AAM 6.3 record greeting from handset instigated from webinterface 1

[montyzummer]

Hi Chaps/Chapesses,

So as per the subject , this is direct sip integration into CM from a co-res AAM , from the user pages i select advanced pref , navigate to greetings , there is already a personal greeting recorded from a handset using the pilot number of the vM etc, but when selecting the record from my extension or another phone (being a different extension number)

I just get a failure , i have run the diag tests and it simply returns , going off hook unkown error near end disconnect , i have allowed outcalling (not sure this is required) , the number of trunks on the pbx and the inegration on the VM match , im unsure if im missing something or there is a deeper issue like a bug , as when looking at all the logs and diagnostics pages i get nothing that leads me in a direction towards config issues , any help much appreciated.

ACSS (UC/SBCE/SM/SME)

Not that they mean a thing anymore , get a brain dump pass the test crash the system.

 

[KenMeans]

i'm sure it is the numbering plan. look at the 2 files i uploaded. The routing on the CM side does not know how to route the call

Ken Means

http://www.tel-is.com

"I find that the harder I work, the more luck I seem to have."
- Thomas Jefferson (1743-1826)

 

[KenMeans]

2nd file

Ken Means

http://www.tel-is.com

"I find that the harder I work, the more luck I seem to have."
- Thomas Jefferson (1743-1826)

 

[montyzummer]

Sorry maybe im am being unclear or have misunderstood , the calls route down to VM from a station coverage path , when i run a list trace on the TAC of the SIP trunk group the call does not leave the voicemail(or at least does not activate the trace)

ACSS (UC/SBCE/SM/SME)

Not that they mean a thing anymore , get a brain dump pass the test crash the system.

 

[KenMeans]

A list trace does not show the SIP data stream unless something has changed recently. I can tell you that every time I have seen this it was due to a public unknown or a private unknown numbering issue. And the CM just drops it.

How many trunks do you have setup for out/transfer?

Ken Means

http://www.tel-is.com

"I find that the harder I work, the more luck I seem to have."
- Thomas Jefferson (1743-1826)

 

[KenMeans]

Did you get it working ?


Ken Means

http://www.tel-is.com

"I find that the harder I work, the more luck I seem to have."
- Thomas Jefferson (1743-1826)

 

[montyzummer]

{"DIRECTION": "IN", "FEATURE": "OUTCALL_NOTIFICATION", "IPADDRESS": "10.141.0.22", "DATE": "03/09/17", "STARTTIME": "13:57:41", "CALLERID": "", "DESTINATIONNUMBER": "201108", "DURATION": "0", "STATUS": "[LINE:48 (IRAPI48), GOT DIAL TONE, THERE WAS AN UNKNOWN ERROR, NEAR END DISCONNECTED, CP=ERROR]" }


ACSS (UC/SBCE/SM/SME)

Not that they mean a thing anymore , get a brain dump pass the test crash the system.

 

[montyzummer]

Hi Ken,

Appreciate your help on this thus far.

Aplogies the post submitted whilst i was editing , essentially i turned on debug (for a short period) and below is the log , when the call fails , as you can see i am in the user options for 201101 mailbox in the webinterface , i am attempting to record the no answer greeting for mailbox 201101 on 201108 IP handset , I double checked my AAR and private tables , all logic is there.

I have got 50 trunks configured on the VM for PBX connections and that number matches on the SIP trunk on CM , i would summise if i had a permissions option in the VM on the COS etc , that the log would notify on that error , again as you see the VM reports dial tone , attempts the call and reports CP error (im guessing that is Call processing) running statt app or sysstat from the CLI of the VM shows all system processes as up , in /var/log there is not anything specific ( however i again would assume this debug log would cover the log info that is most relevant... that is unless you can suggest a more specific log i can interrogate.?)

Or anything else to try , the actuall error that is returned in the web interface when attempting this is , "There was a problem communicating with the application server. Please contact your administrator" , which initially pointed me towards config , but now im not so sure ....ohh i am using the dadmin account to do this , but have tried the individual web link of the mailbox users page.

FlowSummaryManager INFO [] Thu Mar 09 13:51:31
----- Flow Summary S2-000321-----
- CallerID :

- Start Time :Mar 09 2017 13:51:23
- End Time :Mar 09 2017 13:51:31
- VbSessionId :S2-000321
- Channel :47
-
- Time Elapsed Module Name User Input
- 0 ADOMO_SYSTEM_MM No User Input
- 0 HANGUP No User Input
- Call elapsed time (S2-000321) : 8713
----- Flow Summary -----

PhoneGateway DEBUG [] Thu Mar 09 13:52:41 HTTP parameters to PhoneGateway Servlet:
Context Path : /adomo
Servlet Path : /phone/gateway
Request URL : /adomo/phone/gateway
Query String : SessionStatus=Stu.Reeves%40mailrelay.scc.com&XmlOut=true
Request Parameters:
SessionStatus(0): Stu.Reeves@mailrelay.scc.com
XmlOut(0): true

PhoneGateway DEBUG [] Thu Mar 09 13:52:48 HTTP parameters to PhoneGateway Servlet:
Context Path : /adomo
Servlet Path : /phone/gateway
Request URL : /adomo/phone/gateway
Query String : Action=GreetingPlay&EMail=Stu.Reeves%40mailrelay.scc.com&MID=null&Phone=201108&Rings=3&GreetingType=PersonalNoAnswer&EncodedMID=null
Request Parameters:
Phone(0): 201108
EMail(0): Stu.Reeves@mailrelay.scc.com
Action(0): GreetingPlay
Rings(0): 3
EncodedMID(0): null
GreetingType(0): PersonalNoAnswer
MID(0): null

SubscriberDialingAssistant INFO [] Thu Mar 09 13:52:48 Create for site 1
OutcallingManager DEBUG [] Thu Mar 09 13:52:48 Starting new GreetingPlay session for phone: 201108
PhoneSocket INFO [] Thu Mar 09 13:52:48 PhoneSocket<Stu.Reeves@mailrelay.scc.com> sending: call 201108:1

http://localhost:7000/adomo/voice/s...=Stu.Reeves@mailrelay.scc.com&CallerId=201101

15000 - scc 201101 "Stu Reeves" 201101 "Stu Reeves"
PhoneSocket INFO [] Thu Mar 09 13:52:48 PhoneSocket<Stu.Reeves@mailrelay.scc.com> received response[0] = Line:48 (irapi48)
PhoneSocket INFO [] Thu Mar 09 13:52:48 PhoneSocket<Stu.Reeves@mailrelay.scc.com> started on line #48
PhoneSocket INFO [] Thu Mar 09 13:52:48 PhoneSocket<Stu.Reeves@mailrelay.scc.com> received response[1] = Got dial tone
PhoneSocket INFO [] Thu Mar 09 13:52:48 PhoneSocket<Stu.Reeves@mailrelay.scc.com> received response[2] = There was an unknown error
PhoneSocket INFO [] Thu Mar 09 13:52:48 PhoneSocket<Stu.Reeves@mailrelay.scc.com> received response[3] = Near End disconnected
PhoneSocket INFO [] Thu Mar 09 13:52:48 PhoneSocket<Stu.Reeves@mailrelay.scc.com> received response[4] = CP=ERROR
PhoneSocket INFO [] Thu Mar 09 13:52:48 PhoneSocket<Stu.Reeves@mailrelay.scc.com> received response[5] = ok
PhoneSocket INFO [] Thu Mar 09 13:52:48 PhoneSocket<Stu.Reeves@mailrelay.scc.com> ended on line #48
OutcallingManager DEBUG [] Thu Mar 09 13:52:48 Failed to start GreetingPlay session for phone: 201108
AdPhoneLogElement DEBUG [] Thu Mar 09 13:52:48 {"DIRECTION": "IN", "FEATURE": "OUTCALL_NOTIFICATION", "IPADDRESS": "10.141.0.22", "DATE": "03/09/17", "STARTTIME": "13:52:48", "CALLERID": "", "DESTINATIONNUMBER": "201108", "DURATION": "0", "STATUS": "[LINE:48 (IRAPI48), GOT DIAL TONE, THERE WAS AN UNKNOWN ERROR, NEAR END DISCONNECTED, CP=ERROR]"

ACSS (UC/SBCE/SM/SME)

Not that they mean a thing anymore , get a brain dump pass the test crash the system.

 

[montyzummer]

I have continued to play with the private and public tables and also have ensured that the proxy rte is configured in locations , messaging has its own location in CM , the thing that bugs me is the "near end disconnect" i would expect to see "far end disconnect" if CM was binning the call or atleast a little more in the log stating what sort of disconnect (as in client or server) , i would love to see a SIP call flow datagram so i could proove to myself what is dropping the call and why , i am going to try and set up a wireshark on a mirrored port on the VM server , as that will give me a clear picture of what is happening if the call actually does attempt to leave the VM box.

ACSS (UC/SBCE/SM/SME)

Not that they mean a thing anymore , get a brain dump pass the test crash the system.

 

[KenMeans]

I can pretty much guarantee you that the issue is in the CM i'll try to find more info for you

Ken Means

http://www.tel-is.com

"I find that the harder I work, the more luck I seem to have."
- Thomas Jefferson (1743-1826)

 

[KenMeans]

this is what I was talking about trk group 99 and 100 are the sip trunks from the 2 application servers you can use private or public it just depends on how you set it up.

Ken Means

http://www.tel-is.com

"I find that the harder I work, the more luck I seem to have."
- Thomas Jefferson (1743-1826)

 

[montyzummer]

Ok so update , i noticed in the cstrace logs that it was failing due to "ims not enabled on trunk" so i changed the SIG group to IMS enabled and hey presto all working fine


ACSS (UC/SBCE/SM/SME)

Not that they mean a thing anymore , get a brain dump pass the test crash the system.

 

[KenMeans]

Thanks

Thats a new one i have not seen before


Ken Means

http://www.tel-is.com

"I find that the harder I work, the more luck I seem to have."
- Thomas Jefferson (1743-1826)

 

[montyzummer]

Yes agreed , really weird , thankyou for your help at least its sorted.

ACSS (UC/SBCE/SM/SME)

Not that they mean a thing anymore , get a brain dump pass the test crash the system.