Solved: Meet Me and External callers

liamcairns · ‎07-01-2008

Hi,

I am having a little problem setting up meet me. I have setup the conference bridge using this link

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_chapter09186a00801ecf8a.html

And have setup the Meet Me number in line with this link.

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_chapter09186a00801ecf96.html#76011

This all works fine internally but when someone outside tries to call in to an established conference they get in and them immediately cut off.

I have also setup meet me with unity using this solution

http://forum.cisco.com/eforum/servlet/NetProf?page=netprof&forum=Unified%20Communications%20and%20Video&topic=IP%20Telephony&CommCmd=MB?cmd=pass_through&location=outline@^1@@.2cbfa8c4

and this works great internally but externally it is the same problem when the caller gets released into the conference.

Any ideas? Am i missing something obvious?

Ayodeji Okanlawon · ‎07-01-2008

So from the debug you can see two things

1. Call manager is initiating the disconnect because the direction of the disconect is comming from CCM..

*Jul 1 10:15:45.424: ISDN Se0/0/0:15 Q931: TX -> DISCONNECT pd = 8 callref = 0

x8004

Cause i = 0x80AF - Resource unavailable, unspecified ...The arrow indicates which party is initiating the disconnect

2. Resource unavailable shows that you have codec compartibility issues...

Are you using a h.323 gateway..If you are then you need to setup the dial-peer directing calls to callmanager to negotiate for both g729 and g711 codecs..

This is an example..

1. Configure Voice class codec to use different codecs

voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g729r8

2. Apply it to Voip Dial-peer

dial-peer voice 100 voip

destination-pattern xx..

voice-class codec 1...(apply it here)

session target ipv4:x.x.x.x

secondly if you are using MGCP..then you have 2 options...

1. Ensure that your IP phones are using G.711 codecs, check the region settings

Please rate all useful posts

View solution in original post

ashok_boin · ‎07-01-2008

If you have assigned internal partition for your meet me number, then this would be the cause why external callers are failing to join into the conference. If yes, try testing by configuring partition for meet me number as "None".

Regards...

-Ashok.

With best regards...
Ashok

Ayodeji Okanlawon · ‎07-01-2008

This could be codec issue...

The best way to troubleshoot this, is to turn on debuggin on your voice gateway..

debug isdn q931....

make a test call into an already inititated conference session...

look at the casue code in the debug..if you dont know how to interpret this, post it out here, I will help you..

Please rate all useful posts

liamcairns · ‎07-01-2008

thanks for the reply's. I had thought about the partition and have it set to none but it made no difference.

Here is a copy of the debug as requested.

*Jul 1 10:15:45.352: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8 callref = 0x0004

Sending Complete

Bearer Capability i = 0x9090A3

Standard = CCITT

Transfer Capability = 3.1kHz Audio

Transfer Mode = Circuit

Transfer Rate = 64 kbit/s

Channel ID i = 0xA98385

Exclusive, Channel 5

Progress Ind i = 0x8483 - Origination address is non-ISDN

Calling Party Number i = 0x2183, '7881830133'

Plan:ISDN, Type:National

Called Party Number i = 0xA1, '3126'

Plan:ISDN, Type:National

*Jul 1 10:15:45.368: ISDN Se0/0/0:15 Q931: TX -> CALL_PROC pd = 8 callref = 0x

8004

Channel ID i = 0xA98385

Exclusive, Channel 5

*Jul 1 10:15:45.424: ISDN Se0/0/0:15 Q931: TX -> DISCONNECT pd = 8 callref = 0

x8004

Cause i = 0x80AF - Resource unavailable, unspecified

*Jul 1 10:15:45.600: ISDN Se0/0/0:15 Q931: RX <- RELEASE pd = 8 callref = 0x00

04

*Jul 1 10:15:45.600: ISDN Se0/0/0:15 Q931: TX -> RELEASE_COMP pd = 8 callref =

0x8004

*Jul 1 10:15:50.196: ISDN Se0/0/0:15 Q931: TX -> DISCONNECT pd = 8 callref = 0

x8001

Cause i = 0x8090 - Normal call clearing

*Jul 1 10:15:50.412: ISDN Se0/0/0:15 Q931: RX <- RELEASE pd = 8 callref = 0x00

01

*Jul 1 10:15:50.412: ISDN Se0/0/0:15 Q931: TX -> RELEASE_COMP pd = 8 callref =

0x8001

I noticed that the resource is unavailable but any ideas why?

thanks again

Liam

Ayodeji Okanlawon · ‎07-01-2008

So from the debug you can see two things

1. Call manager is initiating the disconnect because the direction of the disconect is comming from CCM..

*Jul 1 10:15:45.424: ISDN Se0/0/0:15 Q931: TX -> DISCONNECT pd = 8 callref = 0

x8004

Cause i = 0x80AF - Resource unavailable, unspecified ...The arrow indicates which party is initiating the disconnect

2. Resource unavailable shows that you have codec compartibility issues...

Are you using a h.323 gateway..If you are then you need to setup the dial-peer directing calls to callmanager to negotiate for both g729 and g711 codecs..

This is an example..

1. Configure Voice class codec to use different codecs

voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g729r8

2. Apply it to Voip Dial-peer

dial-peer voice 100 voip

destination-pattern xx..

voice-class codec 1...(apply it here)

session target ipv4:x.x.x.x

secondly if you are using MGCP..then you have 2 options...

1. Ensure that your IP phones are using G.711 codecs, check the region settings

Please rate all useful posts

liamcairns · ‎07-01-2008

Hi aokanlawon,

That worked and calls can go directly into the conference. Nice one. Thanks for that.

However this has thrown up another problem. As mentioned before I was using unity to transfer the calls to the conference and that bit was working ok previously. Now when prompted to press 1 to enter the conference, it does not accept the input from my mobile. This part worked fine previoulsy but now the new codecs appear to not be accepting the input!

Any ideas?

Ayodeji Okanlawon · ‎07-01-2008

Can you post your gateway config and secondly does this work for internal users or internal users cant press 1 too?

Please rate all useful posts

Ayodeji Okanlawon · ‎07-01-2008

do you have this command on your dial-p

dtmf-relay h245-alphanumeric

Please rate all useful posts

Jaime Valencia · ‎07-01-2008

do you have configured dtmf relay on the dial-peer you're using?

HTH

javalenc

if this helps, please rate

HTH

java

if this helps, please rate

liamcairns · ‎07-01-2008

Thanks for all your help. Added in the following lines and all seems ok now

dtmf-relay h245-alphanumeric

ip qos dscp cs5 media

ip qos dscp cs4 signaling

no vad

Thanks again

liamcairns · ‎07-01-2008

Sorry aokanlawon & javalenc,

I am a bit new to the forums and ratings and wanted to give points to both and individual responses. Did not realise that by ticking that the first one resolved my problem that i would not be able to add any more points. It should also have been a 5 for aokanlawon.

Jaime Valencia · ‎07-01-2008

no prob, i gave 5 to aokanlawon on your behalf =)

javalenc

HTH

java

if this helps, please rate

Ayodeji Okanlawon · ‎07-01-2008

Javalenc,

Thanks for the rating...

Liamcairns,

Good to know everything is okay now.

Please rate all useful posts

cg100668 · ‎08-06-2008

Hi,

I read through the post concerning conference bridge access externally. I have a similar requirement. The meet me numbers can only be accessed by pressing the Meet Me soft key from any phone and dialing the Meet Me number to join the bridge. How can a similar functionality be achieved for users to dial a number and join a conference bridge. The customer has CCM 6.1.1 and MGCP gateway.

Chandrasen

Jaime Valencia · ‎08-06-2008

the FIRST user must press meetme and dial the conference DN to activate it (it must be an internal user, a PSTN user cannnot be initiate the meetme since it cannot send the softkey event needed), ANY other user must only dial the DN.

HTH

java

if this helps, please rate

HTH

java

if this helps, please rate