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New Member

Outgoing Call Problem on new series IPT

Hi,

We have CUCM 8.6.2. Voice Gateway is, 2921 which has E1 - PRI line. Voice gateway is H323. Has dial-peers on it.

The problem is, I can not make GSM calls from new series IPT (9971 - 8945).

I can call local and domestic calls but can not call some GSM numbers. We have also 7945 ad 7921 IPT. Outgoing calls works with them.

8945:

VGW2#show voice call status

CallID     CID ccVdb     Port       Slot/DSP:Ch Called #   Codec   MLPP Dial-peers

0xF1       6F0 0x2BDABEF8 0/0/0:15.1       0/1:1 *507420xxxx None     130/22

1 active call found

Succesfull Call :7945:

VGW2#show voice call status

CallID     CID ccVdb     Port       Slot/DSP:Ch Called #   Codec   MLPP Dial-peers

0xF3       7F0 0x2BDABEF8 0/0/0:15.1       0/1:1 *530939xxxx None     130/22

1 active call found

24 REPLIES
Cisco Employee

Outgoing Call Problem on new series IPT

Interesting. Calling restriction doesnt really depend upon any phone model.

1. Ensure that the CSS/Partition config is same like other working phones, if they are part of same DP.

2. What happens when 9971 user tries to call GSM numbers ? Does it just drop without ringing or the user hears any message etc ?

GP

New Member

Outgoing Call Problem on new series IPT

There could bee an issue with the bearer cap.  I had a similar issue with the Cius.  You can do a debug isdn q931 on the voice gateway to look at the barrier cap. 

On a standard phone like 7900 series the bearer cap is set to speech:

Bearer Capability i = 0x8090A2

  Standard = CCITT

  Transfer Capability = Speech 

 

On newer phones with video capibilities the Bearer Cap can be set to Unrestricted Digital

Bearer Capability i = 0x8890

  Standard = CCITT

  Transfer Capability = Unrestricted Digital

Some carriers cannot or wont accept this capability so the call is dropped. 

If you find your newer phones with Video is not using Speech capability you can force the gateway to use Speech by issuing the following on the voice port:

voice-port x/x/x:23

bearer-cap speech

I hope this helps.

New Member

Outgoing Call Problem on new series IPT

New series IPT is a little bit tricky. I had an issue with them last week. On our branch office,

7945 IPT was registering  but 8961 and 8945 IPT rejected by CUCM. There was a p2p link with headquarter and branchoffice.

I found the problem was an MTU issue. 7945 was bypassing ip fragmentation packets.

https://supportforums.cisco.com/docs/DOC-3797

I read this kb and did the debugging. Nothing wrong.

There is a problem with "9". I pick up line with 9 and then dial the number. I disabled 9 and gsm calls go directly to Voicegateway now.

Hall of Fame Super Silver

Outgoing Call Problem on new series IPT

Add "bearer-cap Speech" under the D-channle voice port, i.e.:

voice-port 0/0/0:23

bearer-cap Speech

HTH,

Chris

New Member

Re: Outgoing Call Problem on new series IPT

Hi,

I do not intend to interrupt the discussion but I am also having a very similar issue.

CUCM  8.6.2. Voice Gateway is, 2651XM which has E1 - PRI line and also the  H.323 Gateway. As we are still in interim in the process of migrating  from the old CUCM, we have Inter-cluster trunk to CUCM 6.0 where we the  gateway configured.

Outbound calls could be  established from any other phones 7945, 7965 etc. but recently getting  problem in making outbound calls from 8945. I have tried add "bearer-cap  Speech" under the D-channle voice port as advised and the outbound call  is now ringing on my mobile and I could take the call but I could not  hear anything. The calling phone 8945 is just showing as ringing and  finally rang out.

Please see attached output of debug isdn q931.

Regards,

Lay

VIP Super Bronze

Re:Outgoing Call Problem on new series IPT

Hi,

I have looked at your debugs and I can see two calls to the same number. The first call was disconnected from the gateway side with cause code 41. I suggest you look at your DSPs.

Show voice DSP group all.

Send the output of that command




Sent from Cisco Technical Support Android App

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New Member

Re: Outgoing Call Problem on new series IPT

Thanks aokanlowan. The following is the output of show voice dsp group all on this gateway

sh voice dsp group all

DSP groups on slot 0:

This command is not applicable to slot 0

DSP groups on slot 1:

This command is not applicable to slot 1

We have an another 2911 router with a PVDM3-128 DSP module in the same network here at HQ, providing central transcoding  and conferencing resources to the network. While each site has a router  with a 32- or 64-channel PVDM3 module, that 2911 router provides backup in  the event that all resources at a site router are consumed. And I have following on that router.

c2911-central-dsp#sh voice dsp group all

DSP groups on slot 0:

dsp 1:

  State: UP, firmware: 28.3.5

  Max signal/voice channel: 43/43

  Max credits: 645, Voice credits: 0, Video credits: 645

  num_of_sig_chnls_allocated: 0

  Transcoding channels allocated: 0

  Slot: 0

  Device idx: 0

  PVDM Slot: 0

  Dsp Type: SP2600

dsp 2:

  State: UP, firmware: 28.3.5

  Max signal/voice channel: 43/43

  Max credits: 645, Voice credits: 615, Video credits: 30

  num_of_sig_chnls_allocated: 0

  Transcoding channels allocated: 9

  Group: FLEX_GROUP_VOICE, complexity: FLEX

    Shared credits: 34, reserved credits: 0

    Signaling channels allocated: 0

    Voice channels allocated: 0

    Credits used (rounded-up): 0

  Group: FLEX_GROUP_XCODE, complexity: FLEX

    Shared credits: 0, reserved credits: 581

    Transcoding channels allocated: 0

    Credits used (rounded-up): 0

  Slot: 0

  Device idx: 0

  PVDM Slot: 0

  Dsp Type: SP2600

dsp 3:

  State: UP, firmware: 28.3.5

  Max signal/voice channel: 42/43

  Max credits: 645, Voice credits: 645, Video credits: 0

  num_of_sig_chnls_allocated: 0

  Transcoding channels allocated: 5

  Group: FLEX_GROUP_CONF, complexity: CONFERENCE

    Shared credits: 0, reserved credits: 323

    Codec: CONF_G711, maximum participants: 16

    Sessions per dsp: 8

    Conference sessions:

      Sess01: Credits allocated: 80

  Group: FLEX_GROUP_XCODE, complexity: FLEX

    Shared credits: 0, reserved credits: 323

    Transcoding channels allocated: 0

    Credits used (rounded-up): 0

  Slot: 0

  Device idx: 0

  PVDM Slot: 0

  Dsp Type: SP2600

DSP groups on slot 1:

This command is not applicable to slot 1

  0 DSP resource allocation failure

c2911-central-dsp#

Regards,

Lay

VIP Super Bronze

Re:Outgoing Call Problem on new series IPT

Sorry for the late response. Can you send a debug isdn q931 for a working call. Lets compare both and see whats breaking in the non working one

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VIP Super Bronze

Outgoing Call Problem on new series IPT

Can you also chck what the region setting between the affected phone(s) and the gateway is set to. You need to check this with the h33 dial-peer matched. The call is been diconnected from the gateway side after the connect. At the connect layer of the h25 set up, media capabilities are exchanged. So you m ay be having a capabilities issue..

You can test again and send debug h225 asn1 and debug h245 asn1

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New Member

Re: Outgoing Call Problem on new series IPT

Thanks again aokanlawon. Please find attached output of debug isdn q931 from a working 7945 phone. And output for debug h225 asn1 and debug h245 asn1 for 8945 which is not working. The gateway got almost frozen when a test call was performed from a working 7945 with those debugs ON. Should I try again to get the debug output for 'debug h225 asn1' and 'debug h245 asn1' from a working 7945?

Regards,

Lay

New Member

Re: Outgoing Call Problem on new series IPT

Hi aokanlawon,

I have reassigned this 8945 into a branch site where we have a SIP trunk and outbound calls from 8945 are tested working from there. Looks like I am missing something here in HQ ISDN gateway and another gateway router where we are doing resource and transcoding? Region setting and gateway set on 8945 were as same as other 7945 on the same site I suppose althought I am not exactly sure what you wanted to check.

Regards,

Lay

Hall of Fame Super Silver

Re: Outgoing Call Problem on new series IPT

Configure the PRI with bearer-cap as speech, such as:

voice-port 0/0/0:23

bearer-cap Speech

HTH,

Chris

New Member

Re: Outgoing Call Problem on new series IPT

Hi Chris,

I already have 'bearer-cap Speech' in in the PRI voice-port. I was getting fast busy tone before and with that in place, I could take the call on my test mobile but it kept ringing on the calling phone which is 8945. NB: It brokes the video capabiity but that is fine for now.

Regards,

Lay

New Member

Re: Outgoing Call Problem on new series IPT

Hi Guys,

I can now make successful outbound calls from 8945 after updating followings:

  • Change 'Video Call Bandwidth' to 'None' for the             corresponding region
  • From the Phone Configuration screen, change the Video             Capabilities parameter to 'Disabled'
  • Added 'bearer-cap Speech' under PRI voice-port

However, it apparently broke the video cabilitiy on our 8945 phones. I haven't made any changes recently except the Publisher CUCM had to be restarted due to a storage issue we had on the site. I wonder if I need to check a service in CUCM? I would like to bring back the video capability as well since I am also testing Cisco VCS for Telepresence. Thanks.

Regards,

Lay

VIP Super Bronze

Re: Outgoing Call Problem on new series IPT

Lay,

I had a look at the debugs and I can see that during TCS echange, the gateway does not send any codec capabilities in the TCS exchange. That is weird. This is why you get resource unavailable. I looked at another log in there dialled number "8521" and I can see the correct codec advertised (g711u and g711alaw)...For this call cucm responds with g711ulaw as the codec it can support.

However in this 8945 case there is no codec advertised to cucm from the gateway...Can you send your sh run please.

I also saw that you are using dtmf-relay rtp-nte on a h323 dial-peer...Why is that?

value MultimediaSystemControlMessage ::= request : terminalCapabilitySet :
    {
      sequenceNumber 1
      protocolIdentifier { 0 0 8 245 0 7 }
      multiplexCapability h2250Capability :
      {
        maximumAudioDelayJitter 20
        receiveMultipointCapability
        {
          multicastCapability FALSE
          multiUniCastConference FALSE
          mediaDistributionCapability
          {

            {
              centralizedControl FALSE
              distributedControl FALSE
              centralizedAudio FALSE
              distributedAudio FALSE
              centralizedVideo FALSE
              distributedVideo FALSE
            }
          }
        }
        transmitMultipointCapability
        {
          multicastCapability FALSE
          multiUniCastConference FALSE
          mediaDistributionCapability
          {

            {
              centralizedControl FALSE
              distributedControl FALSE
              centralizedAudio FALSE
              distributedAudio FALSE
              centralizedVideo FALSE
              distributedVideo FALSE
            }
          }
        }
        receiveAndTransmitMultipointCapability
        {
          multicastCapability FALSE
          multiUniCastConference FALSE
          mediaDistributionCapability
          {

            {
              centralizedControl FALSE
              distributedControl FALSE
              centralizedAudio FALSE
              distributedAudio FALSE
              centralizedVideo FALSE
              distributedVideo FALSE
            }
          }
        }
        mcCapability
        {
          centralizedConferenceMC FALSE
          decentralizedConferenceMC FALSE
        }
        rtcpVideoControlCapability FALSE
        mediaPacketizationCapability
        {
          h261aVideoPacketization FALSE
        }
        logicalChannelSwitchingCapability FALSE
        t120DynamicPortCapability FALSE
      }
      capabilityTable
      {

        {
          capabilityTableEntryNumber 34
          capability receiveRTPAudioTelephonyEventCapability :
          {
            dynamicRTPPayloadType 101------------------------------------------------using dtmf-relay rtp-nte
            audioTelephoneEvent "0-16"
          }
        },
        {
          capabilityTableEntryNumber 32
          capability receiveAndTransmitDataApplicationCapability :
          {
            application t38fax :
            {
              t38FaxProtocol udp : NULL
              t38FaxProfile
              {
                fillBitRemoval FALSE
                transcodingJBIG FALSE
                transcodingMMR FALSE
                version 0
                t38FaxRateManagement transferredTCF : NULL
                t38FaxUdpOptions
                {
                  t38FaxMaxBuffer 200
                  t38FaxMaxDatagram 72
                  t38FaxUdpEC t38UDPRedundancy : NULL
                }
              }
            }
            maxBitRate 144
          }
        },
        {
          capabilityTableEntryNumber 25
          capability receiveAndTransmitDataApplicationCapability :
          {
            application nonStandard :
            {
              nonStandardIdentifier h221NonStandard :
              {
                t35CountryCode 181
                t35Extension 0
                manufacturerCode 18
              }
              data '52747044746D6652656C6179'H
            }
            maxBitRate 0
          }
        },
        {
          capabilityTableEntryNumber 31
          capability receiveUserInputCapability : hookflash : NULL
        },
        {
          capabilityTableEntryNumber 30
          capability receiveUserInputCapability : dtmf : NULL
        },
        {
          capabilityTableEntryNumber 27
          capability receiveUserInputCapability : basicString : NULL
        },
        {
          capabilityTableEntryNumber 17
          capability receiveAudioCapability : nonStandard :
          {
            nonStandardIdentifier h221NonStandard :
            {
              t35CountryCode 181
              t35Extension 0
              manufacturerCode 18
            }
            data '436C65617243686964'H
          }
        }
      }
      capabilityDescriptors
      {

        {
          capabilityDescriptorNumber 1
          simultaneousCapabilities
          {

            {
              32,
              17
            },

            {
              34,
              30,
              27,
              25
            },

            {
              31
            }
          }
        }
      }
    }

Feb  4 00:56:16.953: H245 MSC OUTGOING ENCODE BUFFER::=

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New Member

Re: Outgoing Call Problem on new series IPT

Sorry for the late reply, aokanlawon. You are right codec was None on troubled calls from 8945 but not sure why.

I actually have 'dtmf-relay h245-alphanumeric' in dial-peers of the gateway while I have 'RFC 2833' in CUCM trunks:

-Inter-cluster trunk to CUCM 6.0

-H.323 Gateway on CUCM 6.0 (that points to the gateway)

Please find attached sh run from the gateway.

Regards,

Lay

New Member

Re: Outgoing Call Problem on new series IPT

Any thought, aokanlawon?

Regards,

Lay

VIP Super Bronze

Re: Outgoing Call Problem on new series IPT

Lay,

Sorry, I assumed you found the issue. You said codec was set to  none on the phones. What is the region setting between the phones and the gatewa.  Please check that.

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

Re: Outgoing Call Problem on new series IPT

Thanks aokanlawon, it looks like I am having an underlying issue with allocation of media resources in our design since we are also having 'resources unavailable' and call engaging issue with some sites. But please see below region settings:

In CUCM 6 old system -

Region                  Max Audio bit rate  

Trunk to CUCM 8.6          G.711          384          Use System Default

Site-HQ                           G.711          Use System Default     Low Loss

In CUCM 8.6 new system -

Trunk to CUCM 6           64 kbps (G.722, G.711)      Use System Default      Use System Default

Site-HQ                        64 kbps (G.722, G.711)       None                           Use System Default

Thanks again.

Regards, Lay

VIP Super Bronze

Outgoing Call Problem on new series IPT

Lay,

Is there an intecluster trunk btween two CUCM? I dont understand. How does the old system come into play? Are you sending calls over ICT? Can you describe your call flow again and where what phone is

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

Re: Outgoing Call Problem on new series IPT

Thanks aokanlawon. Yes, we have an interculster trunk between two CUCM. There are a couple of sites on the old CUCM 6 and some are on the new CUCM 8.6. Some sites have been migrated to the new CUCM 8.6. At HQ, we have:

-CUCM 6 (one Publisher and one Subscriber) & CUCM 8.6 (one Publisher and two Subscribers)

-an intercluster trunk (SIP) between two CUCMs

-H323 gateway configured on CUCM 6 that use ISDN trunk (Cisco 2651XM 12.4(15)T with VWIC-1MFT-E1, 59-03 Voice AIM with 4 DSPs, NM-2V= C2600 4 port Voice PM and VIC-2BRI-S/T-TE)

-a router that does media resources such as transcoding (Cisco 2911 15.1(4)M4 with PVDM3 DSP DIMM with 128 Channels)

and in branch sites, we have

-CUBE configured on Cisco routers (2800, 2900) with SRST

-Some sites have ISDN trunk and some have SIP trunk with backup PSTNs

-Some branch site routers do not have MTP, CFB, XCoding locally and come back to HQ to do them

(This is where I am struggling , I doubt that media resources here at HQ runs out when we have many calls coming back from branch sites for media resources and have 'resources unavailable' problem.)

We couldn't make outbound calls more than four yesterday getting that problem. RTMT suggests there are some low diskspace on old CUCM 6 servers, so I have freed up some space on both old CUCM 6 Publisher and Subscriber and restart them using 'utils system restart' and I haven't seem the problem anymore today. However, I suspect I still have an underlying design issue in the way we allocate media resources.

Can you please advise if there is a way I can verify media resources one various calls (i.e. interoffice calls, outbound calls from different sites)? Any other advice you want to share from your experience with such migration would also be much appreciated.

Regards, Lay

VIP Super Bronze

Re: Outgoing Call Problem on new series IPT

Lay,

Hope you are well this morning...

Where do we begin...:)

First of all if you are running low on disk space, that suggests a bigger issue. I think you should try and ensure you migrate all the remaining phones on that cluster as soon as you can. Keep an eye in RTMT on CPU, IOwait utilization especially IOWait as it can affect call processing.

For media resources, it is best practice to keep them local to sites. The reason is because for things like Transcoders and a G711 only MTP, G711 bandwidth will transverse the WAN when these resources are invoked which could be a problem if you dont have enough bandwidth across your WAN/MPLS network.

You can definitely monitor media resources usage using either RTMT of using show commands on the media gateway.

You can monitor transcoder, MTP, CFB allocation on RTMT.

General rule for setting up media resources...

1. Configure Software Media resources and hardware media based resources in a different MRG. If you have a CUCM SW conference bridge and a hardware based conf bridge in the same MRG. When a device invokes a CFB, CUCM allocates devices in the same MRG on a round robin fashio. If the device CUCM allocates is a SW CFB and this is a G729 conference call, it will fail. This is why you should separate them using different MRG

2. Configure separate MRG for MTP and transcoders and ensure MRG ofr MTP is first in your MRGL. MTp and transcoders are not the same. There are things that inly MTP can do. If you put them in the same MRG and CUCM allocates a xcoder when it actually needs an MTP eg for dtmf mismatch, then the call will also fail

3. Ensure that the region setting between your devices and the Xcoder is set to G711. If a device needs to invoke a xcoder, and the region setting to the xcodr is set to G729, the call will fail

4. Similarly if you are using software based MTP in CUCM or you have configured it in IOS, ensure that the region set on the MTP is what your phones are configured to that MTP. In case of CUCM it can only do G711, so ensure the region settings on your phone is set to G711.

Let me know if you have any more queries

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

Re: Outgoing Call Problem on new series IPT

Thanks very much again for the comprehensive explanation, aokanlawon. These are really helpful to ones like me who has been struggling to understand them. I will check all the best practices of media resources against existing and arrange to migrate phones across ASAP.

Can you also please advise how I can monitor transcoder, MTP, CFB allocation using RTMT? and 'show media resource status' only show you the status of registration of those resources but not details? Is there any other useful commands I can use?

Regards, Lay

VIP Super Bronze

Re: Outgoing Call Problem on new series IPT

Yo use RTMT to monitor resources do the following...

1. login to RTMT

2. On the RTMT tab go to system>performance>open performance monitoring

3. Exapnd the tab by the sides to see all the defined counters

4. Select the server you want to monitor

5. Select the Media resource on that server you want to monitor e.g cisco transcode device, cisco MTP device

6. Select the counter you want to monitor on the device e.g ResourceTotal, ResourceActive (just double click on it). Once you have clicked on it, it will appear on the grid on the right

To monitor resoruces on gateway.. use show sccp connection: below an example of a xcoding session on a gateway

ERT01#sh sccp conn
sess_id    conn_id      stype mode     codec   sport rport ripaddr

536967492  537049324    xcode sendrecv g729ab  31246 16384 10.54.74.169    

536967492  537049323    xcode sendrecv g711u   29680 28214 10.205.140.20

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