check that?
load debugger module and enable cfgtrace option via modparam. Then you should 
see what functions are executed from config.

Cheers,
Daniel



I get a different sdp header that is going out of the gateway

v=0
o=Dialogic_SDP 3975025 0 IN IP4 213.173.185.38
s=Dialogic-SIP
c=IN IP4 185.49.222.198
t=0 0
m=audio 20306 RTP/AVP 8 0 125 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:125 CLEARMODE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -
a=sendrecv
a=rtcp:20307
a=ice-ufrag:UMTBynMy
a=ice-pwd:63JdnvLS7cMyCQ978BA6syPmiI
a=candidate:GUTMVOhP7VJyBkZg 1 UDP 2130706431 185.49.222.198 20306 typ host
a=candidate:GUTMVOhP7VJyBkZg 2 UDP 2130706430 185.49.222.198 20307 typ host

Incoming was

v=0
o=Dialogic_SDP 3975025 0 IN IP4 213.173.185.38
s=Dialogic-SIP
c=IN IP4 213.173.185.39
t=0 0
m=audio 9036 RTP/AVP 8 0 125 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:125 CLEARMODE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -



Von: sr-users [mailto:sr-users-boun...@lists.sip-router.org] Im Auftrag von 
Daniel-Constantin Mierla
Gesendet: Mittwoch, 16. November 2016 10:34
An: Kamailio (SER) - Users Mailing List 
<mailto:sr-users@lists.sip-router.org>
Betreff: Re: [SR-Users] SDP Codec not removed with RTPengine - but with 
rtpproxy it worked


Hello,

are you executing rtpengine_manage() before or after removing the codec?

Cheers,
Daniel

On 16/11/16 10:03, Oliver Roth wrote:
Hi there

I have the following problem – I need to remove a codec in the initial INVITE.
This happens since I changed from rtpproxy to rtpengine. I changed all 
rtpproxy_manage() to rtpengine_manage().

Originating INVITE with the “clearmode”

m=audio 9196 RTP/AVP 8 0 125 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:125 CLEARMODE/8000
a=rtpmap:101 telephone-event/8000


I do the following in the config
route[SDP] {
xlog(, "L_INFO", "SDP: request method $rm");

# remove CLEARMODE if Colt
   if($avp(s:todirection) =~"^MyCarrier+") {

   if (is_method("INVITE")) {
   xlog(, "L_INFO", "SDP remove: 
request method $rm");
   msg_apply_changes();
   
sdp_remove_codecs_by_name("CLEARMODE");
   #sdp_remove_codecs_by_id("125");

   }
}
}
This block gets hit as I can see with the xlog entry.

In the sent INVITE the “Clearmode” is still in the sdp header

a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:125 CLEARMODE/8000
a=rtpmap:101 telephone-event/8000

So with rtpproxy – it worked
With rtpengine not any more …

Any idea?

KR,
Oli





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http://www.linkedin.com/in/miconda

Ka

Re: [SR-Users] SDP Codec not removed with RTPengine - but with rtpproxy it worked

2016-11-16 Thread Oliver Roth 
Hi,

I guess it is after executing rtpengine_maange() – but not really sure.
How can I check that?

I get a different sdp header that is going out of the gateway

v=0
o=Dialogic_SDP 3975025 0 IN IP4 213.173.185.38
s=Dialogic-SIP
c=IN IP4 185.49.222.198
t=0 0
m=audio 20306 RTP/AVP 8 0 125 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:125 CLEARMODE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -
a=sendrecv
a=rtcp:20307
a=ice-ufrag:UMTBynMy
a=ice-pwd:63JdnvLS7cMyCQ978BA6syPmiI
a=candidate:GUTMVOhP7VJyBkZg 1 UDP 2130706431 185.49.222.198 20306 typ host
a=candidate:GUTMVOhP7VJyBkZg 2 UDP 2130706430 185.49.222.198 20307 typ host

Incoming was

v=0
o=Dialogic_SDP 3975025 0 IN IP4 213.173.185.38
s=Dialogic-SIP
c=IN IP4 213.173.185.39
t=0 0
m=audio 9036 RTP/AVP 8 0 125 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:125 CLEARMODE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -



Von: sr-users [mailto:sr-users-boun...@lists.sip-router.org] Im Auftrag von 
Daniel-Constantin Mierla
Gesendet: Mittwoch, 16. November 2016 10:34
An: Kamailio (SER) - Users Mailing List 
Betreff: Re: [SR-Users] SDP Codec not removed with RTPengine - but with 
rtpproxy it worked


Hello,

are you executing rtpengine_manage() before or after removing the codec?

Cheers,
Daniel

On 16/11/16 10:03, Oliver Roth wrote:
Hi there

I have the following problem – I need to remove a codec in the initial INVITE.
This happens since I changed from rtpproxy to rtpengine. I changed all 
rtpproxy_manage() to rtpengine_manage().

Originating INVITE with the “clearmode”

m=audio 9196 RTP/AVP 8 0 125 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:125 CLEARMODE/8000
a=rtpmap:101 telephone-event/8000


I do the following in the config
route[SDP] {
xlog(, "L_INFO", "SDP: request method $rm");

# remove CLEARMODE if Colt
   if($avp(s:todirection) =~"^MyCarrier+") {

   if (is_method("INVITE")) {
   xlog(, "L_INFO", "SDP remove: 
request method $rm");
   msg_apply_changes();
   
sdp_remove_codecs_by_name("CLEARMODE");
   #sdp_remove_codecs_by_id("125");

   }
}
}
This block gets hit as I can see with the xlog entry.

In the sent INVITE the “Clearmode” is still in the sdp header

a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:125 CLEARMODE/8000
a=rtpmap:101 telephone-event/8000

So with rtpproxy – it worked
With rtpengine not any more …

Any idea?

KR,
Oli




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--

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http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com
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[SR-Users] SDP Codec not removed with RTPengine - but with rtpproxy it worked

2016-11-16 Thread Oliver Roth 
Hi there

I have the following problem - I need to remove a codec in the initial INVITE.
This happens since I changed from rtpproxy to rtpengine. I changed all 
rtpproxy_manage() to rtpengine_manage().

Originating INVITE with the "clearmode"

m=audio 9196 RTP/AVP 8 0 125 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:125 CLEARMODE/8000
a=rtpmap:101 telephone-event/8000


I do the following in the config
route[SDP] {
xlog(, "L_INFO", "SDP: request method $rm");

# remove CLEARMODE if Colt
   if($avp(s:todirection) =~"^MyCarrier+") {

   if (is_method("INVITE")) {
   xlog(, "L_INFO", "SDP remove: 
request method $rm");
   msg_apply_changes();
   
sdp_remove_codecs_by_name("CLEARMODE");
   #sdp_remove_codecs_by_id("125");

   }
}
}
This block gets hit as I can see with the xlog entry.

In the sent INVITE the "Clearmode" is still in the sdp header

a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:125 CLEARMODE/8000
a=rtpmap:101 telephone-event/8000

So with rtpproxy - it worked
With rtpengine not any more ...

Any idea?

KR,
Oli
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[SR-Users] Advice for routing and sbc functions

2016-09-12 Thread Oliver Roth 
Hi all


I am using kamailio for several years now and I am really happy with it!

Now an new scenario to solve:

Sip-carrier A --> Kamailio loadbalancer --> multiple kamailio routing gateways 
--> freeswitch/sbc/media-gw/whatever --> sip - carriers B,C & D

Kamailio gateways:
They are doing different checks, manipulate the numbers (national 
/international format) and they use the carrierroute module

freeswitch/sbc/media-gw/whatever:
how can I use these services with the routing information from the carrierroute 
module?
These media-gws will do the connection and maybe transcoding to the carriers.

Can you please help me with the interaction within the systems "routing 
gateways" and the "media/sbc gateways"?

Do I have to add all the carriers in the carrierroute as "routes" with the ip 
of the media gw/freeswitch and then splitting it up on the freeswitch again?

Thanks for helping
Oli
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[SR-Users] Contact header / bye

2016-08-15 Thread Oliver Roth 
Hi all

I do have a problem with getting a propper interconnection to our new carrier..
Situation as following

User --> Class5 switch --> Kamailio --> carrier

I always get the ip of the class5 switch in the contact.
But my carrier needs the kamailios contact ip - because of the ack handling.

I replaced in the pstn routing parte the contact by
remove_hf("Contact");
append_hf("Contact: \r\n");

This worked - but the bye did not work any more.

Please help me on that topic ... I am searching for a solution for more than 8 
weeks now ...

KR,
Oli
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Re: [SR-Users] ACK / BYE transaction problem

2016-07-01 Thread Oliver Roth 
Hi Daniel

Yes I get all the traffic in the lbl (first hop) and on the gw (routing gw).
Even the call is processed and I get rtp on both sides.

But: ACK is not routed to the carrier behind the gw. Therefore the call gets 
cut from carrierside – because of missing ACK after 200 OK

I could provide a complete trace from the whole scenario.

About your inputs:
Where should they be done? On the LBL (dispatcher, first hop) or on the gateway 
(gw)?

Regards,
Oli


Von: sr-users [mailto:sr-users-boun...@lists.sip-router.org] Im Auftrag von 
Daniel-Constantin Mierla
Gesendet: Freitag, 1. Juli 2016 09:55
An: Kamailio (SER) - Users Mailing List 
Betreff: Re: [SR-Users] ACK / BYE transaction problem


Hello,

can you get the traffic on load balancer (or the first hop in your network that 
received the invite)? There you can see if the fault is in your network or not.

In kamailio.cfg, be sure you don't use fix_nated_contact() function unless you 
are the first hop receiving the sip message (request or reply). As a matter of 
fact, fix_nated_contact() should not be used anymore, use set_contact_alias() 
instead and handle requests within dialog with handle_ruri_alias() -- see 
default kamailio.cfg for latest kamailio versions.

Cheers,
Daniel

On 29/06/16 17:20, Oliver Roth wrote:
Yes the ip for the [carr] is missing.

But I thought, the [gw ] should create the ack based on the transaction and 
send it to the [carr]

My situation
Class5 system [c5] --> Loadbalancer kamailio (dispatcher module) [lbl] ---> 
gateway kamailio [gw] --> carrier [carr]

So who is doing a mistake? The lbl, the gw or even the c5 system?

If helpful, I could provide a trace with all the stations in it.

Kr,
Oli

Von: sr-users [mailto:sr-users-boun...@lists.sip-router.org] Im Auftrag von 
Francisco Valentin Vinagrero
Gesendet: Mittwoch, 29. Juni 2016 16:17
An: Kamailio (SER) - Users Mailing List 
<mailto:sr-users@lists.sip-router.org>
Betreff: Re: [SR-Users] ACK / BYE transaction problem

Your ACK is missing the right IP in the RURI (should be the one in the contact 
header in the 200 OK) and the Route headers for every Record-Route in the 200 
OK, if I understand well your scenario…

From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of 
Oliver Roth
Sent: 29 June 2016 16:04
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.sip-router.org>>
Subject: Re: [SR-Users] ACK / BYE transaction problem

I removed the changes for the to header – so it is not touched all the time



200 ok from [carr]

SIP/2.0 200 OK
From: ;tag=sc1NXPTEST-4c9b51343502af61
To: ;tag=snl_0015024070
Call-ID: 5773d0ab9b30-5bau50gxp4en
CSeq: 1 INVITE
Via: SIP/2.0/UDP 
185.49.222.44;branch=z9hG4bKbe7c.0f5ca21e3a66f41694ca709ac28c1192.0
Via: SIP/2.0/UDP 
185.49.222.43;branch=z9hG4bKbe7c.a5e4cba8b1d9a4a56d1120d012a06850.0
Via: SIP/2.0/UDP 
212.25.7.69:5060;uac=sc1;branch=z9hG4bKsc1NXPTEST-ae749ab817378131
Record-Route: 
Record-Route: 
Contact: 
Content-Type: application/sdp
Content-Length: 197
Accept-Language: en;q=0.0
Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER, UPDATE
Supported: timer
Session-Expires: 1800;refresher=uas
Date: Wed, 29 Jun 2016 13:44:20 GMT

v=0
o=- 277262053 1 IN IP4 81.7.235.228
s=-
c=IN IP4 81.7.235.228
t=0 0
m=audio 24212 RTP/AVP 8 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=ptime:20


ACK from [lbl] to [gw]

ACK sip:185.49.222.44;did=4b7.9422;lr=on SIP/2.0

From: ;tag=sc1NXPTEST-4c9b51343502af61

To: ;tag=snl_0015024070

Call-ID: 5773d0ab9b30-5bau50gxp4en

CSeq: 1 ACK

Max-Forwards: 28

Via: SIP/2.0/UDP 
185.49.222.43;branch=z9hG4bKbe7c.a6f12b66454fbb7b536aa22cef3d568c.0

Via: SIP/2.0/UDP 
185.49.222.43;branch=z9hG4bKbe7c.7bc483a166362b13ba3cc40ca60308ea.0

Via: SIP/2.0/UDP 212.25.7.69:5060;branch=z9hG4bKsc1NXPTEST-ae749ab817378131A

Content-Length: 0

X-gateway: 

X-SI: 

X-gateway: 

X-SI: 


Initial Inivte to [gw] from [lbl]

INVITE sip:0794567735@185.49.222.43:5060 SIP/2.0

Record-Route: 

From: ;tag=sc1NXPTEST-4c9b51343502af61

To: 

Call-ID: 5773d0ab9b30-5bau50gxp4en

CSeq: 1 INVITE

Allow: INVITE, ACK, CANCEL, BYE, OPTIONS

Max-Forwards: 29

User-Agent: AareSwitch/6.2.8553

Via: SIP/2.0/UDP 
185.49.222.43;branch=z9hG4bKbe7c.a5e4cba8b1d9a4a56d1120d012a06850.0

Via: SIP/2.0/UDP 
212.25.7.69:5060;uac=sc1;branch=z9hG4bKsc1NXPTEST-ae749ab817378131

Contact: 

P-Asserted-Identity: 

Content-Type: application/sdp

Content-Length: 401


Invite sent to [carr] from [gw]

INVITE sip:41794567735@81.7.235.236 SIP/2.0

Record-Route: 

Record-Route: 

From: ;tag=sc1NXPTEST-4c9b51343502af61

To: 

Call-ID: 5773d0ab9b30-5bau50gxp4en

CSeq: 1 INVITE

Allow: INVITE, ACK, CANCEL, BYE, OPTIONS

Max-Forwards: 28

User-Agent: AareSwitch/6.2.8553

Contact: 

Via: SIP/2.0/UDP 
185.49.222.44;branch=z9hG4bKbe7c.0f5ca21e3a66f41694ca709ac28c1192.0

Via: SIP/2.0/UDP 
185.49.222.43;branch=z9hG4bKbe7c.a5e4cba8b1d9a4a56d1120d012a06850.0

Via: SIP/2.0/UDP 
212.2

Re: [SR-Users] ACK / BYE transaction problem

2016-06-29 Thread Oliver Roth 
Yes the ip for the [carr] is missing.

But I thought, the [gw ] should create the ack based on the transaction and 
send it to the [carr]

My situation
Class5 system [c5] --> Loadbalancer kamailio (dispatcher module) [lbl] ---> 
gateway kamailio [gw] --> carrier [carr]

So who is doing a mistake? The lbl, the gw or even the c5 system?

If helpful, I could provide a trace with all the stations in it.

Kr,
Oli

Von: sr-users [mailto:sr-users-boun...@lists.sip-router.org] Im Auftrag von 
Francisco Valentin Vinagrero
Gesendet: Mittwoch, 29. Juni 2016 16:17
An: Kamailio (SER) - Users Mailing List 
Betreff: Re: [SR-Users] ACK / BYE transaction problem

Your ACK is missing the right IP in the RURI (should be the one in the contact 
header in the 200 OK) and the Route headers for every Record-Route in the 200 
OK, if I understand well your scenario...

From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of 
Oliver Roth
Sent: 29 June 2016 16:04
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.sip-router.org>>
Subject: Re: [SR-Users] ACK / BYE transaction problem

I removed the changes for the to header - so it is not touched all the time



200 ok from [carr]

SIP/2.0 200 OK
From: ;tag=sc1NXPTEST-4c9b51343502af61
To: ;tag=snl_0015024070
Call-ID: 5773d0ab9b30-5bau50gxp4en
CSeq: 1 INVITE
Via: SIP/2.0/UDP 
185.49.222.44;branch=z9hG4bKbe7c.0f5ca21e3a66f41694ca709ac28c1192.0
Via: SIP/2.0/UDP 
185.49.222.43;branch=z9hG4bKbe7c.a5e4cba8b1d9a4a56d1120d012a06850.0
Via: SIP/2.0/UDP 
212.25.7.69:5060;uac=sc1;branch=z9hG4bKsc1NXPTEST-ae749ab817378131
Record-Route: 
Record-Route: 
Contact: 
Content-Type: application/sdp
Content-Length: 197
Accept-Language: en;q=0.0
Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER, UPDATE
Supported: timer
Session-Expires: 1800;refresher=uas
Date: Wed, 29 Jun 2016 13:44:20 GMT

v=0
o=- 277262053 1 IN IP4 81.7.235.228
s=-
c=IN IP4 81.7.235.228
t=0 0
m=audio 24212 RTP/AVP 8 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=ptime:20


ACK from [lbl] to [gw]

ACK sip:185.49.222.44;did=4b7.9422;lr=on SIP/2.0

From: ;tag=sc1NXPTEST-4c9b51343502af61

To: ;tag=snl_0015024070

Call-ID: 5773d0ab9b30-5bau50gxp4en

CSeq: 1 ACK

Max-Forwards: 28

Via: SIP/2.0/UDP 
185.49.222.43;branch=z9hG4bKbe7c.a6f12b66454fbb7b536aa22cef3d568c.0

Via: SIP/2.0/UDP 
185.49.222.43;branch=z9hG4bKbe7c.7bc483a166362b13ba3cc40ca60308ea.0

Via: SIP/2.0/UDP 212.25.7.69:5060;branch=z9hG4bKsc1NXPTEST-ae749ab817378131A

Content-Length: 0

X-gateway: 

X-SI: 

X-gateway: 

X-SI: 


Initial Inivte to [gw] from [lbl]

INVITE sip:0794567735@185.49.222.43:5060 SIP/2.0

Record-Route: 

From: ;tag=sc1NXPTEST-4c9b51343502af61

To: 

Call-ID: 5773d0ab9b30-5bau50gxp4en

CSeq: 1 INVITE

Allow: INVITE, ACK, CANCEL, BYE, OPTIONS

Max-Forwards: 29

User-Agent: AareSwitch/6.2.8553

Via: SIP/2.0/UDP 
185.49.222.43;branch=z9hG4bKbe7c.a5e4cba8b1d9a4a56d1120d012a06850.0

Via: SIP/2.0/UDP 
212.25.7.69:5060;uac=sc1;branch=z9hG4bKsc1NXPTEST-ae749ab817378131

Contact: 

P-Asserted-Identity: 

Content-Type: application/sdp

Content-Length: 401


Invite sent to [carr] from [gw]

INVITE sip:41794567735@81.7.235.236 SIP/2.0

Record-Route: 

Record-Route: 

From: ;tag=sc1NXPTEST-4c9b51343502af61

To: 

Call-ID: 5773d0ab9b30-5bau50gxp4en

CSeq: 1 INVITE

Allow: INVITE, ACK, CANCEL, BYE, OPTIONS

Max-Forwards: 28

User-Agent: AareSwitch/6.2.8553

Contact: 

Via: SIP/2.0/UDP 
185.49.222.44;branch=z9hG4bKbe7c.0f5ca21e3a66f41694ca709ac28c1192.0

Via: SIP/2.0/UDP 
185.49.222.43;branch=z9hG4bKbe7c.a5e4cba8b1d9a4a56d1120d012a06850.0

Via: SIP/2.0/UDP 
212.25.7.69:5060;uac=sc1;branch=z9hG4bKsc1NXPTEST-ae749ab817378131

P-Asserted-Identity: 

Content-Type: application/sdp





Von: sr-users [mailto:sr-users-boun...@lists.sip-router.org] Im Auftrag von 
Francisco Valentin Vinagrero
Gesendet: Mittwoch, 29. Juni 2016 16:00
An: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.sip-router.org>>
Betreff: Re: [SR-Users] ACK / BYE transaction problem

Hi,

Does the ACK has the correct Router headers and R-URI? Maybe you can share the 
200 OK and the ACK headers..

I had a similar issue 3 weeks ago.

Cheers, Francisco.

From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of 
Oliver Roth
Sent: 29 June 2016 15:55
To: sr-users@lists.sip-router.org<mailto:sr-users@lists.sip-router.org>
Subject: [SR-Users] ACK / BYE transaction problem


Hi all

Follow scenario

Class5 system [c5] --> Loadbalancer kamailio (dispatcher module) [lbl] ---> 
gateway kamailio [gw] --> carrier [carr]

I get Invites from [c5] with
Request ,To, from, contact, pid in national format 0794445566

[lbl] dispatches this to [gw]

For the [carr] I need international format.

So doing these transactions in [gw]
And sending to [carr] in international format

Request, to, from, contact, ... => 417794445566
Everything ok

Then I get a 100, 183 and even 200 from [carr]
Ack is coming

Re: [SR-Users] ACK / BYE transaction problem

2016-06-29 Thread Oliver Roth 
I removed the changes for the to header - so it is not touched all the time



200 ok from [carr]

SIP/2.0 200 OK
From: ;tag=sc1NXPTEST-4c9b51343502af61
To: ;tag=snl_0015024070
Call-ID: 5773d0ab9b30-5bau50gxp4en
CSeq: 1 INVITE
Via: SIP/2.0/UDP 
185.49.222.44;branch=z9hG4bKbe7c.0f5ca21e3a66f41694ca709ac28c1192.0
Via: SIP/2.0/UDP 
185.49.222.43;branch=z9hG4bKbe7c.a5e4cba8b1d9a4a56d1120d012a06850.0
Via: SIP/2.0/UDP 
212.25.7.69:5060;uac=sc1;branch=z9hG4bKsc1NXPTEST-ae749ab817378131
Record-Route: 
Record-Route: 
Contact: 
Content-Type: application/sdp
Content-Length: 197
Accept-Language: en;q=0.0
Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER, UPDATE
Supported: timer
Session-Expires: 1800;refresher=uas
Date: Wed, 29 Jun 2016 13:44:20 GMT

v=0
o=- 277262053 1 IN IP4 81.7.235.228
s=-
c=IN IP4 81.7.235.228
t=0 0
m=audio 24212 RTP/AVP 8 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=ptime:20


ACK from [lbl] to [gw]

ACK sip:185.49.222.44;did=4b7.9422;lr=on SIP/2.0

From: ;tag=sc1NXPTEST-4c9b51343502af61

To: ;tag=snl_0015024070

Call-ID: 5773d0ab9b30-5bau50gxp4en

CSeq: 1 ACK

Max-Forwards: 28

Via: SIP/2.0/UDP 
185.49.222.43;branch=z9hG4bKbe7c.a6f12b66454fbb7b536aa22cef3d568c.0

Via: SIP/2.0/UDP 
185.49.222.43;branch=z9hG4bKbe7c.7bc483a166362b13ba3cc40ca60308ea.0

Via: SIP/2.0/UDP 212.25.7.69:5060;branch=z9hG4bKsc1NXPTEST-ae749ab817378131A

Content-Length: 0

X-gateway: 

X-SI: 

X-gateway: 

X-SI: 


Initial Inivte to [gw] from [lbl]

INVITE sip:0794567735@185.49.222.43:5060 SIP/2.0

Record-Route: 

From: ;tag=sc1NXPTEST-4c9b51343502af61

To: 

Call-ID: 5773d0ab9b30-5bau50gxp4en

CSeq: 1 INVITE

Allow: INVITE, ACK, CANCEL, BYE, OPTIONS

Max-Forwards: 29

User-Agent: AareSwitch/6.2.8553

Via: SIP/2.0/UDP 
185.49.222.43;branch=z9hG4bKbe7c.a5e4cba8b1d9a4a56d1120d012a06850.0

Via: SIP/2.0/UDP 
212.25.7.69:5060;uac=sc1;branch=z9hG4bKsc1NXPTEST-ae749ab817378131

Contact: 

P-Asserted-Identity: 

Content-Type: application/sdp

Content-Length: 401


Invite sent to [carr] from [gw]

INVITE sip:41794567735@81.7.235.236 SIP/2.0

Record-Route: 

Record-Route: 

From: ;tag=sc1NXPTEST-4c9b51343502af61

To: 

Call-ID: 5773d0ab9b30-5bau50gxp4en

CSeq: 1 INVITE

Allow: INVITE, ACK, CANCEL, BYE, OPTIONS

Max-Forwards: 28

User-Agent: AareSwitch/6.2.8553

Contact: 

Via: SIP/2.0/UDP 
185.49.222.44;branch=z9hG4bKbe7c.0f5ca21e3a66f41694ca709ac28c1192.0

Via: SIP/2.0/UDP 
185.49.222.43;branch=z9hG4bKbe7c.a5e4cba8b1d9a4a56d1120d012a06850.0

Via: SIP/2.0/UDP 
212.25.7.69:5060;uac=sc1;branch=z9hG4bKsc1NXPTEST-ae749ab817378131

P-Asserted-Identity: 

Content-Type: application/sdp





Von: sr-users [mailto:sr-users-boun...@lists.sip-router.org] Im Auftrag von 
Francisco Valentin Vinagrero
Gesendet: Mittwoch, 29. Juni 2016 16:00
An: Kamailio (SER) - Users Mailing List 
Betreff: Re: [SR-Users] ACK / BYE transaction problem

Hi,

Does the ACK has the correct Router headers and R-URI? Maybe you can share the 
200 OK and the ACK headers..

I had a similar issue 3 weeks ago.

Cheers, Francisco.

From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of 
Oliver Roth
Sent: 29 June 2016 15:55
To: sr-users@lists.sip-router.org<mailto:sr-users@lists.sip-router.org>
Subject: [SR-Users] ACK / BYE transaction problem


Hi all

Follow scenario

Class5 system [c5] --> Loadbalancer kamailio (dispatcher module) [lbl] ---> 
gateway kamailio [gw] --> carrier [carr]

I get Invites from [c5] with
Request ,To, from, contact, pid in national format 0794445566

[lbl] dispatches this to [gw]

For the [carr] I need international format.

So doing these transactions in [gw]
And sending to [carr] in international format

Request, to, from, contact, ... => 417794445566
Everything ok

Then I get a 100, 183 and even 200 from [carr]
Ack is coming from [c5] to [lbl] and [gw] - but then it stocks

The ACK is not sent to the [carr]

I kamailio log I see
DEBUG: RFC3261 transaction matching failed
DEBUG: t_lookup_request: no transaction found


So for me, the ACK cannot be assigned to a transaction and gets discarded by

if ( is_method("ACK") ) {
   xlog(,"L_INFO", 
"WITHINDLG ACK - not loose route\n");
   if ( 
t_check_trans() ) {
   
xlog(,"L_INFO", "WITHINDLG ACK - t_check_trans() \n");
   
# no loose-route, but stateful ACK;
   
# must be an ACK after a 487
   
# or e.g. 404 from upstream server
   
t_relay();

[SR-Users] ACK / BYE transaction problem

2016-06-29 Thread Oliver Roth 

Hi all

Follow scenario

Class5 system [c5] --> Loadbalancer kamailio (dispatcher module) [lbl] ---> 
gateway kamailio [gw] --> carrier [carr]

I get Invites from [c5] with
Request ,To, from, contact, pid in national format 0794445566

[lbl] dispatches this to [gw]

For the [carr] I need international format.

So doing these transactions in [gw]
And sending to [carr] in international format

Request, to, from, contact, ... => 417794445566
Everything ok

Then I get a 100, 183 and even 200 from [carr]
Ack is coming from [c5] to [lbl] and [gw] - but then it stocks

The ACK is not sent to the [carr]

I kamailio log I see
DEBUG: RFC3261 transaction matching failed
DEBUG: t_lookup_request: no transaction found


So for me, the ACK cannot be assigned to a transaction and gets discarded by

if ( is_method("ACK") ) {
   xlog(,"L_INFO", 
"WITHINDLG ACK - not loose route\n");
   if ( 
t_check_trans() ) {
   
xlog(,"L_INFO", "WITHINDLG ACK - t_check_trans() \n");
   
# no loose-route, but stateful ACK;
   
# must be an ACK after a 487
   
# or e.g. 404 from upstream server
   
t_relay();
   
exit;
   } else {
   
xlog(,"L_INFO", "WITHINDLG ACK - not t_check_trans() DISCARD!!\n");
   
# ACK without matching transaction ... ignore and discard
   
route(NATMANAGE);
   
#t_relay();
   
#exit;



Any idea?

Problem with modifying the sip tags? Or problem with the dialog?


Thanks for helping
OIi



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[SR-Users] Loadbalancer/dispatcher with dialog/cnxcc

2014-04-23 Thread Oliver Roth 
Hi all

Following situation

1 dispatcher/loadbalancer getting all the inbound traffic and sending it to 3 
different gateway (round robin).
The loadbalancer has no (or even very few) business logic.
Just "in" - split to different gateway - "out"

3 sip gateway doing all the business logic like auth, modifications on header, 
different checks, ...
These 3 gateways use the same database (cluster) with routing tables and so on.
A call gets terminated to a carrier through these sip gateway.

Now I would like to implement call limiting (no of calls) by using either only 
dialog module or cnxcc module based on "source ip" or later on "cli".

My problem:
A call from one source ip can be sent to the (3) different sip gateway - so not 
all calls are processed by the same sip gateway.
How can I ensure, that only a certain number of calls are allowed - even if 
they are split up on the 3 different gateway?

Or do I need to implement this kind of business logic on the 
dispatcher/loadbalancer?
That would not make much sense, because this is just a "stupid" machine...

Thanks for helping
Oli
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Re: [SR-Users] Planning release of v4.1.3

2014-04-22 Thread Oliver Roth 
Hi

One point from my side
https://sip-router.org/tracker/index.php?do=details&task_id=353

I did not hear anything from the developers ...

KR,
Oli

-Ursprüngliche Nachricht-
Von: sr-users-boun...@lists.sip-router.org 
[mailto:sr-users-boun...@lists.sip-router.org] Im Auftrag von Daniel-Constantin 
Mierla
Gesendet: Dienstag, 22. April 2014 09:00
An: sr-dev; Kamailio (SER) - Users Mailing List
Betreff: Re: [SR-Users] Planning release of v4.1.3

Hello,

short reminder about upcoming release of v4.1.3. If there are patches left to 
be backported or issues that you are aware, report them.

Cheers,
Daniel

On 16/04/14 12:07, Daniel-Constantin Mierla wrote:
> Hello,
>
> I am considering releasing v4.1.3 by mid of next week, on Wednesday or 
> Thursday (April 23 or 24). If there are issues you are aware of and 
> not reported to the bug tracker, add them there asap to investigate them.
>
> Also, if you noticed some fixes in the master branch not backported 
> yet, report them to the mailing lists.
>
> Cheers,
> Daniel
>

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


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[SR-Users] kamailio / rtpproxy - remove codec

2014-04-06 Thread Oliver Roth 
Hi all

We use kamailio 3.3.7 and rtpproxy for enduser call-termination.

In case of a fax call, we get an invite from our carrier for codecs G711a/u and 
T38.
As our termination carrier does not support T38 and because the invite contains 
G711 and T38 we get back error 488.

How is it possible to remove the whole T38 part of this invite?

We tried
sdp_remove_codecs_by_name(list) without success - what "name" should we use for 
T38? [T38, t38, t.38, T.38, ...]
sdp_remove_line_by_prefix(string)
sdp_remove_media(type)

None of these functions did really work - best was the last one with type=image 
but then the sip header is malformed.

As we saw with Kamailio version 4.1.x there are a lot of new functions within 
sdpops. Would an upgrade help?

So basically the question is:
How to remove the t38 part of the fax invite? (see attachment)

KR,
Oli

[cid:image001.png@01CF523C.988433F0]



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[SR-Users] Module not up to date in apt-get repository

2014-03-19 Thread Oliver Roth 
Hi all

We updated from kamailio 3.1 to kamailio 3.3.
As shown in module description the function rtpproxy_manage() should be 
available starting version 3.2

But if the update is done to 3.3 - we get the following error when trying to 
start kamailio


* Not starting Kamailio SIP server: invalid configuration file!
*
*  0(24558) ERROR:  [cfg.y:3455]: cfg. parser: failed to find command 
rtpproxy_manage
0(24558) :  [cfg.y:3594]: parse error in config file 
/etc/kamailio/kamailio.cfg, line 905, column 17: unknown command, missing 
loadmodule?


In request_route we have the command :

Rtpproxy_manage() ;

Any idea ?


Kind regards,
Oli
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Re: [SR-Users] Carrierroute - scan_prefix

2014-02-03 Thread Oliver Roth 
Hi all

Nobody an idea?

To measure asr/ner ratio this is really important to us!

Regards,
Oli


Von: sr-users-boun...@lists.sip-router.org 
[mailto:sr-users-boun...@lists.sip-router.org] Im Auftrag von Oliver Roth
Gesendet: Montag, 13. Januar 2014 09:26
An: Kamailio (SER) - Users Mailing List
Betreff: [SR-Users] Carrierroute - scan_prefix

Hi all

I need a solution to get the used scan_prefix in carrierroute module.
We have different carriers for our destinations, sometimes split up 50%/50% or 
other splittings - all routes with failure routes.

Now if one destination is not reachable, then I would like to get this 
information and reroute to another carrier.
As we use the carrierroute module, we use the "description" to get the 
carriername (for billing purposes).

Is there a solution to get the used scan_prefix from the carrierroute module?


Regards,
Oli
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[SR-Users] pdd value

2014-01-21 Thread Oliver Roth 
Hi all

Question about an additional cdr information.
We need to get the pdd value - means a time value till the user gets a ringback 
or an error- to track the delay, till a connection is done --> quality issue 
for customer.

5.6.1 Definition of Post Dial Delay
Post Dial Delay (PDD) is experienced by the originating customer as the time 
from the sending of the final dialled digit to the point at which they hear 
ring tone or other in-band information. Where the originating network is 
required to play an announcement before completing the call then this 
definition of PDD excludes the duration of such announcements.

How can this be done in Kamailio?

We use the carrierroute module with different fallback carriers.
Kamailio version 3.3.5


Thanks in advance
Oli
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[SR-Users] Carrierroute - scan_prefix

2014-01-13 Thread Oliver Roth 
Hi all

I need a solution to get the used scan_prefix in carrierroute module.
We have different carriers for our destinations, sometimes split up 50%/50% or 
other splittings - all routes with failure routes.

Now if one destination is not reachable, then I would like to get this 
information and reroute to another carrier.
As we use the carrierroute module, we use the "description" to get the 
carriername (for billing purposes).

Is there a solution to get the used scan_prefix from the carrierroute module?


Regards,
Oli
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Re: [SR-Users] call limit by trunk/user

2013-12-18 Thread Oliver Roth 
Is it possible to run this module with Kamailio version 3.3?

As we are not able to run Kamailio 4.x with carrierroute and Ubuntu 12.04 - we 
have to stay with Kamailio 3.3

Von: sr-users-boun...@lists.sip-router.org 
[mailto:sr-users-boun...@lists.sip-router.org] Im Auftrag von Carlos Ruiz Díaz
Gesendet: Mittwoch, 18. Dezember 2013 13:54
An: Kamailio (SER) - Users Mailing List
Betreff: Re: [SR-Users] call limit by trunk/user

Hi Oliver,

maybe you can use the cnxcc prepaid module [1].

Hopefully, it is what you are looking for.

[1] http://kamailio.org/docs/modules/devel/modules/cnxcc.html#idp130984

Regards,

On Wed, Dec 18, 2013 at 9:42 AM, Oliver Roth 
mailto:oliver.r...@triotel.ch>> wrote:
Hi all

I need a solution to limit the amount of concurrent calls by trunk or 
user/subscriber.

We have the following situation:

Kamailio loadbalancer ==> 3 kamailio gateway with routing/business logic

Kamailio is version 3.3 for all 4 systems.
Loadbalancer has only the absolute minimum config to act as loadbalancert, the 
whole business logic is on gateway side (db connections, carrierroute, ...)

Do you have a suggestion how to limit calls/connections?

Has this to be done on the loadbalancer?
Or is there a solution to do this on the gateways - so every gateway knows the 
currently opened connections?

Thanks for helping
Oli


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--
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http://caruizdiaz.com
+595981146623
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[SR-Users] call limit by trunk/user

2013-12-18 Thread Oliver Roth 
Hi all

I need a solution to limit the amount of concurrent calls by trunk or 
user/subscriber.

We have the following situation:

Kamailio loadbalancer ==> 3 kamailio gateway with routing/business logic

Kamailio is version 3.3 for all 4 systems.
Loadbalancer has only the absolute minimum config to act as loadbalancert, the 
whole business logic is on gateway side (db connections, carrierroute, ...)

Do you have a suggestion how to limit calls/connections?

Has this to be done on the loadbalancer?
Or is there a solution to do this on the gateways - so every gateway knows the 
currently opened connections?

Thanks for helping
Oli

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[SR-Users] rtpproxy / sdp / problem with signalling

2013-11-27 Thread Oliver Roth 
Problem with signalling - RTP gets lost!
Rtpproxy not working properly?

I am absolutely stuck ... cause this happens in a live environement.

I have the following situation

A calls B over carrier 1 - number is not valid and I get back error 404 from 
carrier and now freeswitch should play a message saying: "number not valid".
But from carrier 1 I get back an RTP stream that is useless [1] - and if the 
correct streams opens from freeswitch - this does not get back to A [2].

I tested with rtpproxy on Kamailio - and all the rtp streams arrive at the 
Kamailio - but they cannot be "connected" correctly.

I guess the problem is the 183 I get back from carrier 1 - after whitch rtp is 
opened.
Or there is a wrong sdp singallisation if the "correct" stream arrives [3].

Sorry - I cannot get a solution - but I could provide various tcpdumps and 
pcaps.



A   KamailioCarrier 1   Freeswitch

INVITE
->
100 Your call is important
<-
INVITE
>
100 Trying
<
183 Session Progress SDP
<
183 Session Progress SDP
<
RTP
<=
RTP
<=
RTP
=>
RTP
=>  [1]

404 not found
<---
ACK
>

INVITE

->
100 Trying

<
200 OK SDP

<   [3]
200 OK SDP
<---
RTP (Announcment - number not valid")   [2]
<===
ACK
>
ACK

-->
INFO
>
INFO

-->
200 OK

<
200 OK
<---
BYE
>
BYE

-->
200 OK

<
200 OK
<---

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Re: [SR-Users] Loosing rtp only in carrierroute failureroute

2013-11-26 Thread Oliver Roth 
No idea?

-Ursprüngliche Nachricht-
Von: sr-users-boun...@lists.sip-router.org 
[mailto:sr-users-boun...@lists.sip-router.org] Im Auftrag von Oliver Roth
Gesendet: Dienstag, 29. Oktober 2013 12:50
An: Kamailio (SER) - Users Mailing List
Betreff: [SR-Users] Loosing rtp only in carrierroute failureroute

Hi all 

We do have a strange problem with loosing rtp in case of carrierroute - 
failureroute.
If we send traffic directly to the failure gateway, we do have rtp without any 
problem

Situation:
Ua --> Freeswitch --> kamailio 3.3 --> gw 1 Error 404 --> failureroute --> 
media gateway voiceprompt for error 404 ==> this does not work

Ua --> Freeswitch --> kamailio 3.3 rewrite to dest for voiceprompt 404  
-->media gateway voiceprompt for error 404 ==> this works!!

Any idea?
For me it seems being a problem with branching or so.

Regards,
Oli

Configs below:

Main routing block

request_route {
--- do all the checks

route(CARRIERROUTE);
route(RELAY);
}


route[RELAY] {
xlog(, "L_INFO", "RELAY: Outbound sent via $avp(s:trunk_out)");
# enable additional event routes for forwarded requests
# - serial forking, RTP relaying handling, a.s.o.
if (is_method("INVITE|SUBSCRIBE")) {
t_on_branch("MANAGE_BRANCH");
t_on_reply("MANAGE_REPLY");
xlog(,"L_INFO", "RELAX - Manage branch ...\n");
}
if (is_method("INVITE")) {
t_on_failure("MANAGE_FAILURE");
}

if (!t_relay()) {
sl_reply_error();
}
exit;
}

# carrierroute
route[CARRIERROUTE] {
xlog(,"L_INFO","Carrierroute module start \n");
$avp(s:tree) = "default";

# lookup from user and from domain
cr_user_carrier("$fU", "$avp(s:trunk_in)", "$avp(s:cr_pref_carr)");

if($avp(s:cr_pref_carr) == 0){
# lookup from domain
cr_user_carrier("", "$avp(s:trunk_in)", "$avp(s:cr_pref_carr)");
}   

if(!cr_route("$avp(s:cr_pref_carr)", "$avp(s:tree)", "$rU", "$rU", 
"call_id", "$avp(s:todirection)" )){
xlog("cr_route 403 no route found $avp(s:cr_pref_carr)");
sl_send_reply("403", "Not allowed");
exit;
}
$avp(s:trunk_out) = $avp(s:todirection);
route(ALTERHEADER);

}


failure_route[MANAGE_FAILURE] {

if (t_grep_status("486")){
xlog("L_INFO", "Status 486 - busy");
t_reply("486", "Busy");
}


#revert_uri();
route(NATMANAGE);

if (t_is_canceled()) {
exit;
}

xlog("L_INFO", "failure_route $rd $T_reply_code $avp(s:trunk_in), 
$avp(s:tree), $rU, $rd, $T_reply_code, $avp(s:tree) \n");

if(!cr_next_domain("$avp(s:cr_pref_carr)", "$avp(s:tree)", "$rU", 
"$avp(s:trunk_in)", "$T_reply_code", "$avp(s:tree)" )){
xlog("cr_next_domain failed");
exit;
}

if(!cr_route("$avp(s:cr_pref_carr)", "$avp(s:tree)", "$rU", "$rU", 
"call_id", "$avp(s:todirection)" )){
xlog("cr_route failed");
exit;
}
$avp(s:trunk_out) = $avp(s:todirection);


if (is_method("INVITE"))
{
setflag(FLT_ACC); # do accounting
setflag(FLT_ACCMISSED); # oro 28.10.13
}



route(ALTERHEADER);

t_on_failure("MANAGE_FAILURE");

if (!t_relay()) {
xlog("failureroute t_relay failed");
exit;
}

}

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[SR-Users] voice prompts / early media and kamailio

2013-11-26 Thread Oliver Roth 
Hi all

Based on my problem reported with subject "error 
handling" 
I have some other questions.
I think it is a conceptual question - and I do not see any solution.

I would like to handle Kamailio with carrierroute / carrierfailureroute module 
different errors.
Like 404, 403, busy - or whatever sip error occurs.
Some of them need to be sent to a freeswitch playing an announcement (like 
"this number is blocked", "no more credit", ...).

So if an error occur (lets say 403) then the call is routed by 
carrierfailurroute to fresswitch playing message for 403.

If I am listening the whole message - I get back error 403 at the end and the 
call is logged in missed calls as error 403 sent from the freeswitch - 
everything ok.
If I cancel the listening by hanging up - 487 is stored in missed_calls - cause 
I terminated the call before getting error 403 back from freeswitch.  ==> so I 
loose this important information

How can I get back error 403 - play an  announcement and make sure, it is 
logged as 403 in database?


Thanks for helping ...
Oli
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Re: [SR-Users] error handling

2013-11-24 Thread Oliver Roth 
Sounds good - but I do not know howo to handle that.

Do you have a config example?

Von: sr-users-boun...@lists.sip-router.org 
[mailto:sr-users-boun...@lists.sip-router.org] Im Auftrag von Carsten Bock
Gesendet: Sonntag, 24. November 2013 20:52
An: SR-Users
Betreff: Re: [SR-Users] error handling


You could check, if you can send the announcement as early media (183 Session 
progress) and afterwards decline the call from FreeSwitch. In this case, you 
should not get a 200 Ok from FreeSwitch,  but another SIP reply, which you 
could mofify in yout Kamailio config...

Carsten
Am 24.11.2013 20:44 schrieb "Oliver Roth" 
mailto:oliver.r...@triotel.ch>>:
Sorry - I do not understand what you mean ...

I added the following code at the end of the failure_route - seems to work


if (t_grep_status("404")){
xlog("L_INFO", "ORO Status 404 - not found");
t_reply("404", "not found");
exit;
}


Von: 
sr-users-boun...@lists.sip-router.org<mailto:sr-users-boun...@lists.sip-router.org>
 
[mailto:sr-users-boun...@lists.sip-router.org<mailto:sr-users-boun...@lists.sip-router.org>]
 Im Auftrag von Brandon Armstead
Gesendet: Sonntag, 24. November 2013 20:41
An: Kamailio (SER) - Users Mailing List
Cc: Kamailio (SER) - Users Mailing List
Betreff: Re: [SR-Users] error handling

Acc_db_request

Sent from my iPhone

On Nov 24, 2013, at 11:20 AM, Oliver Roth 
mailto:oliver.r...@triotel.ch>> wrote:
Hi all

Question about error handling with kamailio.

We send call to carrier and get back error 404.
In carrierfailureroute we catch up this error and send call to an internal 
freeswitch that plays a voiceprompt saying: "destination not available"

In accounting this calls is collected like a "normal" call - cause the internal 
freeswitch did the connection.
I would like to play the voiceprompt but get the error 404 and see the call in 
the missed calls acc table.

With 486 (busy) it is simple because we do not need an rtp response ... just 
fast busy.

How can we handle this for errors we need to play a voiceprompt?

What we do in failure route:

failure_route[MANAGE_FAILURE] {

sip_trace();
setflag(22);
if (t_grep_status("486")){
xlog("L_INFO", "Status 486 - busy");
t_reply("486", "Busy");
exit;
}


revert_uri();
route(NATMANAGE);

if (t_is_canceled()) {
exit;
}

xlog("L_INFO", "failure_route rd: $rd replCode:  $T_reply_code 
trunk_in: $avp(s:trunk_in) Tree: $avp(s:tree), rU:  $rU, todirection 
$avp(s:todirection) fu: $fu\n");

if(!cr_next_domain("$avp(s:cr_pref_carr)", "$avp(s:tree)", "$rU", 
"$avp(s:trunk_in)", "$T_reply_code", "$avp(s:tree)" )){
xlog("cr_next_domain failed");
exit;
}

if(!cr_route("$avp(s:cr_pref_carr)", "$avp(s:tree)", "$rU", "$rU", 
"call_id", "$avp(s:todirection)" )){
xlog("cr_route failed");
exit;
}


xlog("L_INFO", "failure_route rd: $rd replCode:  $T_reply_code 
trunk_in: $avp(s:trunk_in) Tree: $avp(s:tree), rU:  $rU, todirection 
$avp(s:todirection) fu: $fu \n");
$avp(s:trunk_out) = $avp(s:todirection);

xlog(, "L_INFO", "RELAY - FailureRoute: Outbound sent via 
$avp(s:trunk_out) rU $rU 
 ");
if (is_method("INVITE"))
{
setflag(FLT_ACC); # do accounting
setflag(FLT_ACCMISSED); # oro 28.10.13
}


route(ALTERHEADER);

t_on_failure("MANAGE_FAILURE");
#append_branch();

if (!t_relay()) {
xlog("failureroute t_relay failed");
exit;
}

}
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Re: [SR-Users] error handling

2013-11-24 Thread Oliver Roth 
Sorry – I do not understand what you mean …

I added the following code at the end of the failure_route – seems to work


if (t_grep_status("404")){
xlog("L_INFO", "ORO Status 404 - not found");
t_reply("404", "not found");
exit;
}


Von: sr-users-boun...@lists.sip-router.org 
[mailto:sr-users-boun...@lists.sip-router.org] Im Auftrag von Brandon Armstead
Gesendet: Sonntag, 24. November 2013 20:41
An: Kamailio (SER) - Users Mailing List
Cc: Kamailio (SER) - Users Mailing List
Betreff: Re: [SR-Users] error handling

Acc_db_request

Sent from my iPhone

On Nov 24, 2013, at 11:20 AM, Oliver Roth 
mailto:oliver.r...@triotel.ch>> wrote:
Hi all

Question about error handling with kamailio.

We send call to carrier and get back error 404.
In carrierfailureroute we catch up this error and send call to an internal 
freeswitch that plays a voiceprompt saying: “destination not available”

In accounting this calls is collected like a “normal” call – cause the internal 
freeswitch did the connection.
I would like to play the voiceprompt but get the error 404 and see the call in 
the missed calls acc table.

With 486 (busy) it is simple because we do not need an rtp response … just fast 
busy.

How can we handle this for errors we need to play a voiceprompt?

What we do in failure route:

failure_route[MANAGE_FAILURE] {

sip_trace();
setflag(22);
if (t_grep_status("486")){
xlog("L_INFO", "Status 486 - busy");
t_reply("486", "Busy");
exit;
}


revert_uri();
route(NATMANAGE);

if (t_is_canceled()) {
exit;
}

xlog("L_INFO", "failure_route rd: $rd replCode:  $T_reply_code 
trunk_in: $avp(s:trunk_in) Tree: $avp(s:tree), rU:  $rU, todirection 
$avp(s:todirection) fu: $fu\n");

if(!cr_next_domain("$avp(s:cr_pref_carr)", "$avp(s:tree)", "$rU", 
"$avp(s:trunk_in)", "$T_reply_code", "$avp(s:tree)" )){
xlog("cr_next_domain failed");
exit;
}

if(!cr_route("$avp(s:cr_pref_carr)", "$avp(s:tree)", "$rU", "$rU", 
"call_id", "$avp(s:todirection)" )){
xlog("cr_route failed");
exit;
}


xlog("L_INFO", "failure_route rd: $rd replCode:  $T_reply_code 
trunk_in: $avp(s:trunk_in) Tree: $avp(s:tree), rU:  $rU, todirection 
$avp(s:todirection) fu: $fu \n");
$avp(s:trunk_out) = $avp(s:todirection);

xlog(, "L_INFO", "RELAY - FailureRoute: Outbound sent via 
$avp(s:trunk_out) rU $rU 
 ");
if (is_method("INVITE"))
{
setflag(FLT_ACC); # do accounting
setflag(FLT_ACCMISSED); # oro 28.10.13
}


route(ALTERHEADER);

t_on_failure("MANAGE_FAILURE");
#append_branch();

if (!t_relay()) {
xlog("failureroute t_relay failed");
exit;
}

}
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[SR-Users] error handling

2013-11-24 Thread Oliver Roth 
Hi all

Question about error handling with kamailio.

We send call to carrier and get back error 404.
In carrierfailureroute we catch up this error and send call to an internal 
freeswitch that plays a voiceprompt saying: "destination not available"

In accounting this calls is collected like a "normal" call - cause the internal 
freeswitch did the connection.
I would like to play the voiceprompt but get the error 404 and see the call in 
the missed calls acc table.

With 486 (busy) it is simple because we do not need an rtp response ... just 
fast busy.

How can we handle this for errors we need to play a voiceprompt?

What we do in failure route:

failure_route[MANAGE_FAILURE] {

sip_trace();
setflag(22);
if (t_grep_status("486")){
xlog("L_INFO", "Status 486 - busy");
t_reply("486", "Busy");
exit;
}


revert_uri();
route(NATMANAGE);

if (t_is_canceled()) {
exit;
}

xlog("L_INFO", "failure_route rd: $rd replCode:  $T_reply_code 
trunk_in: $avp(s:trunk_in) Tree: $avp(s:tree), rU:  $rU, todirection 
$avp(s:todirection) fu: $fu\n");

if(!cr_next_domain("$avp(s:cr_pref_carr)", "$avp(s:tree)", "$rU", 
"$avp(s:trunk_in)", "$T_reply_code", "$avp(s:tree)" )){
xlog("cr_next_domain failed");
exit;
}

if(!cr_route("$avp(s:cr_pref_carr)", "$avp(s:tree)", "$rU", "$rU", 
"call_id", "$avp(s:todirection)" )){
xlog("cr_route failed");
exit;
}


xlog("L_INFO", "failure_route rd: $rd replCode:  $T_reply_code 
trunk_in: $avp(s:trunk_in) Tree: $avp(s:tree), rU:  $rU, todirection 
$avp(s:todirection) fu: $fu \n");
$avp(s:trunk_out) = $avp(s:todirection);

xlog(, "L_INFO", "RELAY - FailureRoute: Outbound sent via 
$avp(s:trunk_out) rU $rU 
 ");
if (is_method("INVITE"))
{
setflag(FLT_ACC); # do accounting
setflag(FLT_ACCMISSED); # oro 28.10.13
}


route(ALTERHEADER);

t_on_failure("MANAGE_FAILURE");
#append_branch();

if (!t_relay()) {
xlog("failureroute t_relay failed");
exit;
}

}
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[SR-Users] Billing

2013-11-18 Thread Oliver Roth 
Hi all

I am looking for a free billing solution usable with Kamailio.
I currently use the sp from siremis - so far it works fine!

I need to get a solution where I can add credit limits for users/trunks or 
credit limits for carriers (eg. If we made a prepayment).
I thing JBilling could be great - but I think the Telco version will cost a 
monthly fee..

Other ideas or solutions?

Btw:
The invoice itself will be made in an ERP system - so I actually just need the 
rating and credit-management to handle the routings.

Regards,
Oli

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[SR-Users] extract part of string / INVITE

2013-11-13 Thread Oliver Roth 
Hi all

I need to extract a part of the INVITE msg:

INVITE sip:+4179615@82.197.185.185;user=CSC10824 SIP/2.0

I need
10824
in a avp variable.

Is there a regex function where I can extract this? Or how can this be done?

Something like:
Avp(myCSC) = string after CSC - length 5
Please be aware, that ip and destination number in the invite can change ...
Otherwise substring would be a solution.

Thanks for help
Oli
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Re: [SR-Users] WG: Carrierroute module

2013-11-11 Thread Oliver Roth 
Thanks for your reply.

Yesterday I tested with compiling Kamailio and the modules on Ubuntu 12.04 lts 
-the compilation worked without errors.
But when trying to load the carrierroute module - the same errors.

So: 
No difference between apt-get installation and compilation

So it seems to be a "Ubuntu" (Debian) problem with libconfuse and carrierroute.

 

-Ursprüngliche Nachricht-
Von: sr-users-boun...@lists.sip-router.org 
[mailto:sr-users-boun...@lists.sip-router.org] Im Auftrag von dotnetdub
Gesendet: Sonntag, 10. November 2013 13:11
An: Kamailio (SER) - Users Mailing List
Betreff: Re: [SR-Users] WG: Carrierroute module

http://sip-router.1086192.n5.nabble.com/Error-loading-carrierroute-in-Kamailio-4-0-x-td122311.html

On 10 November 2013 10:46, Oliver Roth  wrote:
> Nobody an idea how to fix this?
>
>
>
>
>
>
>
> Hi
>
>
>
> I get the following error when trying to load the carrierroute module
>
>
>
> load_module(): ERROR: load_module: could not open module
> :
> /usr/lib64/kamailio/modules/carrierroute.so: undefined symbol:
> cfg_set_error_function
>
>
>
>
>
> I installed Kamailio 4.0.3 by apt-get on Ubuntu 12.04 lts
>
> It seems, that with version 3.3 it is working .
>
>
>
> root@sipgw22:/home/triotel# kamailio -V
>
> version: kamailio 4.0.3 (x86_64/linux)
>
> flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, 
> USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, 
> SHM_MMAP, PKG_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, 
> USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, 
> HAVE_RESOLV_RES
>
> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, 
> MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB
>
> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
>
> id: unknown
>
> compiled on 17:03:55 Aug 19 2013 with gcc 4.6.3
>
>
> ___
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Re: [SR-Users] implemented specifiers not processed

2013-11-11 Thread Oliver Roth 
Sorry - I didn't get that ...

We use Kamailio 3.3.x so xprint is not available (I guess)
The other solution I did not understand.

I need to get the systemname or the system-ip address Kamailio is running on.

- replace % with %, load pv and xlog modules, replace selects @name with 
$sel(name), avps with $avp(name)

Can you make an example how to get the systems ip address / or name?

Thanks in advance
Oli

Von: sr-users-boun...@lists.sip-router.org 
[mailto:sr-users-boun...@lists.sip-router.org] Im Auftrag von Daniel-Constantin 
Mierla
Gesendet: Montag, 11. November 2013 08:54
An: Kamailio (SER) - Users Mailing List
Betreff: Re: [SR-Users] implemented specifiers not processed

Hello,

you are using the (old) SER-style specfiers. There are two options:
- replace % with %, load pv and xlog modules, replace selects @name with 
$sel(name), avps with $avp(name)
- use xprint module and replace xlog() with appropriate new function name

Cheers,
Daniel
On 11/10/13 1:01 PM, Oliver Roth wrote:
Hi all

Having some problems with the following part of the script in a timer:

We have some kamailios running doing more or less the same job - they are used 
by a loadbalancer kamailio.
Actually I wanted to check the ip or systemname of the current system do 
perform some actions depending on the used kamailio - based on a timer

route[CDRS] {
sql_query("ca","call kamailio_cdrs()","rb");
sql_query("ca","call kamailio_rating()","rb");
xlog("timer routine: time is %TF\n");
xlog("L_ERR", "time [%Tf] method <%rm> r-uri <%ru> 2nd via <%{via[1]}>\n");

xlog("L_INFO", "CDRS- rated Host %Hi");

if %Hi = ipOfGateway1 then do something
else if %Hi = ipOfGateway2 then do something different
}


What I get in log

Nov 10 12:54:12 sipgw21 /usr/sbin/kamailio[16231]: ERROR:

[SR-Users] implemented specifiers not processed

2013-11-10 Thread Oliver Roth 
Hi all

Having some problems with the following part of the script in a timer:

We have some kamailios running doing more or less the same job - they are used 
by a loadbalancer kamailio.
Actually I wanted to check the ip or systemname of the current system do 
perform some actions depending on the used kamailio - based on a timer

route[CDRS] {
sql_query("ca","call kamailio_cdrs()","rb");
sql_query("ca","call kamailio_rating()","rb");
xlog("timer routine: time is %TF\n");
xlog("L_ERR", "time [%Tf] method <%rm> r-uri <%ru> 2nd via <%{via[1]}>\n");

xlog("L_INFO", "CDRS- rated Host %Hi");

if %Hi = ipOfGateway1 then do something
else if %Hi = ipOfGateway2 then do something different
}


What I get in log

Nov 10 12:54:12 sipgw21 /usr/sbin/kamailio[16231]: ERROR:

Re: [SR-Users] Regex question

2013-11-03 Thread Oliver Roth 
Got it - thanks for your help.

$avp(s:fromCLI) = $(ai{re.subst,/^sip:(.*)@(.*)/\1/});



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Von: sr-users-boun...@lists.sip-router.org 
[mailto:sr-users-boun...@lists.sip-router.org] Im Auftrag von Juha Heinanen
Gesendet: Sonntag, 3. November 2013 21:58
An: Kamailio (SER) - Users Mailing List
Cc: kamai...@aaronlux.com
Betreff: [SR-Users] Regex question

Oliver Roth writes:

> I need to do some string operations in kamailio.cfg.
> How can I get the cli from the following string:
> sip:+41523940347@195.216.67.103;user=phone
> 
> I only  would need  +41523940347 in a variable
> 
> Something like
> 
> $avp(s:myCli) = ^sip:\+(\d{11})@.{1,40}$
> 
> Any idea?

see if any of these would help:

http://www.kamailio.org/wiki/cookbooks/devel/transformations#uri_transformations

-- juha

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[SR-Users] Regex question

2013-11-03 Thread Oliver Roth 
Hi all

Maybe a very simple question - but I cannot see the solution - I am more or 
less newbie ;)

I need to do some string operations in kamailio.cfg.
How can I get the cli from the following string:
sip:+41523940347@195.216.67.103;user=phone

I only  would need  +41523940347 in a variable

Something like

$avp(s:myCli) = ^sip:\+(\d{11})@.{1,40}$

Any idea?

Thanks in advance
Oli

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[SR-Users] Loosing rtp only in carrierroute failureroute

2013-10-29 Thread Oliver Roth 
Hi all 

We do have a strange problem with loosing rtp in case of carrierroute - 
failureroute.
If we send traffic directly to the failure gateway, we do have rtp without any 
problem

Situation:
Ua --> Freeswitch --> kamailio 3.3 --> gw 1 Error 404 --> failureroute --> 
media gateway voiceprompt for error 404
==> this does not work

Ua --> Freeswitch --> kamailio 3.3 rewrite to dest for voiceprompt 404  
-->media gateway voiceprompt for error 404
==> this works!!

Any idea?
For me it seems being a problem with branching or so.

Regards,
Oli

Configs below:

Main routing block

request_route {
--- do all the checks

route(CARRIERROUTE);
route(RELAY);
}


route[RELAY] {
xlog(, "L_INFO", "RELAY: Outbound sent via $avp(s:trunk_out)");
# enable additional event routes for forwarded requests
# - serial forking, RTP relaying handling, a.s.o.
if (is_method("INVITE|SUBSCRIBE")) {
t_on_branch("MANAGE_BRANCH");
t_on_reply("MANAGE_REPLY");
xlog(,"L_INFO", "RELAX - Manage branch ...\n");
}
if (is_method("INVITE")) {
t_on_failure("MANAGE_FAILURE");
}

if (!t_relay()) {
sl_reply_error();
}
exit;
}

# carrierroute
route[CARRIERROUTE] {
xlog(,"L_INFO","Carrierroute module start \n");
$avp(s:tree) = "default";

# lookup from user and from domain
cr_user_carrier("$fU", "$avp(s:trunk_in)", "$avp(s:cr_pref_carr)");

if($avp(s:cr_pref_carr) == 0){
# lookup from domain
cr_user_carrier("", "$avp(s:trunk_in)", "$avp(s:cr_pref_carr)");
}   

if(!cr_route("$avp(s:cr_pref_carr)", "$avp(s:tree)", "$rU", "$rU", 
"call_id", "$avp(s:todirection)" )){
xlog("cr_route 403 no route found $avp(s:cr_pref_carr)");
sl_send_reply("403", "Not allowed");
exit;
}
$avp(s:trunk_out) = $avp(s:todirection);
route(ALTERHEADER);

}


failure_route[MANAGE_FAILURE] {

if (t_grep_status("486")){
xlog("L_INFO", "Status 486 - busy");
t_reply("486", "Busy");
}


#revert_uri();
route(NATMANAGE);

if (t_is_canceled()) {
exit;
}

xlog("L_INFO", "failure_route $rd $T_reply_code $avp(s:trunk_in), 
$avp(s:tree), $rU, $rd, $T_reply_code, $avp(s:tree) \n");

if(!cr_next_domain("$avp(s:cr_pref_carr)", "$avp(s:tree)", "$rU", 
"$avp(s:trunk_in)", "$T_reply_code", "$avp(s:tree)" )){
xlog("cr_next_domain failed");
exit;
}

if(!cr_route("$avp(s:cr_pref_carr)", "$avp(s:tree)", "$rU", "$rU", 
"call_id", "$avp(s:todirection)" )){
xlog("cr_route failed");
exit;
}
$avp(s:trunk_out) = $avp(s:todirection);


if (is_method("INVITE"))
{
setflag(FLT_ACC); # do accounting
setflag(FLT_ACCMISSED); # oro 28.10.13
}



route(ALTERHEADER);

t_on_failure("MANAGE_FAILURE");

if (!t_relay()) {
xlog("failureroute t_relay failed");
exit;
}

}

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