[cisco-voip] Pressing Confrn Causes Call To Disconnect?

Curtis, Adam ACurtis at pizzagalli.com
Fri Jul 31 10:04:38 EDT 2009


Over the couple of days our users have noticed that if they are on a call and want to conference someone else in they press the Confrn button, dial the number and the person answers, but when they press the Confrn button again one or both parties are dropped from the call. Is this a codec issue? If so, what log traces should I be looking for? I tried this with a remote site, cell phone, and internal phones and it always disconnects but if I conference in voicemail and they auto attendant it works fine.. Any ideas would be greatly appreciated!

Thank you,

Adam Curtis
Pizzagalli Construction Company
IT Support Technician
(802) 651-1319


From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
Sent: Thursday, July 30, 2009 10:31 AM
To: Leslie Meade
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] intermittent missing outbound ring tone

Hi Leslie,

The next messages after setup are the ones that determine how the user hears ringback.  If the PSTN sends Proceeding, Alerting, or Facility with a Progress Indicator IE and that Progress Indicator ends in 8 then the IOS gateway and CM must cut through audio immediately.  This allows the user to hear any inband tones provided by the transit PSTN gear or by the far end customer premise equipment.

I am not aware of a single one-size-fits-all solution to this.  Some services, such as American Express, send PI=8 to play prompts to caller at the beginning of the call.  In this case it is necessary to cut through the audio.  You may call other CPE equipment that provide PI=8 but then fail to provide inband audio of either announcement or ringback.

That said there are a few tools at your disposal:
http://www.cisco.com/en/US/tech/tk1077/technologies_tech_note09186a0080094c33.shtml

/Wes

On Wednesday, July 29, 2009 6:07:07 PM , Leslie Meade <lmeade at signal.ca><mailto:lmeade at signal.ca> wrote:


Since converting my gateways to H.323, I am getting users reporting that sometimes they do not get a ring tone when they ring people.
It is not every time, not every number. It can be long distance or international, it does not happen all the time.  Any ideas

I am running ccm7.1 and my gateway is a 2821 H.323

4743212: *Jul 29 14:42:22.891 UTC: ISDN Se0/0/0:23 Q931: pak_private_number: Invalid type/plan 0x0 0x0 may be overriden; sw-type 13
4743213: *Jul 29 14:42:22.891 UTC: ISDN Se0/0/0:23 Q931: pak_private_number: Invalid type/plan 0x0 0x0 may be overriden; sw-type 13
4743214: *Jul 29 14:42:22.891 UTC: ISDN Se0/0/0:23 Q931: Sending SETUP  callref = 0x5AF1 callID = 0xDBB7 switch = primary-ni interface = User
4743215: *Jul 29 14:42:22.895 UTC: ISDN Se0/0/0:23 Q931: TX -> SETUP pd = 8  callref = 0x5AF1
        Bearer Capability i = 0x8090A2
                Standard = CCITT
                Transfer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA9838B
                Exclusive, Channel 11
        Facility i = 0x9F8B0100A11502012E020100800D4B72797374792057696465656E
                Protocol Profile =  Networking Extensions
                0xA11502012E020100800D4B72797374792057696465656E
                Component = Invoke component
                        Invoke Id = 46
                        Operation = CallingName
                                Name Presentation Allowed Extended
                                Name = Krysty
        Display i = 'Krysty'
        Calling Party Number i = 0x2181, '604899XXXX'
                Plan:ISDN, Type:National
        Called Party Number i = 0x80, '01152818421XXXX'
                Plan:Unknown, Type:Unknown
4743216: *Jul 29 14:42:22.931 UTC: ISDN Se0/0/0:23 Q931: RX <- STATUS pd = 8  callref = 0xDAF1
        Cause i = 0x80E328 - Information element not implemented

Leslie







________________________________






_______________________________________________

cisco-voip mailing list

cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>

https://puck.nether.net/mailman/listinfo/cisco-voip



-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20090731/e32ae094/attachment.html>


More information about the cisco-voip mailing list