Hello All ,
my customer have SIP devices that register to standalone SES server version 5.2 integrated with CM4.0 , these SIP devices output are analog phones .
Issue is that phone calls from analog phones ( connected to SIP devices ) to any other phone ( EX: IP phone ) is not working properly ! sometimes i have no dial tone , sometimes i can dial but get fast busy tone , sometimes no ring at all .
Administration Done on CM side
----------------------
- Created SIP trunk / SIP signaling group with CLAN as near end SES server as far end , using TLS with port 5061.
- Created administrator account profile 18.
- Added the CLAN in ip-services and made it allow SAT access , port 5023
Administration Done on SES side
----------------------
- Administered the CM interface with CLAN .
- used the created account on CM and choose telnet over 5023.
when i am doing test link on SES i got the SMS status is DOWN due to time out to reach localhost.
And on the SES Alarms there is the EventID 68 minor error repeated (avCCSPPMResourceError: Authentication Failure )
I tried to access the CM with the created account i used in SES and i was able to login to SAT normally !!
Any idea behind this alarm and SMS DOWN status and how i can bring it up ? i want to fix that to isolate the issue from being in my devices before checking the third party SIP devices.
Also below a trace captured on SES between IP Phone ( 30003) and analog phone attached to SIP device ( 54326 ) and i got fast busy tone !
----------
Mar 13 23:02:46 2016 matching filter label <30003 to 54326>: elgouna.elgouna.com: [Send Request ]
{connection: host=192.168.3.72 port=5060 protocol=UDP}
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Call-ID: 8066502182f0e51ea4f56cb1b8800
CSeq: 1 INVITE
From: "Telecom Network Dep." <sip:[email protected]:5061>;tag=8066502182f0e51e9 4f56cb1b8800
Record-Route: <sip:192.168.1.35:5060;lr>,<sip:192.168.1.33:5061; lr;transport=tls>
To: "54326" <sip:[email protected]>
Via: SIP/2.0/UDP 192.168.1.35:5060;branch=z9hG4bK838383030303636363 31a343b.0,SIP/2.0/TLS 192.168.1.33;psrrposn=2;received=192.168.1.33;bran ch=z9hG4bK8066502182f0e51eb4f56cb1b8800
Content-Length: 271
Content-Type: application/sdp
Contact: "Telecom Network Dep." <sip:[email protected]:5061;transport=tls>
Max-Forwards: 70
User-Agent: Avaya CM/R014x.00.5.742.0
Allow: INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER ,OPTIONS
Accept-Contact: *;+avaya-cm-line=1
Supported: 100rel,timer,replaces,join,histinfo
Alert-Info: <cid:[email protected]>;avaya-cm-alert-type=internal
Min-SE: 1200
Session-Expires: 1200;refresher=uac
P-Asserted-Identity: "Telecom Network Dep." <sip:[email protected]:5061>
History-Info: <sip:[email protected]>;index=1,"54326" <sip:[email protected]>;index=1.1
v=0
o=- 1 1 IN IP4 192.168.1.33
s=-
c=IN IP4 192.168.1.29
t=0 0
m=audio 27196 RTP/AVP 0 18 4 8 127
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:8 PCMA/8000
a=rtpmap:127 telephone-event/8000
-----------
Appreciate any support on that .
afahmy
my customer have SIP devices that register to standalone SES server version 5.2 integrated with CM4.0 , these SIP devices output are analog phones .
Issue is that phone calls from analog phones ( connected to SIP devices ) to any other phone ( EX: IP phone ) is not working properly ! sometimes i have no dial tone , sometimes i can dial but get fast busy tone , sometimes no ring at all .
Administration Done on CM side
----------------------
- Created SIP trunk / SIP signaling group with CLAN as near end SES server as far end , using TLS with port 5061.
- Created administrator account profile 18.
- Added the CLAN in ip-services and made it allow SAT access , port 5023
Administration Done on SES side
----------------------
- Administered the CM interface with CLAN .
- used the created account on CM and choose telnet over 5023.
when i am doing test link on SES i got the SMS status is DOWN due to time out to reach localhost.
And on the SES Alarms there is the EventID 68 minor error repeated (avCCSPPMResourceError: Authentication Failure )
I tried to access the CM with the created account i used in SES and i was able to login to SAT normally !!
Any idea behind this alarm and SMS DOWN status and how i can bring it up ? i want to fix that to isolate the issue from being in my devices before checking the third party SIP devices.
Also below a trace captured on SES between IP Phone ( 30003) and analog phone attached to SIP device ( 54326 ) and i got fast busy tone !
----------
Mar 13 23:02:46 2016 matching filter label <30003 to 54326>: elgouna.elgouna.com: [Send Request ]
{connection: host=192.168.3.72 port=5060 protocol=UDP}
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Call-ID: 8066502182f0e51ea4f56cb1b8800
CSeq: 1 INVITE
From: "Telecom Network Dep." <sip:[email protected]:5061>;tag=8066502182f0e51e9 4f56cb1b8800
Record-Route: <sip:192.168.1.35:5060;lr>,<sip:192.168.1.33:5061; lr;transport=tls>
To: "54326" <sip:[email protected]>
Via: SIP/2.0/UDP 192.168.1.35:5060;branch=z9hG4bK838383030303636363 31a343b.0,SIP/2.0/TLS 192.168.1.33;psrrposn=2;received=192.168.1.33;bran ch=z9hG4bK8066502182f0e51eb4f56cb1b8800
Content-Length: 271
Content-Type: application/sdp
Contact: "Telecom Network Dep." <sip:[email protected]:5061;transport=tls>
Max-Forwards: 70
User-Agent: Avaya CM/R014x.00.5.742.0
Allow: INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER ,OPTIONS
Accept-Contact: *;+avaya-cm-line=1
Supported: 100rel,timer,replaces,join,histinfo
Alert-Info: <cid:[email protected]>;avaya-cm-alert-type=internal
Min-SE: 1200
Session-Expires: 1200;refresher=uac
P-Asserted-Identity: "Telecom Network Dep." <sip:[email protected]:5061>
History-Info: <sip:[email protected]>;index=1,"54326" <sip:[email protected]>;index=1.1
v=0
o=- 1 1 IN IP4 192.168.1.33
s=-
c=IN IP4 192.168.1.29
t=0 0
m=audio 27196 RTP/AVP 0 18 4 8 127
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:8 PCMA/8000
a=rtpmap:127 telephone-event/8000
-----------
Appreciate any support on that .
afahmy
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