[SR-Users] Configuration Issue on Kamailio.
Laura
red_dra at plugit.net
Wed Aug 10 14:35:07 CEST 2016
Ciao Daniel,
here the data you request..
Of course Kamailio2 use the color 9990 to send the call to CISCO Gw
because its a required.. so CISCO send back the call with 9990 to
Kamailio2 and it to Kamailio1 and after that to Customer..
What I espect was that CISCO replied 9990 to Kamailio2, Kama2 replied
9053 to Kamailio1 and Kamailio1 replied 9999 to Customer1.
Here the sip trace you request
Kamailio1 --> Kamailio2
U 2016/08/10 10:54:29.269917 2.2.2.2:5060 -> 3.3.3.3:5060
INVITE sip:90534912345678 at 3.3.3.3 SIP/2.0.
Record-Route: <sip:2.2.2.2;lr;did=4f3.8501;nat=yes>.
Via: SIP/2.0/UDP
2.2.2.2:5060;branch=z9hG4bK5c29.8288fcc6bf463fb8f82d3609b0c4893c.0.
Via: SIP/2.0/UDP
1.1.1.1:5060;received=1.1.1.1;branch=z9hG4bK06b62a40;rport=5060.
Max-Forwards: 69.
From: "151512345678" <sip:151512345678 at 1.1.1.1>;tag=as7f0dee78.
To: <sip:90534912345678 at 3.3.3.3>.
Contact: <sip:151512345678 at 1.1.1.1:5060>.
Call-ID: 406a307158b1016b3a9936cd476b3c89 at 1.1.1.1:5060.
CSeq: 102 INVITE.
Date: Wed, 10 Aug 2016 08:54:27 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length: 308.
User-Agent: Fagians VOIP 2.4.
.
v=0.
o=root 869935480 869935480 IN IP4 1.1.1.1.
s=Asterisk PBX 1.8.32.3.
c=IN IP4 x.x.x.x.x.
t=0 0.
m=audio 36398 RTP/AVP 3 18 8 101.
a=rtpmap:3 GSM/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.
Kamailio2 --> CISCO.. LCR need to use 9990 to send call to CISCO..
U 2016/08/10 10:54:29.301085 3.3.3.3:5060 -> 4.4.4.4:5060
INVITE sip:99904912345678 at 4.4.4.4 SIP/2.0.
Record-Route: <sip:3.3.3.3;lr;did=4f3.9ce2;nat=yes>.
Record-Route: <sip:2.2.2.2;lr;did=4f3.8501;nat=yes>.
Via: SIP/2.0/UDP
3.3.3.3:5060;branch=z9hG4bK5c29.c1bcba318da7a0a1ae45efbe2ee682c5.0.
Via: SIP/2.0/UDP
2.2.2.2:5060;rport=5060;branch=z9hG4bK5c29.8288fcc6bf463fb8f82d3609b0c4893c.0.
Via: SIP/2.0/UDP
1.1.1.1:5060;received=1.1.1.1;branch=z9hG4bK06b62a40;rport=5060.
Max-Forwards: 68.
From: "151512345678" <sip:151512345678 at 1.1.1.1>;tag=as7f0dee78.
To: <sip:99904912345678 at 4.4.4.4>.
Contact: <sip:151512345678 at 1.1.1.1:5060>.
Call-ID: 406a307158b1016b3a9936cd476b3c89 at 1.1.1.1:5060.
CSeq: 102 INVITE.
Date: Wed, 10 Aug 2016 08:54:27 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length: 309.
User-Agent: Fagians VOIP 2.4.
.
v=0.
o=root 869935480 869935480 IN IP4 1.1.1.1.
s=Asterisk PBX 1.8.32.3.
c=IN IP4 x.x.x.x..
t=0 0.
m=audio 58242 RTP/AVP 3 18 8 101.
a=rtpmap:3 GSM/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.
CISCO ---> Kamailio2 180/200 messages
U 2016/08/10 10:54:29.361634 4.4.4.4:5060 -> 3.3.3.3:5060
SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP
3.3.3.3:5060;branch=z9hG4bK5c29.c1bcba318da7a0a1ae45efbe2ee682c5.0,SIP/2.0/UDP
2.2.2.2:5060;rport=5060;branch=z9hG4bK5c29.8288fcc6bf463fb8f82d3609b0c4893c.0,SIP/2.0/UDP
1.1.1.1:5060;received=1.1.1.1;branch=z9hG4bK06b62a40;rport=5060.
From: "151512345678" <sip:151512345678 at 1.1.1.1>;tag=as7f0dee78.
To: <sip:99904912345678 at 4.4.4.4>;tag=5F0E7DF4-172F.
Date: Wed, 10 Aug 2016 08:54:29 GMT.
Call-ID: 406a307158b1016b3a9936cd476b3c89 at 1.1.1.1:5060.
Server: Cisco-SIPGateway/IOS-12.x.
CSeq: 102 INVITE.
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER.
Allow-Events: telephone-event.
Contact: <sip:99904912345678 at 4.4.4.4:5060>.
Record-Route:
<sip:3.3.3.3;lr;did=4f3.9ce2;nat=yes>,<sip:2.2.2.2;lr;did=4f3.8501;nat=yes>.
Content-Disposition: session;handling=required.
Content-Type: application/sdp.
Content-Length: 249.
.
v=0.
o=CiscoSystemsSIP-GW-UserAgent 9803 1571 IN IP4 4.4.4.4.
s=SIP Call.
c=IN IP4 x.x.x.x.
t=0 0.
m=audio 18838 RTP/AVP 3 101.
c=IN IP4 83.147.65.249.
a=rtpmap:3 GSM/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:10.
U 2016/08/10 10:54:39.486505 4.4.4.4:5060 -> 3.3.3.3:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP
3.3.3.3:5060;branch=z9hG4bK5c29.c1bcba318da7a0a1ae45efbe2ee682c5.0,SIP/2.0/UDP
2.2.2.2:5060;rport=5060;branch=z9hG4bK5c29.8288fcc6bf463fb8f82d3609b0c4893c.0,SIP/2.0/UDP
185.24.22
0.141:5060;received=1.1.1.1;branch=z9hG4bK06b62a40;rport=5060.
From: "151512345678" <sip:151512345678 at 1.1.1.1>;tag=as7f0dee78.
To: <sip:99904912345678 at 4.4.4.4>;tag=5F0E7DF4-172F.
Date: Wed, 10 Aug 2016 08:54:29 GMT.
Call-ID: 406a307158b1016b3a9936cd476b3c89 at 1.1.1.1:5060.
Server: Cisco-SIPGateway/IOS-12.x.
CSeq: 102 INVITE.
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER.
Supported: replaces.
Allow-Events: telephone-event.
Contact: <sip:99904912345678 at 4.4.4.4:5060>.
Record-Route:
<sip:3.3.3.3;lr;did=4f3.9ce2;nat=yes>,<sip:2.2.2.2;lr;did=4f3.8501;nat=yes>.
Content-Type: application/sdp.
Content-Length: 249.
.
v=0.
o=CiscoSystemsSIP-GW-UserAgent 9803 1571 IN IP4 4.4.4.4.
s=SIP Call.
c=IN IP4 x.x.x.x.
t=0 0.
m=audio 18838 RTP/AVP 3 101.
c=IN IP4 83.147.65.249.
a=rtpmap:3 GSM/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:10.
Il 10/08/16 13:33, Daniel Grotti ha scritto:
> Ciao Laura,
> would be interesting to see the INVITE from kamailo2-->Cisco and see
> the headers there, as well as the 180/200 from Cisco->kamailio2.
> As Carsten said, probably Cisco is messing up From/To headers. The
> 9990 color is not present in any of the INVITEs you provided, so would
> be nice to understand where is come from.
>
>
> Cheers,
> Daniel
>
>
>
> On 08/10/2016 12:27 PM, Laura wrote:
>>
>> Sorry for delay on my reply..
>>
>>
>> I need to expalin better the situazione..
>>
>> Customer1 Ip : 1.1.1.1
>> Kamailio1 ip : 2.2.2.2
>> Kamailio2 ip: 3.3.3.3
>> CiscoGW ip: 4.4.4.4
>>
>> Kamailio1 is on USA for example
>> Kamailio2 is on Germany for example
>>
>> Customer1 --> Kamailio platform1 --> Kamailio Platform2 --> CISCO GW
>> SIP/TDM for PTSN termination
>>
>> Customer1 is sending a call using his specific color 9999 to number
>> 4912345678 and from sender 151512345678
>>
>> U 2016/08/10 09:54:29.250974 1.1.1.1:5060 ->2.2.2.2:5060
>> INVITE sip:*9999*4912345678 at 2.2.2.2 SIP/2.0.
>> Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK06b62a40;rport.
>> Max-Forwards: 70.
>> From: "151512345678" <sip:151512345678 at 1.1.1.1>;tag=as7f0dee78.
>> To: <sip:*9999*4912345678 at 2.2.2.2>.
>> Contact: <sip:151512345678 at 1.1.1.1:5060>.
>> Call-ID: 406a307158b1016b3a9936cd476b3c89 at 1.1.1.1:5060.
>> CSeq: 102 INVITE.
>> User-Agent: Asterisk PBX 1.8.32.3.
>> Date: Wed, 10 Aug 2016 08:54:27 GMT.
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
>> INFO, PUBLISH, MESSAGE.
>> Supported: replaces, timer.
>> Content-Type: application/sdp.
>> Content-Length: 309.
>> .
>> v=0.
>> o=root 869935480 869935480 IN IP4 1.1.1.1.
>> s=Asterisk PBX 1.8.32.3.
>> c=IN IP4 1.1.1.1.
>> t=0 0.
>> m=audio 15710 RTP/AVP 3 18 8 101.
>> a=rtpmap:3 GSM/8000.
>> a=rtpmap:18 G729/8000.
>> a=fmtp:18 annexb=no.
>> a=rtpmap:8 PCMA/8000.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-16.
>> a=ptime:20.
>> a=sendrecv.
>>
>>
>> After that the Kamailio1 platform is checking the LCR and route it
>> with the color of its supplier (9053) to Kamailio2. Kamailio2 is a
>> supplier of Kamailio1
>>
>> U 2016/08/10 09:54:29.2525272.2.2.2:5060 -> 3.3.3.3:5060
>> INVITE sip:*9053*4912345678 at 3.3.3.3 SIP/2.0.
>> Record-Route: <sip:2.2.2.2;lr;did=4f3.8501;nat=yes>.
>> Via:
>> SIP/2.0/UDP2.2.2.2:5060;branch=z9hG4bK5c29.8288fcc6bf463fb8f82d3609b0c4893c.0.
>> Via: SIP/2.0/UDP
>> 1.1.1.1:5060;received=1.1.1.1;branch=z9hG4bK06b62a40;rport=5060.
>> Max-Forwards: 69.
>> From: "151512345678" <sip:151512345678 at 1.1.1.1>;tag=as7f0dee78.
>> To: <sip:*9053*4912345678 at 3.3.3.3>.
>> Contact: <sip:151512345678 at 1.1.1.1:5060>.
>> Call-ID: 406a307158b1016b3a9936cd476b3c89 at 1.1.1.1:5060.
>> CSeq: 102 INVITE.
>> Date: Wed, 10 Aug 2016 08:54:27 GMT.
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
>> INFO, PUBLISH, MESSAGE.
>> Supported: replaces, timer.
>> Content-Type: application/sdp.
>> Content-Length: 308.
>> User-Agent: Fagians VOIP 2.4.
>> .
>> v=0.
>> o=root 869935480 869935480 IN IP4 1.1.1.1.
>> s=Asterisk PBX 1.8.32.3.
>> c=IN IP4 51.254.158.37.
>> t=0 0.
>> m=audio 36398 RTP/AVP 3 18 8 101.
>> a=rtpmap:3 GSM/8000.
>> a=rtpmap:18 G729/8000.
>> a=fmtp:18 annexb=no.
>> a=rtpmap:8 PCMA/8000.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-16.
>> a=ptime:20.
>> a=sendrecv.
>>
>> Kamailio2 use its LCR and send the call to Cisco Gateway that use its
>> color and send the call on termination to TDM Switch.
>> Naturally Kamailio2 receive the replies from Cisco and send it back
>> to Kamailio1.
>>
>>
>> Here is the Session progress Kamailio1 receive from Kamailio2 that it
>> got from Cisco.
>>
>> U 2016/08/10 09:54:29.375669 3.3.3.3:5060 ->2.2.2.2:5060
>> SIP/2.0 183 Session Progress.
>> Via:
>> SIP/2.0/UDP2.2.2.2:5060;rport=5060;branch=z9hG4bK5c29.8288fcc6bf463fb8f82d3609b0c4893c.0,SIP/2.0/UDP
>> 1.1.1.1:5060;received=1.1.1.1;branch=z9hG4bK06b62a40;rport=5060.
>> From: "151512345678" <sip:151512345678 at 1.1.1.1>;tag=as7f0dee78.
>> To: <sip:*9990*4912345678 at 4.4.4.4>;tag=5F0E7DF4-172F.
>> Date: Wed, 10 Aug 2016 08:54:29 GMT.
>> Call-ID: 406a307158b1016b3a9936cd476b3c89 at 1.1.1.1:5060.
>> CSeq: 102 INVITE.
>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
>> SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER.
>> Allow-Events: telephone-event.
>> Contact: <sip:99904912345678 at 4.4.4.4:5060>.
>> Record-Route:
>> <sip:3.3.3.3;lr;did=4f3.9ce2;nat=yes>,<sip:2.2.2.2;lr;did=4f3.8501;nat=yes>.
>> Content-Disposition: session;handling=required.
>> Content-Type: application/sdp.
>> Content-Length: 251.
>> User-Agent: Fagians VOIP 2.4.
>> .
>> v=0.
>> o=CiscoSystemsSIP-GW-UserAgent 9803 1571 IN IP4 4.4.4.4.
>> s=SIP Call.
>> c=IN IP4 83.147.127.247.
>> t=0 0.
>> m=audio 58240 RTP/AVP 3 101.
>> c=IN IP4 83.147.127.247.
>> a=rtpmap:3 GSM/8000.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-16.
>> a=ptime:10.
>>
>> To: <sip:99904912345678 at 4.4.4.4>;tag=5F0E7DF4-172F. ->> 9990 is the
>> color that use CISCO to terminate the call on TDM Switch
>>
>> After some other messages Kamailio1 receive the 200 OK and send it
>> back to Customer1
>>
>>
>> Kamailio2 --> Kamailio1
>>
>> U 2016/08/10 09:54:39.507885 3.3.3.3:5060 ->2.2.2.2:5060
>> SIP/2.0 200 OK.
>> Via:
>> SIP/2.0/UDP2.2.2.2:5060;rport=5060;branch=z9hG4bK5c29.8288fcc6bf463fb8f82d3609b0c4893c.0,SIP/2.0/UDP
>> 1.1.1.1:5060;received=1.1.1.1;branch=z9hG4bK06b62a40;rport=5060.
>> From: "151512345678" <sip:151512345678 at 1.1.1.1>;tag=as7f0dee78.
>> To: <sip:*9990*4912345678 at 4.4.4.4>;tag=5F0E7DF4-172F.
>> Date: Wed, 10 Aug 2016 08:54:29 GMT.
>> Call-ID: 406a307158b1016b3a9936cd476b3c89 at 1.1.1.1:5060.
>> CSeq: 102 INVITE.
>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
>> SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER.
>> Supported: replaces.
>> Allow-Events: telephone-event.
>> Contact: <sip:99904912345678 at 4.4.4.4:5060>.
>> Record-Route:
>> <sip:3.3.3.3;lr;did=4f3.9ce2;nat=yes>,<sip:2.2.2.2;lr;did=4f3.8501;nat=yes>.
>> Content-Type: application/sdp.
>> Content-Length: 251.
>> User-Agent: Fagians VOIP 2.4.
>> .
>> v=0.
>> o=CiscoSystemsSIP-GW-UserAgent 9803 1571 IN IP4 4.4.4.4.
>> s=SIP Call.
>> c=IN IP4 83.147.127.247.
>> t=0 0.
>> m=audio 58240 RTP/AVP 3 101.
>> c=IN IP4 83.147.127.247.
>> a=rtpmap:3 GSM/8000.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-16.
>> a=ptime:10.
>>
>> Kamailio1 --> Customer1
>>
>> U 2016/08/10 09:54:39.5120362.2.2.2:5060 -> 1.1.1.1:5060
>> SIP/2.0 200 OK.
>> Via: SIP/2.0/UDP
>> 1.1.1.1:5060;received=1.1.1.1;branch=z9hG4bK06b62a40;rport=5060.
>> From: "151512345678" <sip:151512345678 at 1.1.1.1>;tag=as7f0dee78.
>> To: <sip:*9990*4912345678 at 4.4.4.4>;tag=5F0E7DF4-172F.
>> Date: Wed, 10 Aug 2016 08:54:29 GMT.
>> Call-ID: 406a307158b1016b3a9936cd476b3c89 at 1.1.1.1:5060.
>> CSeq: 102 INVITE.
>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
>> SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER.
>> Supported: replaces.
>> Allow-Events: telephone-event.
>> Contact: <sip:99904912345678 at 4.4.4.4:5060>.
>> Record-Route:
>> <sip:3.3.3.3;lr;did=4f3.9ce2;nat=yes>,<sip:2.2.2.2;lr;did=4f3.8501;nat=yes>.
>> Content-Type: application/sdp.
>> Content-Length: 249.
>> User-Agent: Fagians VOIP 2.4.
>> .
>> v=0.
>> o=CiscoSystemsSIP-GW-UserAgent 9803 1571 IN IP4 4.4.4.4.
>> s=SIP Call.
>> c=IN IP4 51.254.158.37.
>> t=0 0.
>> m=audio 56710 RTP/AVP 3 101.
>> c=IN IP4 51.254.158.37.
>> a=rtpmap:3 GSM/8000.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-16.
>> a=ptime:10.
>>
>> So the real question is how to fix that on Kamailio ?..
>>
>> We need to use always the original messages and data into sdp header
>> when we talk with other parts..
>>
>> On our configuration we permit to transit that modified messages..
>> like you can see Customer1 is getting back datas modified from CiscoGW.
>>
>>
>> Hope that will be more clear to you all..
>>
>>
>> Anyone can suggest us a way ?
>>
>>
>> Regards
>>
>> Laura
>>
>>
>> Il 01/08/16 14:25, Carsten Bock ha scritto:
>>> Hi,
>>>
>>> do you use "uac_replace_from" or "uac_replace_to" in your logic?
>>>
>>> If not, it seems to me, that your supplier is messing around with
>>> the SIP-Replies.
>>>
>>> Thanks,
>>> Carsten
>>>
>>> 2016-08-01 14:10 GMT+02:00 Laura <red_dra at plugit.net
>>> <mailto:red_dra at plugit.net>>:
>>>
>>> Dear list,
>>>
>>> i'm asking here a question about Kamailio config.
>>>
>>> We are testing a wide area configuration of Kamailio over separates
>>> countries and we are still facing with an issue.
>>>
>>> We configured Kamailio 4.3.5 with dialog support over the TM
>>> modules and
>>> we use LCR module for menage ours LCRs rule set profiles.
>>>
>>> For some technicals reasons we use tech prefix for our customer
>>> so for
>>> exaples customer1 send traffic to us with 1111 prefix, customer2
>>> send
>>> traffic to us with 2222 and something similar..
>>>
>>> Our supplier, of course, are using tech prefix too so for
>>> examples if i
>>> want to send the call to supplier1 i need to use tech prefix 1789 or
>>> something similar..
>>>
>>> The point is..
>>>
>>>
>>> When customer1 is sending an invite to us.. it send us something
>>> like
>>> (Bangladesh mobile 8801xxx)
>>>
>>> INVITE sip:11118801xxxxxxx at aaa.bbb.ccc.ddd
>>>
>>> Our Kamailio will reply with the Trying and then it goes to LCR
>>> module
>>> and match our supplier1 so it make a new invite like this
>>>
>>> INVITE sip:17898801xxxxxx at supplier.ip
>>>
>>> The problem come when supplier1 reply to us and we replies back to
>>> customer1..
>>>
>>> Customer1 view the From: field with the 17898801xxxxxx numbers.. and
>>> some of our customers don't like it.
>>>
>>> We don't use anymore the topoh module becuase we found some troubles
>>> using it.. so..
>>>
>>> Is there a way that we can use for fix this situation ?
>>>
>>>
>>> Best regards.
>>>
>>>
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
>>> mailing list
>>> sr-users at lists.sip-router.org <mailto:sr-users at lists.sip-router.org>
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>>
>>>
>>> --
>>> Carsten Bock
>>> CEO (Geschäftsführer)
>>>
>>> ng-voice GmbH
>>> Millerntorplatz 1
>>> 20359 Hamburg / Germany
>>>
>>> http://www.ng-voice.com
>>> mailto:carsten at ng-voice.com <mailto:carsten at ng-voice.com>
>>>
>>> Office +49 40 5247593-40
>>> Fax +49 40 5247593-99
>>>
>>> Sitz der Gesellschaft: Hamburg
>>> Registergericht: Amtsgericht Hamburg, HRB 120189
>>> Geschäftsführer: Carsten Bock
>>> Ust-ID: DE279344284
>>>
>>> Hier finden Sie unsere handelsrechtlichen Pflichtangaben:
>>> http://www.ng-voice.com/imprint/
>>>
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> sr-users at lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
> _______________________________________________
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