I have a combined Home/edge server. Avaya G450 S8300D SES. It seems that the ACM and the SES are not communicating completely. I can NOT dial sip to sip nor H.323 to Sip. I can call out sip to H.323.
The problem has come up and I do not know which test to run. I have tried list trace tac - List trace station - and I have some logs on the sip server. If I do a status station in ASA it shows Off PBX Service State: In-service/Idle and In-service/ active when ringing.
I am able to call a sip extension and it will ring the sip extension I will go off hook on the sip extn and it will stop ringing called sip extn and I hear a stuttered dial tone - but the calling party continues to hear ringing. When I go off hook and hear stutter dial tone-- and I dial any digit on the dialpad it will connect me to the caller and work. If I simply go off hook and wait the call continues on its coverage path and the next station rings.
When I look at the Sip trace log I can see that two Bye messages are exchanged and the call rings the next phone in cover path ( when I just go off hook). If I dial a digit than it shows up as a second ack message and all other notify messages.
It seemed to happen all of a sudden. They also have a VPN between sites cisco 5505 to Cisco 5520. I would appreciate any help in call flow of a sip phone to sip phone call. Also any test to determine the culprit is it the ACM or is it the Sip enablement Server.
I was able to make a sip call to external IPO and the call connected. SIP TRACE LOG SHOWS 2 DIFFERENCES ON THE (Recv Request) Invite the non working has a line below History info named Accept-Contact: *;+avaya-CM-line=1 ( What is this line all about)
Rcvd Request Ack the to: "xxxx"shows the ip address of the sip phone after the tag=**************_T10.x.x.x ( Assume trusted site cause of the T )
Record Route: <sip:x.x.x.x:5061; Transport=tls; lr> The call that worked does not show the Transport=tls it just shows lr> at the end.
The problem has come up and I do not know which test to run. I have tried list trace tac - List trace station - and I have some logs on the sip server. If I do a status station in ASA it shows Off PBX Service State: In-service/Idle and In-service/ active when ringing.
I am able to call a sip extension and it will ring the sip extension I will go off hook on the sip extn and it will stop ringing called sip extn and I hear a stuttered dial tone - but the calling party continues to hear ringing. When I go off hook and hear stutter dial tone-- and I dial any digit on the dialpad it will connect me to the caller and work. If I simply go off hook and wait the call continues on its coverage path and the next station rings.
When I look at the Sip trace log I can see that two Bye messages are exchanged and the call rings the next phone in cover path ( when I just go off hook). If I dial a digit than it shows up as a second ack message and all other notify messages.
It seemed to happen all of a sudden. They also have a VPN between sites cisco 5505 to Cisco 5520. I would appreciate any help in call flow of a sip phone to sip phone call. Also any test to determine the culprit is it the ACM or is it the Sip enablement Server.
I was able to make a sip call to external IPO and the call connected. SIP TRACE LOG SHOWS 2 DIFFERENCES ON THE (Recv Request) Invite the non working has a line below History info named Accept-Contact: *;+avaya-CM-line=1 ( What is this line all about)
Rcvd Request Ack the to: "xxxx"shows the ip address of the sip phone after the tag=**************_T10.x.x.x ( Assume trusted site cause of the T )
Record Route: <sip:x.x.x.x:5061; Transport=tls; lr> The call that worked does not show the Transport=tls it just shows lr> at the end.
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