[SR-Users] Kamailio 3.2, NAT, rtpproxy bridging between interfaces

Daniel-Constantin Mierla miconda at gmail.com
Wed Sep 12 16:12:46 CEST 2012


Hello,

are the SDP bodies full in your traces? I don't see the 'a=' rtpproxy 
marker line.

You should not give anymore the second parameter to rtpproxy_manage() if 
the rtpproxy is in bridging mode and you use flags I, E.

Also, I don't see the logs from kamailio, you added some xlog() lines in 
your config...

Cheers,
Daniel

On 9/11/12 3:12 PM, Russell McConnachie wrote:
> Hello,
>
> I'm having some issues getting media bridging working in kamailio and 
> rtpproxy. I've attempted to attach as much relevant information as 
> possible. Thanks in advance for taking a look; I'm completely stuck 
> right now.
>
> Actual results:No audio is being passed through the rtpproxy from 
> caller "A" to caller "B" even though both media server and UAC are 
> sending media to the rtpproxy.
> Expected results:Audio that is sent to rtpproxy is bridged/proxied 
> across from caller "A" to caller "B" and vice-versa.
>
> ------------------------------------------------------------------------------- 
>
> Signaling trace:
> ------------------------------------------------------------------------------- 
>
>
> 08:34:57.543224 IP (tos 0x0, ttl  49, id 3237, offset 0, flags [none], 
> proto: UDP (17), length: 1456) x.x.x.199.31280 > x.x.x.102.sip: SIP, 
> length: 1428
>         INVITE sip:9991 at r18.test-bwks.local:5060 SIP/2.0
>         Via: SIP/2.0/UDP 
> 192.168.2.250:51620;branch=z9hG4bK1158b9f850c008ff7.6835b1eb6e3bffee5
>         Route: <sip:x.x.x.102:5060;lr>
>         Max-Forwards: 70
>         From: "99999999" 
> <sip:99999999 at r18.test-bwks.local:5060>;tag=1cf0b02fab
>         To: "9991" <sip:9991 at r18.test-bwks.local:5060>
>         Call-ID: b2af8b83e20b0740
>         CSeq: 29663 INVITE
>         Allow:  INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, 
> UPDATE, PRACK, SUBSCRIBE, INFO
>         Allow-Events: talk, hold, conference, LocalModeStatus
>         Contact: "99999999" 
> <sip:99999999 at 192.168.2.250:51620;transport=udp;srcadr=192.168.2.250:5060>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D2DF3BB>"
>         Supported: gruu, path, timer, 100rel, replaces
>         User-Agent: Aastra 57i/2.6.0.1007
>         Content-Type: application/sdp
>         Content-Length: 619
>
>         v=0
>         o=MxSIP 0 0 IN IP4 192.168.2.250
>         s=SIP Call
>         c=IN IP4 192.168.2.250
>         t=0 0
>         m=audio 51720 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 
> 96 9 8 101
>         a=rtpmap:0 PCMU/8000
>         a=rtpmap:18 G729/8000
>         a=rtpmap:106 BV16/8000
>         a=rtpmap:107 BV32/16000
>         a=rtpmap:113 L16/16000
>         a=rtpmap:110 PCMU/16000
>         a=rtpmap:111 PCMA/16000
>         a=rtpmap:112 L16/8000
>         a=rtpmap:98 G726-16/8000
>         a=rtpmap:97 G726-24/8000
>         a=rtpmap:115 G726-32/8000
>         a=rtpmap:96 G726-40/8000
>         a=rtpmap:9 G722/8000
>         a=rtpmap:8 PCMA/8000
>         a=rtpmap:101 telephone-event/8000
>         a=silenceSupp:on - - - -
>         a=fmtp:18 annexb=yes
>         a=fmtp:101 0-15
>         a=ptime:30
>         a=sendrecv
>
> 08:34:57.545766 IP (tos 0x10, ttl  64, id 58136, offset 0, flags 
> [none], proto: UDP (17), length: 439) x.x.x.102.sip > x.x.x.199.31280: 
> SIP, length: 411
>         SIP/2.0 100 trying -- your call is important to us
>         Via: SIP/2.0/UDP 
> 192.168.2.250:51620;branch=z9hG4bK1158b9f850c008ff7.6835b1eb6e3bffee5;rport=31280;received=x.x.x.199
>         From: "99999999" 
> <sip:99999999 at r18.test-bwks.local:5060>;tag=1cf0b02fab
>         To: "9991" <sip:9991 at r18.test-bwks.local:5060>
>         Call-ID: b2af8b83e20b0740
>         CSeq: 29663 INVITE
>         Server: kamailio (3.2.4 (x86_64/linux))
>         Content-Length: 0
>
> 08:34:57.547619 IP (tos 0x10, ttl  64, id 47936, offset 0, flags [+], 
> proto: UDP (17), length: 1500) 10.0.0.38.sip > 10.0.0.29.sip: SIP, 
> length: 1472
>         INVITE sip:9991 at 10.0.0.29 SIP/2.0
>         Record-Route: <sip:10.0.0.38;r2=on;lr=on;nat=yes>
>         Record-Route: <sip:x.x.x.102;r2=on;lr=on;nat=yes>
>         Via: SIP/2.0/UDP 10.0.0.38;branch=z9hG4bK4ed5.ec1e41a.0
>         Via: SIP/2.0/UDP 
> 192.168.2.250:51620;rport=31280;received=x.x.x.199;branch=z9hG4bK1158b9f850c008ff7.6835b1eb6e3bffee5
>         Max-Forwards: 69
>         From: "99999999" 
> <sip:99999999 at r18.test-bwks.local:5060>;tag=1cf0b02fab
>         To: "9991" <sip:9991 at r18.test-bwks.local:5060>
>         Call-ID: b2af8b83e20b0740
>         CSeq: 29663 INVITE
>         Allow:  INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, 
> UPDATE, PRACK, SUBSCRIBE, INFO
>         Allow-Events: talk, hold, conference, LocalModeStatus
>         Contact: "99999999" 
> <sip:99999999 at x.x.x.199:31280;transport=udp;srcadr=192.168.2.250:5060>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D2DF3BB>"
>         Supported: gruu, path, timer, 100rel, replaces
>         User-Agent: Aastra 57i/2.6.0.1007
>         Content-Type: application/sdp
>         Content-Length: 629
>
>         v=0
>         o=MxSIP 0 0 IN IP4 10.0.0.38
>         s=SIP Call
>         c=IN IP4 10.0.0.38
>         t=0 0
>         m=audio 29682 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 
> 96 9 8 101
>         a=rtpmap:0 PCMU/8000
>         a=rtpmap:18 G729/8000
>         a=rtpmap:106 BV16/8000
>         a=rtpmap:107 BV32/16000
>         a=rtpmap:113 L16/16000
>         a=rtpmap:110 PCMU/16000
>         a=rtpmap:111 PCMA/16000
>         a=rtpmap:112 L16/8000
>         a=rtpmap:98 G726-16/8000
>         a=rtpmap:97 G726-24/8000
>         a=rtpmap:115 G726-32/8000
>         a=rtpmap:96 G726-40/8000
>         a=rtpmap:9 G722/8000
>         a=rtpmap:8 PCMA/8000
>         a=rtpmap:101 telephone-event
>
> 08:34:57.551126 IP (tos 0x0, ttl  64, id 0, offset 0, flags [DF], 
> proto: UDP (17), length: 409) 10.0.0.29.55620 > 10.0.0.38.sip: SIP, 
> length: 381
>         SIP/2.0 100 Trying
>         Via:SIP/2.0/UDP 
> 10.0.0.38;branch=z9hG4bK4ed5.ec1e41a.0,SIP/2.0/UDP 
> 192.168.2.250:51620;received=x.x.x.199;branch=z9hG4bK1158b9f850c008ff7.6835b1eb6e3bffee5;rport=31280
> From:"99999999"<sip:99999999 at r18.test-bwks.local:5060>;tag=1cf0b02fab
>         To:"9991"<sip:9991 at r18.test-bwks.local:5060>
>         Call-ID:b2af8b83e20b0740
>         CSeq:29663 INVITE
>         Content-Length:0
>
> 08:34:57.561084 IP (tos 0x0, ttl  64, id 0, offset 0, flags [DF], 
> proto: UDP (17), length: 723) 10.0.0.29.55620 > 10.0.0.38.sip: SIP, 
> length: 695
>         SIP/2.0 180 Ringing
>         Via:SIP/2.0/UDP 
> 10.0.0.38;branch=z9hG4bK4ed5.ec1e41a.0,SIP/2.0/UDP 
> 192.168.2.250:51620;received=x.x.x.199;branch=z9hG4bK1158b9f850c008ff7.6835b1eb6e3bffee5;rport=31280
> From:"99999999"<sip:99999999 at r18.test-bwks.local:5060>;tag=1cf0b02fab
> To:"9991"<sip:9991 at r18.test-bwks.local:5060>;tag=1693866893-1347366897367
>         Call-ID:b2af8b83e20b0740
>         CSeq:29663 INVITE
>         Supported:
>         Contact:<sip:10.0.0.29:5060>
> Record-Route:<sip:10.0.0.38;r2=on;lr=on;nat=yes>,<sip:x.x.x.102;r2=on;lr=on;nat=yes> 
>
>         P-Asserted-Identity:"Auto 
> Attendant"<sip:9991 at 10.0.0.29;user=phone>
>         Privacy:none
> Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
>         Content-Length:0
>
> 08:34:57.562404 IP (tos 0x10, ttl  64, id 58137, offset 0, flags 
> [none], proto: UDP (17), length: 672) x.x.x.102.sip > x.x.x.199.31280: 
> SIP, length: 644
>         SIP/2.0 180 Ringing
>         Via:SIP/2.0/UDP 
> 192.168.2.250:51620;received=x.x.x.199;branch=z9hG4bK1158b9f850c008ff7.6835b1eb6e3bffee5;rport=31280
> From:"99999999"<sip:99999999 at r18.test-bwks.local:5060>;tag=1cf0b02fab
> To:"9991"<sip:9991 at r18.test-bwks.local:5060>;tag=1693866893-1347366897367
>         Call-ID:b2af8b83e20b0740
>         CSeq:29663 INVITE
>         Supported:
>         Contact:<sip:10.0.0.29:5060>
> Record-Route:<sip:10.0.0.38;r2=on;lr=on;nat=yes>,<sip:x.x.x.102;r2=on;lr=on;nat=yes> 
>
>         P-Asserted-Identity:"Auto 
> Attendant"<sip:9991 at 10.0.0.29;user=phone>
>         Privacy:none
> Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
>         Content-Length:0
>
> 08:34:57.609771 IP (tos 0x0, ttl  64, id 0, offset 0, flags [DF], 
> proto: UDP (17), length: 953) 10.0.0.29.55620 > 10.0.0.38.sip: SIP, 
> length: 925
>         SIP/2.0 200 OK
>         Via:SIP/2.0/UDP 
> 10.0.0.38;branch=z9hG4bK4ed5.ec1e41a.0,SIP/2.0/UDP 
> 192.168.2.250:51620;received=x.x.x.199;branch=z9hG4bK1158b9f850c008ff7.6835b1eb6e3bffee5;rport=31280
> From:"99999999"<sip:99999999 at r18.test-bwks.local:5060>;tag=1cf0b02fab
> To:"9991"<sip:9991 at r18.test-bwks.local:5060>;tag=1693866893-1347366897367
>         Call-ID:b2af8b83e20b0740
>         CSeq:29663 INVITE
>         Supported:
>         Contact:<sip:10.0.0.29:5060>
> Record-Route:<sip:10.0.0.38;r2=on;lr=on;nat=yes>,<sip:x.x.x.102;r2=on;lr=on;nat=yes> 
>
>         P-Asserted-Identity:"Auto 
> Attendant"<sip:9991 at 10.0.0.29;user=phone>
>         Privacy:none
> Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
> Accept:application/media_control+xml,application/sdp,multipart/mixed
>         Content-Type:application/sdp
>         Content-Length:133
>
>         v=0
>         o=BroadWorks 196 1 IN IP4 10.0.0.24
>         s=-
>         c=IN IP4 10.0.0.24
>         t=0 0
>         m=audio 10498 RTP/AVP 0
>         a=rtpmap:0 PCMU/8000
>         a=ptime:20
>
> 08:34:57.612047 IP (tos 0x10, ttl  64, id 58138, offset 0, flags 
> [none], proto: UDP (17), length: 930) x.x.x.102.sip > x.x.x.199.31280: 
> SIP, length: 902
>         SIP/2.0 200 OK
>         Via:SIP/2.0/UDP 
> 192.168.2.250:51620;received=x.x.x.199;branch=z9hG4bK1158b9f850c008ff7.6835b1eb6e3bffee5;rport=31280
> From:"99999999"<sip:99999999 at r18.test-bwks.local:5060>;tag=1cf0b02fab
> To:"9991"<sip:9991 at r18.test-bwks.local:5060>;tag=1693866893-1347366897367
>         Call-ID:b2af8b83e20b0740
>         CSeq:29663 INVITE
>         Supported:
>         Contact:<sip:10.0.0.29:5060>
> Record-Route:<sip:10.0.0.38;r2=on;lr=on;nat=yes>,<sip:x.x.x.102;r2=on;lr=on;nat=yes> 
>
>         P-Asserted-Identity:"Auto 
> Attendant"<sip:9991 at 10.0.0.29;user=phone>
>         Privacy:none
> Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
> Accept:application/media_control+xml,application/sdp,multipart/mixed
>         Content-Type:application/sdp
>         Content-Length:161
>
>         v=0
>         o=BroadWorks 196 1 IN IP4 x.x.x.102
>         s=-
>         c=IN IP4 x.x.x.102
>         t=0 0
>         m=audio 29774 RTP/AVP 0
>         a=rtpmap:0 PCMU/8000
>         a=ptime:20
>         a=sdpmangled:yes
>
> 08:34:57.844645 IP (tos 0x0, ttl  49, id 3239, offset 0, flags [none], 
> proto: UDP (17), length: 506) x.x.x.199.31280 > x.x.x.102.sip: SIP, 
> length: 478
>         ACK sip:10.0.0.29:5060 SIP/2.0
>         Via: SIP/2.0/UDP 
> 192.168.2.250:51620;branch=z9hG4bK15c3c5e900b3f6d36.26b037b92d0bba7c9
>         Route: <sip:x.x.x.102;r2=on;lr=on;nat=yes>, 
> <sip:10.0.0.38;r2=on;lr=on;nat=yes>
>         Max-Forwards: 70
>         From: "99999999" 
> <sip:99999999 at r18.test-bwks.local:5060>;tag=1cf0b02fab
>         To: "9991" 
> <sip:9991 at r18.test-bwks.local:5060>;tag=1693866893-1347366897367
>         Call-ID: b2af8b83e20b0740
>         CSeq: 29663 ACK
>         User-Agent: Aastra 57i/2.6.0.1007
>         Content-Length: 0
>
> 08:34:57.845182 IP (tos 0x10, ttl  64, id 47937, offset 0, flags 
> [none], proto: UDP (17), length: 509) 10.0.0.38.sip > 10.0.0.29.sip: 
> SIP, length: 481
>         ACK sip:10.0.0.29:5060 SIP/2.0
>         Via: SIP/2.0/UDP 10.0.0.38;branch=z9hG4bKcydzigwkX
>         Via: SIP/2.0/UDP 
> 192.168.2.250:51620;rport=31280;received=x.x.x.199;branch=z9hG4bK15c3c5e900b3f6d36.26b037b92d0bba7c9
>         Max-Forwards: 69
>         From: "99999999" 
> <sip:99999999 at r18.test-bwks.local:5060>;tag=1cf0b02fab
>         To: "9991" 
> <sip:9991 at r18.test-bwks.local:5060>;tag=1693866893-1347366897367
>         Call-ID: b2af8b83e20b0740
>         CSeq: 29663 ACK
>         User-Agent: Aastra 57i/2.6.0.1007
>         Content-Length: 0
>
> ------------------------------------------------------------------------------- 
>
>
> The call session is up at this point.
>
> CPE Router, Source NAT IP x.x.x.199, Internal Network: 192.168.2.0/24
>     192.168.2.250  is an Aastra 6757i IP Phone, behind the
>
> x.x.x.102 is the external interface on our SIP proxy allowing traffic on:
>     Inbound: port 5060/udp
>     Inbound: range 16384 - 32768/udp
>
>     10.0.0.38      is the internal interface facing our BroadWorks lab 
> cluster
>     10.0.0.29       is the internal interface of our BroadWorks 
> Application Server
>     10.0.0.24       is the internal interface of our BroadWorks Media 
> Server
>
> RTP Proxy has been spawned with the following command line arguments:
>
>     /usr/bin/rtpproxy -p /var/run/rtpproxy.pid -u rtpproxy \
>         -u root \
>         -s udp:localhost:7722 \
>         -l x.x.x.102/10.0.0.38 \
>         -d DBUG LOG_LOCAL5 \
>         -m 16384 -M 32768
>
> We've changed the /etc/kamailio/kamailio.cfg and added the #!define's 
> at the top of the
> configuration file:
>
> #!define WITH_NAT
> #!define WITH_DEBUG
>
> In the route[NATMANAGE] { } block we've added a few additional lines 
> to handle the setup
> of the rtpproxy_manage() function which handles the rtpproxy_offer(), 
> rtpproxy_answer()
> and unforce_rtp_proxy() convientenly in one function.
>
> The logic we've got in place currently is:
>
>     if ((src_ip =~ "10\.0\.0\.*") && (dst_ip =~ "10\.0\.0\.*")) {
>             xlog("L_INFO", "Media session for ($ci) with source 
> address ($si): Internal > External: SDP contact/origin will be 
> x.x.x.102");
>             rtpproxy_manage("FAIEOC", "x.x.x.102");
>
>     } else if (!(src_ip =~ "10\.0\.0\.*") && (dst_ip == x.x.x.102)) {
>             xlog("L_INFO", "Media session for ($ci) with source 
> address ($si): External > Internal: SDP contact/origin will be 
> 10.0.0.38");
>             rtpproxy_manage("FAEIOC", "10.0.0.38");
>
>     } else {
>             xlog("L_INFO", "!!! Uncaught condition for media session 
> for ($ci) with source address ($si) !!!");
>     }
>
> This rtpproxy log is not from the call above, but the same source #, 
> destination # were used:
>
> Sep 11 08:50:27 sip-proxy1 rtpproxy[3101]: INFO:handle_command: new 
> session 84bf2a7cb9d4ab24, tag dcfa6a860e;1 requested, type strong
> Sep 11 08:50:27 sip-proxy1 rtpproxy[3101]: INFO:handle_command: new 
> session on a port 28282 created, tag dcfa6a860e;1
> Sep 11 08:50:27 sip-proxy1 rtpproxy[3101]: INFO:handle_command: 
> pre-filling caller's address with 192.168.2.250:51720
> Sep 11 08:50:27 sip-proxy1 rtpproxy[3101]: INFO:handle_command: lookup 
> on ports 28282/22500, session timer restarted
> Sep 11 08:50:27 sip-proxy1 rtpproxy[3101]: INFO:handle_command: 
> pre-filling callee's address with 10.0.0.24:10502
>
> ... Call is currently active and then hung-up ...
>
> Sep 11 08:51:28 sip-proxy1 rtpproxy[3101]: INFO:process_rtp: session 
> timeout
> Sep 11 08:51:28 sip-proxy1 rtpproxy[3101]: INFO:remove_session: RTP 
> stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped
> Sep 11 08:51:28 sip-proxy1 rtpproxy[3101]: INFO:remove_session: RTCP 
> stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped
> Sep 11 08:51:28 sip-proxy1 rtpproxy[3101]: INFO:remove_session: 
> session on ports 28282/22500 is cleaned up
>
> Example of the RTP:
>
> 08:53:13.555834 IP (tos 0x0, ttl  48, id 510, offset 0, flags [none], 
> proto: UDP (17), length: 200) x.x.x.199.32052 > x.x.x.102.16552: UDP, 
> length 172
>
>     x.x.x.199:32052 > x.x.x.102:16552
>         Why is this packet not being forwarded to the media proxy
>         (rtpproxy) to 10.0.0.24 (specified in the 200 OK from the 
> BroadWorks Application Server)
>
> 08:53:13.562206 IP (tos 0x10, ttl  64, id 0, offset 0, flags [DF], 
> proto: UDP (17), length: 200) 10.0.0.24.10504 > 10.0.0.38.17968: UDP, 
> length 172
>
>     10.0.0.24:10504 > 10.0.0.38:17968
>         Why is this packet not being forwarded from rtpproxy to the 
> UAC at x.x.x.199?
>
>
> ------------------------------------------------------------------------------- 
>
> System details:
> ------------------------------------------------------------------------------- 
>
> OS: CentOS 5.8 x86-64, Hypervisor is qemu-kvm
> Memory: 2GB
> Disk: 40G
> VCPU: 1
>
> /etc/sysctl.conf:
>
> # Controls IP packet forwarding
> net.ipv4.ip_forward = 1
>
> ------------------------------------------------------------------------------- 
>
> Current software versions:
> ------------------------------------------------------------------------------- 
>
> [root at sip-proxy1 kamailio]# rpm -qa | grep -E 'kamailio|rtpproxy' | 
> xargs rpm -qi
> Name        : rtpproxy                     Relocations: (not relocatable)
> Version     : 1.2.1                             Vendor: Fedora Project
> Release     : 2.el5                         Build Date: Sun 19 Sep 
> 2010 08:12:23 AM EDT
> Install Date: Mon 10 Sep 2012 09:23:28 PM EDT      Build Host: 
> x86-10.phx2.fedoraproject.org
> Group       : Applications/Internet         Source RPM: 
> rtpproxy-1.2.1-2.el5.src.rpm
> Size        : 105682                           License: BSD
> Signature   : DSA/SHA1, Mon 20 Sep 2010 12:40:46 PM EDT, Key ID 
> 119cc036217521f6
> Packager    : Fedora Project
> URL         : http://www.rtpproxy.org
> Summary     : A symmetric RTP proxy
> Description :
> This is symmetric RTP proxy designed to be used in conjunction with
> the SIP Express Router (SER) or any other SIP proxy capable of
> rewriting SDP bodies in SIP messages that it processes.
>
> Name        : kamailio                     Relocations: (not relocatable)
> Version     : 3.2.4                             Vendor: kamailio.org
> Release     : 1.1                           Build Date: Fri 03 Aug 
> 2012 05:14:14 AM EDT
> Install Date: Tue 11 Sep 2012 08:22:32 AM EDT      Build Host: build08
> Group       : Productivity/Telephony/SIP/Servers   Source RPM: 
> kamailio-3.2.4-1.1.src.rpm
> Size        : 10200784                         License: GPL
> Signature   : DSA/SHA1, Fri 03 Aug 2012 05:14:38 AM EDT, Key ID 
> c9a75909941fdbdd
> Packager    : Daniel-Constantin Mierla <miconda at gmail.com>
> URL         : http://kamailio.org/
> Summary     : Kamailio, very fast, reliable and flexible SIP Server
> Description :
> Kamailio is a very fast, reliable and flexible SIP (RFC3261)
> proxy server. Written entirely in C, kamailio can handle thousands calls
> per second even on low-budget hardware. A C Shell like scripting language
> provides full control over the server's behaviour. It's modular
> architecture allows only required functionality to be loaded.
> Among available features: IPv4, IPv6, digest authentication, accounting,
> CPL scripts, instant messaging, MySQL and UNIXODBC support, SIMPLE 
> presence
> agent, radius authentication, record routing, SMS gateway, ENUM, UDP, 
> TCP,
> TLS and SCTP, transaction and dialog module, OSP module, statistics 
> support,
> registrar and user location, SNMP, SIMPLE Presence and Perl programming
> interface.
>
> Name        : kamailio-cpl                 Relocations: (not relocatable)
> Version     : 3.2.4                             Vendor: kamailio.org
> Release     : 1.1                           Build Date: Fri 03 Aug 
> 2012 05:14:14 AM EDT
> Install Date: Tue 11 Sep 2012 08:22:33 AM EDT      Build Host: build08
> Group       : Productivity/Telephony/SIP/Servers   Source RPM: 
> kamailio-3.2.4-1.1.src.rpm
> Size        : 296947                           License: GPL
> Signature   : DSA/SHA1, Fri 03 Aug 2012 05:14:38 AM EDT, Key ID 
> c9a75909941fdbdd
> Packager    : Daniel-Constantin Mierla <miconda at gmail.com>
> URL         : http://kamailio.org/
> Summary     : CPL module (CPL interpreter engine) for Kamailio
> Description :
> The kamailio-cpl package provides a CPL interpreter engine for Kamailio
>
>
> _______________________________________________
> 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

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat
Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu




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