[SR-Users] Stress Testing

Jack Stevens Jack.Stevens at netcall.com
Tue Aug 16 16:47:47 CEST 2016 

Hi,

Yeah we get calls time out please see the below

SIP messages for Call-ID bdc7441b322047868e294188984bbd09

15:46:49.599 [+0.00ms] [TX] INVITE to 192.168.3.204:5060
INVITE sip:12345 at 192.168.3.204:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.79:5070;rport;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b
Max-Forwards: 70
From: sip:55123 at 192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d
To: sip:12345 at 192.168.3.204
Contact: <sip:55123 at 192.168.1.79:5070>
Call-ID: bdc7441b322047868e294188984bbd09
CSeq: 5624 INVITE
Allow: INFO, PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE
Supported: 100rel, timer
User-Agent: StarTrinity.SIP 2016-07-13 16.10 UTC
Session-Expires: 3600;refresher=uac
Content-Type: application/sdp
Content-Length:   329

v=0
o=- 3680351217 3680351217 IN IP4 192.168.1.79
s=o14160.proxy.stream0
c=IN IP4 192.168.1.79
t=0 0
m=audio 16114 RTP/AVP 8 0 4 18 101
a=rtcp:16115 IN IP4 192.168.1.79
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

============================================================end of message=================
15:46:49.604 [+5.11ms] [RX] trying -- your call is important to us from 192.168.3.204:5060
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 192.168.1.79:5070;rport=5070;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b;received=192.168.1.79
From: sip:55123 at 192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d
To: sip:12345 at 192.168.3.204
Call-ID: bdc7441b322047868e294188984bbd09
CSeq: 5624 INVITE
Content-Length: 0


============================================================end of message=================
15:46:51.554 [+1,955.27ms] [RX] Request Timeout from 192.168.3.204:5060
SIP/2.0 408 Request Timeout
Via: SIP/2.0/UDP 192.168.1.79:5070;rport=5070;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b;received=192.168.1.79
From: sip:55123 at 192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d
To: sip:12345 at 192.168.3.204;tag=604fc74b9f0d3ee4c15ae560ab8f892f-8007
Call-ID: bdc7441b322047868e294188984bbd09
CSeq: 5624 INVITE
Content-Length: 0


============================================================end of message=================
15:46:51.554 [+1,955.29ms] [TX] ACK to 192.168.3.204:5060
ACK sip:12345 at 192.168.3.204:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.79:5070;rport;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b
Max-Forwards: 70
From: sip:55123 at 192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d
To: sip:12345 at 192.168.3.204;tag=604fc74b9f0d3ee4c15ae560ab8f892f-8007
Call-ID: bdc7441b322047868e294188984bbd09
CSeq: 5624 ACK
Content-Length:  0


============================================================end of message=================
15:46:59.312 [+9,713.29ms] [RX] Trying from 192.168.3.204:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.79:5070;received=192.168.1.79;rport=5070;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b
Record-Route: <sip:10.200.0.56;line=sr-mUuGcDnGhDNG4FYGhD8UcPNa1TKecUya1TKe>
Record-Route: <sip:192.168.3.204;r2=on;lr=on>
From: sip:55123 at 192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d
To: sip:12345 at 192.168.3.204
Call-ID: bdc7441b322047868e294188984bbd09
CSeq: 5624 INVITE
Server: FPBX-2.11.0(11.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:10.200.0.56;line=sr-mUuGcDna4b8g8qnGhDNG4FYGhDLPcDLGvDB*>
Content-Length: 0


============================================================end of message=================
15:46:59.312 [+9,713.43ms] [RX] OK from 192.168.3.204:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.79:5070;received=192.168.1.79;rport=5070;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b
Record-Route: <sip:10.200.0.56;line=sr-mUuGcDnGhDNG4FYGhD8UcPNa1TKecUya1TKe>
Record-Route: <sip:192.168.3.204;r2=on;lr=on>
From: sip:55123 at 192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d
To: sip:12345 at 192.168.3.204;tag=as5772e8de
Call-ID: bdc7441b322047868e294188984bbd09
CSeq: 5624 INVITE
Server: FPBX-2.11.0(11.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:10.200.0.56;line=sr-mUuGcDna4b8g8qnGhDNG4FYGhDLPcDLGvDB*>
Content-Type: application/sdp
Require: timer
Content-Length: 258

v=0
o=root 1651160446 1651160446 IN IP4 10.200.0.57
s=Asterisk PBX 11.20.0
c=IN IP4 10.200.0.57
t=0 0
m=audio 10720 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

============================================================end of message=================
15:46:59.312 [+9,713.44ms] [TX] ACK to 192.168.3.204:5060
ACK sip:12345 at 192.168.3.204:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.79:5070;rport;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b
Max-Forwards: 70
From: sip:55123 at 192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d
To: sip:12345 at 192.168.3.204;tag=604fc74b9f0d3ee4c15ae560ab8f892f-8007
Call-ID: bdc7441b322047868e294188984bbd09
CSeq: 5624 ACK
Content-Length:  0


============================================================end of message=================
15:46:59.313 [+9,714.41ms] [RX] Trying from 192.168.3.204:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.79:5070;received=192.168.1.79;rport=5070;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b
Record-Route: <sip:10.200.0.56;line=sr-mUuGcDnGhDNG4FYGhD8UcPNa1TKecUya1TKe>
Record-Route: <sip:192.168.3.204;r2=on;lr=on>
From: sip:55123 at 192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d
To: sip:12345 at 192.168.3.204
Call-ID: bdc7441b322047868e294188984bbd09
CSeq: 5624 INVITE
Server: FPBX-2.11.0(11.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:10.200.0.56;line=sr-mUuGcDna4b8g8qnGhDNG4FYGhDLPcDLGvDB*>
Content-Length: 0


============================================================end of message=================
15:46:59.314 [+9,714.49ms] [RX] OK from 192.168.3.204:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.79:5070;received=192.168.1.79;rport=5070;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b
Record-Route: <sip:10.200.0.56;line=sr-mUuGcDnGhDNG4FYGhD8UcPNa1TKecUya1TKe>
Record-Route: <sip:192.168.3.204;r2=on;lr=on>
From: sip:55123 at 192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d
To: sip:12345 at 192.168.3.204;tag=as5772e8de
Call-ID: bdc7441b322047868e294188984bbd09
CSeq: 5624 INVITE
Server: FPBX-2.11.0(11.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:10.200.0.56;line=sr-mUuGcDna4b8g8qnGhDNG4FYGhDLPcDLGvDB*>
Content-Type: application/sdp
Require: timer
Content-Length: 258

v=0
o=root 1651160446 1651160447 IN IP4 10.200.0.57
s=Asterisk PBX 11.20.0
c=IN IP4 10.200.0.57
t=0 0
m=audio 10720 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

============================================================end of message=================
15:46:59.314 [+9,714.50ms] [TX] ACK to 192.168.3.204:5060
ACK sip:12345 at 192.168.3.204:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.79:5070;rport;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b
Max-Forwards: 70
From: sip:55123 at 192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d
To: sip:12345 at 192.168.3.204;tag=604fc74b9f0d3ee4c15ae560ab8f892f-8007
Call-ID: bdc7441b322047868e294188984bbd09
CSeq: 5624 ACK
Content-Length:  0


============================================================end of message=================
15:46:59.450 [+9,850.82ms] [RX] Trying from 192.168.3.204:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.79:5070;received=192.168.1.79;rport=5070;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b
Record-Route: <sip:10.200.0.56;line=sr-mUuGcDnGhDNG4FYGhD8UcPNa1TKecUya1TKe>
Record-Route: <sip:192.168.3.204;r2=on;lr=on>
From: sip:55123 at 192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d
To: sip:12345 at 192.168.3.204
Call-ID: bdc7441b322047868e294188984bbd09
CSeq: 5624 INVITE
Server: FPBX-2.11.0(11.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:10.200.0.56;line=sr-mUuGcDna4b8g8qnGhDNG4FYGhDLPcDLGvDB*>
Content-Length: 0


============================================================end of message=================
15:46:59.450 [+9,850.84ms] [RX] OK from 192.168.3.204:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.79:5070;received=192.168.1.79;rport=5070;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b
Record-Route: <sip:10.200.0.56;line=sr-mUuGcDnGhDNG4FYGhD8UcPNa1TKecUya1TKe>
Record-Route: <sip:192.168.3.204;r2=on;lr=on>
From: sip:55123 at 192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d
To: sip:12345 at 192.168.3.204;tag=as5772e8de
Call-ID: bdc7441b322047868e294188984bbd09
CSeq: 5624 INVITE
Server: FPBX-2.11.0(11.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:10.200.0.56;line=sr-mUuGcDna4b8g8qnGhDNG4FYGhDLPcDLGvDB*>
Content-Type: application/sdp
Require: timer
Content-Length: 258

v=0
o=root 1651160446 1651160448 IN IP4 10.200.0.57
s=Asterisk PBX 11.20.0
c=IN IP4 10.200.0.57
t=0 0
m=audio 10720 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

============================================================end of message=================
15:46:59.450 [+9,850.86ms] [TX] ACK to 192.168.3.204:5060
ACK sip:12345 at 192.168.3.204:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.79:5070;rport;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b
Max-Forwards: 70
From: sip:55123 at 192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d
To: sip:12345 at 192.168.3.204;tag=604fc74b9f0d3ee4c15ae560ab8f892f-8007
Call-ID: bdc7441b322047868e294188984bbd09
CSeq: 5624 ACK
Content-Length:  0


============================================================end of message=================
15:46:59.853 [+10,253.75ms] [RX] OK from 192.168.3.204:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.79:5070;received=192.168.1.79;rport=5070;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b
Record-Route: <sip:10.200.0.56;line=sr-mUuGcDnGhDNG4FYGhD8UcPNa1TKecUya1TKe>
Record-Route: <sip:192.168.3.204;r2=on;lr=on>
From: sip:55123 at 192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d
To: sip:12345 at 192.168.3.204;tag=as5772e8de
Call-ID: bdc7441b322047868e294188984bbd09
CSeq: 5624 INVITE
Server: FPBX-2.11.0(11.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:10.200.0.56;line=sr-mUuGcDna4b8g8qnGhDNG4FYGhDLPcDLGvDB*>
Content-Type: application/sdp
Require: timer
Content-Length: 258

v=0
o=root 1651160446 1651160446 IN IP4 10.200.0.57
s=Asterisk PBX 11.20.0
c=IN IP4 10.200.0.57
t=0 0
m=audio 10720 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

============================================================end of message=================
15:46:59.853 [+10,253.77ms] [TX] ACK to 192.168.3.204:5060
ACK sip:12345 at 192.168.3.204:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.79:5070;rport;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b
Max-Forwards: 70
From: sip:55123 at 192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d
To: sip:12345 at 192.168.3.204;tag=604fc74b9f0d3ee4c15ae560ab8f892f-8007
Call-ID: bdc7441b322047868e294188984bbd09
CSeq: 5624 ACK
Content-Length:  0


============================================================end of message=================
15:47:00.800 [+11,200.72ms] [RX] OK from 192.168.3.204:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.79:5070;received=192.168.1.79;rport=5070;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b
Record-Route: <sip:10.200.0.56;line=sr-mUuGcDnGhDNG4FYGhD8UcPNa1TKecUya1TKe>
Record-Route: <sip:192.168.3.204;r2=on;lr=on>
From: sip:55123 at 192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d
To: sip:12345 at 192.168.3.204;tag=as5772e8de
Call-ID: bdc7441b322047868e294188984bbd09
CSeq: 5624 INVITE
Server: FPBX-2.11.0(11.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:10.200.0.56;line=sr-mUuGcDna4b8g8qnGhDNG4FYGhDLPcDLGvDB*>
Content-Type: application/sdp
Require: timer
Content-Length: 258

v=0
o=root 1651160446 1651160446 IN IP4 10.200.0.57
s=Asterisk PBX 11.20.0
c=IN IP4 10.200.0.57
t=0 0
m=audio 10720 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

============================================================end of message=================
15:47:00.800 [+11,200.73ms] [TX] ACK to 192.168.3.204:5060
ACK sip:12345 at 192.168.3.204:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.79:5070;rport;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b
Max-Forwards: 70
From: sip:55123 at 192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d
To: sip:12345 at 192.168.3.204;tag=604fc74b9f0d3ee4c15ae560ab8f892f-8007
Call-ID: bdc7441b322047868e294188984bbd09
CSeq: 5624 ACK
Content-Length:  0


============================================================end of message=================
15:47:02.811 [+13,211.65ms] [RX] OK from 192.168.3.204:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.79:5070;received=192.168.1.79;rport=5070;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b
Record-Route: <sip:10.200.0.56;line=sr-mUuGcDnGhDNG4FYGhD8UcPNa1TKecUya1TKe>
Record-Route: <sip:192.168.3.204;r2=on;lr=on>
From: sip:55123 at 192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d
To: sip:12345 at 192.168.3.204;tag=as5772e8de
Call-ID: bdc7441b322047868e294188984bbd09
CSeq: 5624 INVITE
Server: FPBX-2.11.0(11.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:10.200.0.56;line=sr-mUuGcDna4b8g8qnGhDNG4FYGhDLPcDLGvDB*>
Content-Type: application/sdp
Require: timer
Content-Length: 258

v=0
o=root 1651160446 1651160446 IN IP4 10.200.0.57
s=Asterisk PBX 11.20.0
c=IN IP4 10.200.0.57
t=0 0
m=audio 10720 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

============================================================end of message=================
15:47:02.811 [+13,211.67ms] [TX] ACK to 192.168.3.204:5060
ACK sip:12345 at 192.168.3.204:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.79:5070;rport;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b
Max-Forwards: 70
From: sip:55123 at 192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d
To: sip:12345 at 192.168.3.204;tag=604fc74b9f0d3ee4c15ae560ab8f892f-8007
Call-ID: bdc7441b322047868e294188984bbd09
CSeq: 5624 ACK
Content-Length:  0


============================================================end of message=================
===============================saved by StarTrinity SIP Tester at 16/08/2016 15:47:05======

Jack Stevens



Cloud Systems and Network Administrator



Netcall

t   0330 333 6100

f   0330 333 0102

e  jack.stevens at netcall.com<mailto:jack.stevens at netcall.com>

w www.netcall.com<http://www.netcall.com>

b  www.netcall.com/blog<http://www.netcall.com/blog>

n  www.netcall.com/subscribe<http://www.netcall.com/subscribe>





From: sr-users [mailto:sr-users-bounces at lists.sip-router.org] On Behalf Of Olle E. Johansson
Sent: 16 August 2016 15:45
To: Kamailio (SER) - Users Mailing List <sr-users at lists.sip-router.org>
Subject: Re: [SR-Users] Stress Testing


On 16 Aug 2016, at 16:26, Jack Stevens <Jack.Stevens at netcall.com<mailto:Jack.Stevens at netcall.com>> wrote:

Hi Guys,

I have been stress testing my Kamailio box but I am unable to get it upto 2000 concurrent calls it starts to fall over at 1300 have you got any ideas on how I can increase the performance of kamilio btw I am also using rtpengine
Can you describe “fall over”

As Kamailio doesn’t handle media it’s likely an issue with rtpengine and the developers there needs to respond,
but some more facts would be good. :-)

/O


Kind Regards






CONFIDENTIAL EMAIL FROM NETCALL TELECOM LIMITED

This email, and any attachments, is intended only for the above addressee. It may contain private and/or confidential information. If you have received this email in error you are on notice of its status, please immediately notify the sender by return email then delete this message and any attachments. If you are not the addressee, except to notify the sender, you must not use, disclose, copy or distribute this email and/or its attachments. Netcall Telecom accepts no responsibility for any changes made to this message after it has been sent by the original author. Opinions or views expressed in this email may be those of the individual sender and not Netcall Telecom. Nothing in this email shall bind Netcall Telecom in any contract or obligation

Netcall Telecom Ltd Registered in England 2831215. Registered Office : 3rd Floor, Hamilton House, 111 Marlowes, Hemel Hempstead, Herts, HP1 1BB
_______________________________________________
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


-- 





CONFIDENTIAL EMAIL FROM NETCALL TELECOM LIMITED

This email, and any attachments, is intended only for the above addressee. 
It may contain private and/or confidential information. If you have 
received this email in error you are on notice of its status, please 
immediately notify the sender by return email then delete this message and 
any attachments. If you are not the addressee, except to notify the sender, 
you must not use, disclose, copy or distribute this email and/or its 
attachments. Netcall Telecom accepts no responsibility for any changes made 
to this message after it has been sent by the original author. Opinions or 
views expressed in this email may be those of the individual sender and not 
Netcall Telecom. Nothing in this email shall bind Netcall Telecom in any 
contract or obligation

Netcall Telecom Ltd Registered in England 2831215. Registered Office : 3rd 
Floor, Hamilton House, 111 Marlowes, Hemel Hempstead, Herts, HP1 1BB
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20160816/2d3510b6/attachment.html>

More information about the sr-users mailing list