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Not receiving the 486 message from CUCM to Genesys via SIP trunk.

I have setup where Genesys is used along with CUCM 9.1

Below is the snapshot how it will look for call flow.

PRI----V.G----CUCM---SIP trunk (created in CUCM)-----Geneys server.

Query here is for outbound call from SIP softphone to PSTN, where if the PSTN user cancel the call.

the SIP phone is still assuming the call is continuing and after 40 sec its getting disconnected.

after looking in to the sip traces... it looks like that SIP trunk from cucm is not sending the user busy message 486....

(checked in V.G and its giving user busy)...but in the CUCM its not getting sent to the genesys...

After some time in genesys server itself send the 480 Temporarily Not Available message...

I assume I  should get the 486 message from CUCM to genesys when the PSTN party disconnect the call without answering.

Please assist.

1 ACCEPTED SOLUTION

Accepted Solutions

Re: Not receiving the 486 message from CUCM to Genesys via SIP t

From logs what i can see is after one min call legs stops transmitting and receiving packets.

You certainly need to check this Genesys support for this behavior , as far as I know  there is no problem either with the CUCM or with VG.

1477 : 1515 22820290ms.1 +0 pid:0 Originate  connecting

dur 00:01:15 tx:3765/602400 rx:3387/541760

IP 10.129.0.45:32596 SRTP: off rtt:0ms pl:67720/140ms lost:0/1/0 delay:55/55/65ms g711ulaw TextRelay: off

1477 : 1514 22820290ms.2 +0 pid:0 Originate  active

dur 00:01:16 tx:3387/568856 rx:3838/614080

Tele 0/3/0:15 (1514) [0/3/0.31] tx:76760/76760/0ms g711ulaw noise:-68 acom:3  i/0:-64/-62 dBm

1477 : 1515 22820290ms.1 +0 pid:0 Originate  connecting

dur 00:01:26 tx:4330/692800 rx:3387/541760

IP 10.129.0.45:32596 SRTP: off rtt:0ms pl:67720/140ms lost:0/1/0 delay:55/55/65ms g711ulaw TextRelay: off

1477 : 1514 22820290ms.2 +0 pid:0 Originate  active

dur 00:01:30 tx:3387/568856 rx:4515/722400

Tele 0/3/0:15 (1514) [0/3/0.31] tx:90290/90290/0ms g711ulaw noise:-68 acom:3  i/0:-67/-61 dBm

Rate all the helpful post.

Thanks

Manish

10 REPLIES

Not receiving the 486 message from CUCM to Genesys via SIP trunk

Hello Rajkumar,

when the PSTN caller cancels the call, what is the isdn message do you get? Generally you will receive annoucement saying "the caller is not ready to take the calls".

For ex: If you receive the announcement from provider, the same has to be forwarded to Genesys, not the 486 message.

could you please post the below debugs for a test call?

1) debug isdn q931

2) debug voice ccapi inout

3) debug ccsip message

Also collect the detailed ccm traces for that test call. Please include calling, called numbers & time of call.

//Suresh

Please rate all the useful posts

//Suresh Please rate all the useful posts.

Re: Not receiving the 486 message from CUCM to Genesys via SIP t

Can you enable "debug Isdn q931" on VG and see if PSTN is not sending any PI message like this below..

ISDN Se0/0/0:15 Q931: call_disc: PI received in disconnect; Postpone sending RELEASE for callid 0x8041

Thanks

Manish


Not receiving the 486 message from CUCM to Genesys via SIP trunk

Good point Manish. If the PI is received in the disconnect as you mentioned, then we need to issue the voice call disc-pi-off command in global configuration mode and check the behaviour.

Please rate all the useful posts

//Suresh Please rate all the useful posts.

Re: Not receiving the 486 message from CUCM to Genesys via SIP t

Guys the issue was resolved after the SIP softphone in Genesys Interaction workspace option refer enable was made true.

after that we were able to hear the CRBT (ringback tone) as well as the 486 message was flowing properly.

now there is new issue with genesys soft phone is that when call is connected to mobile user then after approx 1 min of conversation, no party can hear eachother (only call remains connected but no voice).

for this i check the Transcoder and MTP and both are  working during the call.

sess_id    conn_id      stype mode     codec   sport rport ripaddr

33560517   50382033     xcode sendrecv g711u   23298 27324 10.129.69.80

33560517   50382032     xcode sendrecv g711u   24722 8624  10.129.5.112

since early offer is selected from my side.

One more point i want to add ....when i tried the Xlite sip phone registered with genesys sip server, eveything works fine.

Getting

CSeq: 102 BYE

Reason: Q.850;cause=16 in the bye message.


Not receiving the 486 message from CUCM to Genesys via SIP trunk

Hi RajKumar,

Cause=16 is normal call clearing , it do not indicate any error.

Check the value set is ME-SE. service parameters-->active server-->call manager---Min-se ? The initial INVITE  from Genesys contains MIN-SE=90 .

Can you also attach SIP logs  when you call from Xlite soft phone and post the output of "show voip rtp connections" of

genesys soft phone before and after one min.

Thanks

Manish

Re: Not receiving the 486 message from CUCM to Genesys via SIP t

thanks Manish for helping me here.

I'm aware cause =16 is normal call clearing, but the concern here is that no party disconnect the call, altough the RTP session gets disconnected reulting in the normal call clearing.

for the same i have attached some more logs, Genesys interaction space toolbar is very customizable.

I have attached the X-lite as well genesys call flow.

Please have a look.

Also scenario here is when the SIP phone (interaction workspace) is without the refer enabled (true) option.

we get false ringback tone and if pstn party rejects the call there is no 486 message coming and its continous to ring with false ring.

after the  refer enabled made true, the actual PSTN party call back ring tone (PSTN ringback wav file) was received, but the call would last for only 1 min approx, (only call reamins but no voice, rtp lost).

In X-lite all good, no issues.

Also the MIN-SE value in CUCM is 1800 (default), its 90 in Genesys SIP server. however the same vlaue appears in X-lite too.

Please let me know if you need more info.

Re: Not receiving the 486 message from CUCM to Genesys via SIP t

Hi RajKumar,

I have checked logs for both Xlite and Genesys SIP Phone , there is no such difference which highlights any alteration. The media IP is correct for RTP stream ( To and From) on both these calls.

When you enable "refer enabled" , i think there is some proxy or firewall come into play.

Can you check one more thing.... When you make outbound call to a mobile , run this command "show call active voice brief" on VG and see the tx and rx parameters there.Verify which one is increasing and which one is not , that may highlight which side is actually blocking RTP stream.

Thanks

Manish

Re: Not receiving the 486 message from CUCM to Genesys via SIP t

Please find the attached logs.

Re: Not receiving the 486 message from CUCM to Genesys via SIP t

From logs what i can see is after one min call legs stops transmitting and receiving packets.

You certainly need to check this Genesys support for this behavior , as far as I know  there is no problem either with the CUCM or with VG.

1477 : 1515 22820290ms.1 +0 pid:0 Originate  connecting

dur 00:01:15 tx:3765/602400 rx:3387/541760

IP 10.129.0.45:32596 SRTP: off rtt:0ms pl:67720/140ms lost:0/1/0 delay:55/55/65ms g711ulaw TextRelay: off

1477 : 1514 22820290ms.2 +0 pid:0 Originate  active

dur 00:01:16 tx:3387/568856 rx:3838/614080

Tele 0/3/0:15 (1514) [0/3/0.31] tx:76760/76760/0ms g711ulaw noise:-68 acom:3  i/0:-64/-62 dBm

1477 : 1515 22820290ms.1 +0 pid:0 Originate  connecting

dur 00:01:26 tx:4330/692800 rx:3387/541760

IP 10.129.0.45:32596 SRTP: off rtt:0ms pl:67720/140ms lost:0/1/0 delay:55/55/65ms g711ulaw TextRelay: off

1477 : 1514 22820290ms.2 +0 pid:0 Originate  active

dur 00:01:30 tx:3387/568856 rx:4515/722400

Tele 0/3/0:15 (1514) [0/3/0.31] tx:90290/90290/0ms g711ulaw noise:-68 acom:3  i/0:-67/-61 dBm

Rate all the helpful post.

Thanks

Manish

Not receiving the 486 message from CUCM to Genesys via SIP trunk

Thank You Manish.

I too agree with you.

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