Mobile calls to E&M circuits fail - IP Phones work fine

Answered Question
Feb 2nd, 2010

Hi Everyone,

I have a strange problem here which I hope you can give me some guidance / help on.

We have a Call Manager Cluster running 4.1.3 with Pub, 6 subs and 2 TFTP severs at the main site and 1 sub at the remote site. We have a number of gateways at both sites, but most are at the main site and have links to BT, O2 Mobile and Vodafone Mobile amongst others. We also have a number of E&M gateways with links to Radio equipment.

Each site runs G711 intra site, but run G.729 between sites.

The problem I have is that although calls from any IPT phone at either site can call the Radio equipment OK, call from an O2 mobile or a Vodafone mobile fail.

Any call from a mobile to the Radio equipment will seize the circuit, but then immediately drops out. The E & M circuits must be configured correctly as calls from IPT phones are fine dialling the same Route Pattern.

If we get the mobile to dial into an IVR and then dial the same code from the IVR, then the call succeeds.

I have run a number of debugs both the E1 gateway where the Mobile E1 terminates and on the E&M gateway where the Radio circuits terminate and it seems to be failing due to 'Resource Unavailiable.

The following is a 'cut-down' ISDN Q931 debug from the E1 gateway.

Calling Party = 07956 80007 - Called Party = 5760 1010

Jan  5 10:56:08.165: ISDN Se1/1:15 Q931: RX <- SETUP pd = 8  callref = 0x0001
Sending Complete
Bearer Capability i = 0x8090A3
  Standard = CCITT
  Transfer Capability = Speech 
  Transfer Mode = Circuit
  Transfer Rate = 64 kbit/s
Channel ID i = 0xA98381
  Exclusive, Channel 1
Facility i = 0x9FAA068001008201008B0100A10D02010402012903050000000000
Facility i = 0x9FAA068001008201008B0100A11402010306082B0C02885302010603050080000000
Facility i = 0x9FAA068001008201018B0100A14E02010206082B0C028853020100303F812A7E280D012703000210001115002A3136372A3139393030312A36352A31333535231107002A31392A55238211030F02000000807E7FFF80610000008080
Facility i = 0x9FAA068001008201008B0100A113020101020155300B8209005C08000000000000
Calling Party Number i = 0x0080, '07595680007'
  Plan:Unknown, Type:Unknown
Called Party Number i = 0x80, '57601010'
  Plan:Unknown, Type:Unknown
Locking Shift to Codeset 4
Codeset 4 IE 0x31  i = 0x85

Jan  5 10:56:08.193: ISDN Se1/1:15 Q931: TX -> CALL_PROC pd = 8  callref = 0x8001
Channel ID i = 0xA98381
  Exclusive, Channel 1

Jan  5 10:56:10.073: ISDN Se1/1:15 Q931: TX -> PROGRESS pd = 8  callref = 0x8001
Progress Ind i = 0x8088 - In-band info or appropriate now available

Jan  5 10:56:10.137: ISDN Se1/1:15 Q931: TX -> DISCONNECT pd = 8  callref = 0x8001

Cause i = 0x80AF - Resource unavailable, unspecified
Jan  5 10:56:10.289: ISDN Se1/1:15 Q931: RX <- RELEASE pd = 8  callref = 0x0001
Cause i = 0x81A2 - No circuit/channel available

Jan  5 10:56:10.313: ISDN Se1/1:15 Q931: TX -> RELEASE_COMP pd = 8  callref = 0x8001

If I check the E&M gateway, I can see the call being seized with both both dial peers (pots - 104 & voip - 200):-

(Dialled number 57601010 has now been translated to 5761010) and g729 codec being assigned.

EM_Gateway#sh voice call sta
CallID     CID  ccVdb      Port        DSP/Ch  Called #   Codec    Dial-peers
0x5C51     3522 0x832C0C48 1/1         0/1    *5761010    None     200/104
1 active call found

EM_Gateway#sh voice call sta
CallID     CID  ccVdb      Port        DSP/Ch  Called #   Codec    Dial-peers
0x5C51     3522 0x832C0C48 1/1         0/1    *5761010    g729r8   200/104
1 active call found

'debug voice ccapi' on the E&M Gateway shows the call being disconnected, firstly with Cause Value=16, then with Cause Value=47.

( I believe cause value of 47 means 'Resource Unavailiable)

Feb  2 11:53:27.074: //23639/006E7590B704/CCAPI/ccCallDisconnect:
   Cause Value=16, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
Feb  2 11:53:27.074: //23639/006E7590B704/CCAPI/ccCallDisconnect:
   Cause Value=16, Call Entry(Responsed=TRUE, Cause Value=16)
Feb  2 11:53:27.074: //23639/006E7590B704/CCAPI/cc_api_get_transfer_info:
   Transfer Number Is Null
Feb  2 11:53:27.474: //23639/006E7590B704/CCAPI/cc_api_call_feature:
   Feature Type=6, Interface=0x832C0C48, Call Id=23639
Feb  2 11:53:27.482: //23639/006E7590B704/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x832C0C48, Tag=0x0, Call Id=23639,
   Call Entry(Disconnect Cause=16, Voice Class Cause Code=0, Retry Count=0)
Feb  2 11:53:27.482: //23639/006E7590B704/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
Feb  2 11:53:27.486: //23638/006E7590B704/CCAPI/ccCallDisconnect:
   Cause Value=47, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=47)
Feb  2 11:53:27.486: //23638/006E7590B704/CCAPI/ccCallDisconnect:
   Cause Value=47, Call Entry(Responsed=TRUE, Cause Value=47)
Feb  2 11:53:27.486: //23638/006E7590B704/CCAPI/cc_api_get_transfer_info:
   Transfer Number Is Null
Feb  2 11:53:27.490: //23638/006E7590B704/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x81AD4108, Tag=0x0, Call Id=23638,
   Call Entry(Disconnect Cause=47, Voice Class Cause Code=0, Retry Count=0)
Feb  2 11:53:27.490: //23638/006E7590B704/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent

A debug voice vtsp all shows the following:- (disconnect with Cause Value = 16)

Feb  2 11:55:58.647: //23647/xxxxxxxxxxxx/VTSP:(1/1):-1:0:1/vtsp_dsm_bridge_status_cb:
Feb  2 11:55:58.651: //23647/xxxxxxxxxxxx/VTSP:(1/1):-1:0:1/vtsp_process_event: 
   [state:S_SETUP_REQ_PROC, event:E_CC_DISCONNECT]
Feb  2 11:55:58.651: //23647/xxxxxxxxxxxx/VTSP:(1/1):-1:0:1/act_disconnect: 
   Cause Value=16

If a run a 'sh voice dsp' on the E&M gateway, I see port 1/1 making DSP 001 01 Busy

EM_GATEWAY#sh voice dsp

DSP  DSP             DSPWARE CURR  BOOT                         PAK     TX/RX
TYPE NUM CH CODEC    VERSION STATE STATE   RST AI VOICEPORT TS ABORT  PACK COUNT
==== === == ======== ======= ===== ======= === == ========= == ===== ============
C549 000 00 g711ulaw  4.1.40 Idle  Idle      0  0 0/0       NA     0  9643/11242
C549 000 01 g711ulaw  4.1.40 Idle  Idle      0  0 0/1       NA     0 57583/66685
C549 001 00 g711ulaw  4.1.40 Idle  Idle      0  0 1/0       NA     0 776374/10186
                                                                     41         
C549 001 01 None      4.1.40 Busy  Idle      0  0 1/1        1     0 12955246/156
                                                                     17399      
C549 002 00 g711ulaw  4.1.40 Idle  Idle      0  0 2/0       NA     0         1/1
C549 002 01 g711ulaw  4.1.40 Idle  Idle      0  0 2/1       NA     0         1/1
C549 003 00 g711ulaw  4.1.40 Idle  Idle      0  0 3/0       NA     0         1/1
C549 003 01 g711ulaw  4.1.40 Idle  Idle      0  0 3/1       NA     0         1/1

Active Voice Call details

C549 001 00 None      4.1.40 Busy  Idle      0  0 1/1        1

So my thoughts are:-

1) This is some sort of codec issue

2) This is some sort of capabilites exchange where the mobil eis making a request for service, but the E&M gateway cannot service the request, so is droping the call.

Any thoughts?

I have this problem too.
0 votes
Correct Answer by markbatts about 6 years 11 months ago

Hi,

I had quick look through, and i think it is a codec issue.You say that the codec is g729 , is the e&m gateway in the same location as the mobile gateway?

If so should this not be g711?

I can only assume that the gateway is trying g711 and then going out g729 , its trying to insert a transcoder and then failing.Have you checked MRGL etc for both gateways.


cheers

Mark

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Correct Answer
markbatts Tue, 02/02/2010 - 05:00

Hi,

I had quick look through, and i think it is a codec issue.You say that the codec is g729 , is the e&m gateway in the same location as the mobile gateway?

If so should this not be g711?

I can only assume that the gateway is trying g711 and then going out g729 , its trying to insert a transcoder and then failing.Have you checked MRGL etc for both gateways.


cheers

Mark

Peter Bishop Tue, 02/02/2010 - 05:34

Hi Mark,

The E1 gateway is at the main site, the E & M gateway is at the remote site. Becuase of restricted bandwidth, we use G.729 between sites, but G.711 at each site for 'intra' site calls.

IP Phone calls between sites use G.729 ok nad an IP phone call at the main site to the E & M gateway at the remote site also work OK.

I've just tried a mobile call to my IP phone - (gateway at main site to my IP phone at main site) - then calling the radio phone - (IP phone at main site to E&M gateway at remote site) - but then when I try to transfer the mobile to the Radio phone, I get NU on the mobile.

I've checked calling search spaces / partitions of all gateways / phones in the call scenario and should all be allowed to call each other. I've just tried setting up a single transcoder in a MRG / MGRL for the E&M gateway, but no difference yet.

Peter Bishop Tue, 02/02/2010 - 06:38

Hi Mark,

All working now thanks.

This is the (relevent) configuration I had on the E&M gateway:-

voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g711alaw

codec preference 3 g729r8

dial-peer voice 200 voip

preference 1

destination-pattern .T

voice-class codec 1

voice-class h323 1

session target ipv4:10.201.104.50

no vad

I did a search on NetPro for 'Cause Value = 47' and saw a few similar issues with one recommending trying 'codec negotiate' under the 'voice class codec' command.

Unfortunately, probably due to the version of IOS, no other commands were avaliable, so knowing we only use G.711 and G.729, I added a further code preference as follows:-

voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g711alaw

codec preference 3 g729r8

codec preference 4 g729br8

Now everything works as it should

Thanks for all you help. I'll now try and work out how to rate you answer!

Thanks,

Peter

Paolo Bevilacqua Tue, 02/02/2010 - 06:41

You can eliminate g711a that not used in Cisco world.

Stick to g711u if traffic goes on LAN only, or you have enough WAN BW.

Peter Bishop Tue, 02/02/2010 - 07:23

Legacy configuration before I started here! Bandwidth between sites is very restricted; (carries all Customer data, voice & video); so we use G.729 between sites.

We are going to 6.1.3 in 2 weeks time, so won't be making any other changes until after the upgrade.

Thanks for the advice & tips

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