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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:54326@192.168.3.72:5060;transport=udp SIP/2.0 Call-ID: 8066502182f0e51ea4f56cb1b8800 CSeq: 1 INVITE From: "Telecom Network Dep." <sip:30003@elgouna.com: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:54326@elgouna.com> 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:30003@192.168.1.33:5061;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:internal@elgouna.com>;avaya-cm-alert-type=internal Min-SE: 1200 Session-Expires: 1200;refresher=uac P-Asserted-Identity: "Telecom Network Dep." <sip:30003@elgouna.com:5061> History-Info: <sip:54326@elgouna.com>;index=1,"54326" <sip:54326@elgouna.com>;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 |
#2
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Hello Guys ,
Any support here ?? afahmy |
#3
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Make sure to have the same CODEC between UAs as well as CM and SMS
__________________
Wellington Paez Senior Convergence Specialist @ Carousel Industries http://wellingtonpaez.com |
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