11-08-2004 02:04 PM - edited 03-13-2019 06:57 AM
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
Solved! Go to Solution.
11-10-2004 07:23 AM
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
11-08-2004 02:22 PM
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
11-09-2004 07:17 AM
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
11-09-2004 08:54 AM
Can you please post the configs of main site router.
Thanks,
Radhika
11-09-2004 01:25 PM
Attached is the router config for the main site.
Thanks,
Jason
11-09-2004 01:26 PM
11-09-2004 01:34 PM
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
11-09-2004 01:46 PM
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
11-09-2004 02:35 PM
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
11-09-2004 01:41 PM
Hi,
Specify the codec type in the dial-peers. Its getting defaulting to the deafult codec.
Hope this helps.
Thanks,
Arijit
11-09-2004 02:51 PM
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
11-09-2004 03:15 PM
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
11-09-2004 07:18 PM
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
11-10-2004 06:54 AM
11-10-2004 07:23 AM
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
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