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Call disconnect after one ring

jaywydra
Level 1
Level 1

I have a CCM cluster at my main location running 3.3.3sr4a. My remote site has a PRI that I use for PSTN access. When I place a call from my main site to a PSTN location their phone rings one time. I hear ringback one time and then it goes busy. The call routes from my main site to the remote and out the PRI. My debugs show normal call clearing with no other obvious info. Any ideas what might cause this?

Thanks,

Jason

1 Accepted Solution

Accepted Solutions

Hello Jason,

According to the configs the codec that is there on the voip is g711ulaw.What is the codec between the two sites .Is it g711 or g729.

Try doing the following

Rtr(config)#voice class codec 1

Rtr(config-voice)#codec preferce 1 g711ulaw

Rtr(config-voice)#codec preferce 2 g729r8

Rtr(config-voice)#codec preferce 3 g729br8

dial-peer voice 1 voip

no codec

voice class codec 1

dial-peer voice 2 voip

no codec

voice class codec 1

Also what is the extn at the main site that you are using.

Thanks,

RAdhika

View solution in original post

16 Replies 16

rnarayana
Level 5
Level 5

Hello ,

Can you please post the debug isdn q931 output.

Also what type of gateway is it.H323 or MGCP.What type of connection do you have at central and at remote site ie is it 6608 blade or nmhdv etc.

Thanks,

Radhika

Here is a Q931. The first call is a successful call from another phone at the remote site (where the PRI is). The second call is from the main site (this is the one that fails). The Q931 looks identical to me aside from the numbers. Also, the first couple of lines show an error that I saw in the CCM trace. Everything else in the CCM trace looked normal. Disconnect reason is normal call clearing.

Thanks,

11/08/2004 13:03:40.985 CCM|***MsgTrans - Has error in known message -- Cause Value (ms) = A, IE (ieid) = 4, Reserved Corrupted (RC) = 0|<:STANDALONECLUSTER><:192.168.30.12>

Nov 8 15:04:47.110: ISDN Se1/0:23 Q931: Applying typeplan for sw-type 0x3 is 0x

2 0x1, Calling num 2622524003

Nov 8 15:04:47.118: ISDN Se1/0:23 Q931: Applying typeplan for sw-type 0x3 is 0x

2 0x1, Called num 14142582117

Nov 8 15:04:47.118: ISDN Se1/0:23 Q931: TX -> SETUP pd = 8 callref = 0x042B

Bearer Capability i = 0x8090A2

Standard = CCITT

Transer Capability = Speech

Transfer Mode = Circuit

Transfer Rate = 64 kbit/s

Channel ID i = 0xA98396

Exclusive, Channel 22

Progress Ind i = 0x8183 - Origination address is non-ISDN

Calling Party Number i = 0x2181, '2622524003'

Plan:ISDN, Type:National

Called Party Number i = 0xA1, '14142582117'

Plan:ISDN, Type:National

Nov 8 15:04:47.338: ISDN Se1/0:23 Q931: RX <- CALL_PROC pd = 8 callref = 0x842

B

Channel ID i = 0xA98396

Exclusive, Channel 22

Nov 8 15:04:48.228: ISDN Se1/0:23 Q931: RX <- ALERTING pd = 8 callref = 0x842B

Progress Ind i = 0x8488 - In-band info or appropriate now available

Nov 8 15:04:48.636: ISDN Se1/0:23 Q931: Applying typeplan for sw-type 0x3 is 0x

2 0x1, Calling num 2622524003

Nov 8 15:04:48.640: ISDN Se1/0:23 Q931: Applying typeplan for sw-type 0x3 is 0x

2 0x1, Called num 18473574676

Nov 8 15:04:48.640: ISDN Se1/0:23 Q931: TX -> SETUP pd = 8 callref = 0x042C

Bearer Capability i = 0x8090A2

Standard = CCITT

Transer Capability = Speech

Transfer Mode = Circuit

Transfer Rate = 64 kbit/s

Channel ID i = 0xA98394

Exclusive, Channel 20

Progress Ind i = 0x8183 - Origination address is non-ISDN

Calling Party Number i = 0x2181, '2622524003'

Plan:ISDN, Type:National

Called Party Number i = 0xA1, '18473574676'

Plan:ISDN, Type:National

Nov 8 15:04:48.716: ISDN Se1/0:23 Q931: RX <- CALL_PROC pd = 8 callref = 0x842

C

Channel ID i = 0xA98394

Exclusive, Channel 20

Nov 8 15:04:50.636: ISDN Se1/0:23 Q931: RX <- ALERTING pd = 8 callref = 0x842C

Progress Ind i = 0x8488 - In-band info or appropriate now available

Nov 8 15:04:50.772: ISDN Se1/0:23 Q931: TX -> DISCONNECT pd = 8 callref = 0x04

2C

Cause i = 0x8090 - Normal call clearing

Nov 8 15:04:50.796: ISDN Se1/0:23 Q931: RX <- RELEASE pd = 8 callref = 0x842C

Nov 8 15:04:50.804: ISDN Se1/0:23 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x

042C

Can you please post the configs of main site router.

Thanks,

Radhika

Attached is the router config for the main site.

Thanks,

Jason

Attached is the router config for the main site.

Thanks,

Jason

Hello Jason,

After seeing the configs I am confused whether it is a MGCP gateway or h323.

Can you please confirm what type of gateway it is.

Thanks,

Radhika

Radhika,

It seems like someone palnned to configure the router as a SRST box but didnt completed putting the call manager failover coomands.

Thats the reason you see MGCP command as well as a bunch of Dial-Peers defined there.

Thanks,

Arijit

This was an MGCP gateway that was recently converted to H323. Could the MGCP commands be the problem? CCM is set up for H323.

Thanks,

Jason

Hi,

Specify the codec type in the dial-peers. Its getting defaulting to the deafult codec.

Hope this helps.

Thanks,

Arijit

I would do few things.

1.Remove all the MGCP configs.Reset the gateway.

2,On the VOIP dial peer( ie DP 500) put the destination pattern for the internal exten ie destination-pattern 5... ( if your internal extn).Dial-peer 500 is operational down I think as there is no destination pattern configured.

3.Configure a voice class codec and apply voip dial-peer.

Also I thought at the central site it was a PRI .I see only copper trunks.

Thanks,

Radhika

Thanks for catching that missing destination pattern, though I don't think that is causing my problem. Bye the way, this is sort of a goofy set-up. The PRI is at the remote site while the CCM cluster is at the main site (which is the site I sent you the config for). I'm used to seeing the PRI at the same site as the CCM but not in this case. The test call we are placing is being made from the main site and the route pattern sends it to the remote PRI. I will remove the MGCP configs and reboot as you suggested. Also, I will add the codec statement. Just for clarification, will this call ever hit the 500 dial-peer at the main site? I didn't think it would.

Thanks,

Jason

Hello Jason,

If you are using the remote site PRI to send the calls yes you don't need the dial-peer 500 at the main site.How ever for the calls coming in is you are using the main site gateway yes you need this dial-peer.Yes the missing destination pattern is not causing the issue in your case.

MAke sure at the remote site you have a voip dial-peer with destination -pattern of the main site extn and session target being the callmanager .Also add the voice class codec in this dial-peer

If possible can youpost the configs of remote site also.

Thanks,

Radhika

The codec command is on the remote site voip peer. Attached is the remote site config.

Thanks,

Jason

Hello Jason,

According to the configs the codec that is there on the voip is g711ulaw.What is the codec between the two sites .Is it g711 or g729.

Try doing the following

Rtr(config)#voice class codec 1

Rtr(config-voice)#codec preferce 1 g711ulaw

Rtr(config-voice)#codec preferce 2 g729r8

Rtr(config-voice)#codec preferce 3 g729br8

dial-peer voice 1 voip

no codec

voice class codec 1

dial-peer voice 2 voip

no codec

voice class codec 1

Also what is the extn at the main site that you are using.

Thanks,

RAdhika