IPT Call Transfer issue from PSTN to IP Phone.

[rajesh.kumar]

Hi

We have Cisco CUCM 8.5 and 2821 router ver 15.x. H323 pri gateway for CUCM.

HW CONF and TXCODE is configured on VG router.

Problem I am facing is,

PSTN--------->VG------------------->CUCM------------------->WAN LINK---------->(Phone C)

                                    (Phone A)(Phone B)

Main Office I have Cisco VG, CUCM and Phone A,Phone B at Branch I have Phone C

G711 codec used in Main Office and G729 codec is used across wan link.

HW MRGL is PhoneA, Phone B, Phone C

Incoming call to Phone A cant be trasfered to Phone C

VG, PhoneA,B are in MainOfiice Device Pool, Phone C is in WAN Device Pool.

All the phones are added to Hardware MRGL.

Note:

Hardware MRGL is PhoneA,PhoneB and PhoneC

HW CONF and TXCODE are registered with CUCM,  MRGL has these conf, txcode as well as MTP also added.

I can Call PhoneA and PhoneC and establish conference to PhoneB

I was not able to find out the reason, when I get incoming call from PSTN to MainOffice IP Phone, I cant transfer the call to Branch office phone.

However all the Phones(A,B,C) are added to Hardware MRGL.

VG TXCODE, CONF config

!

sccp local FastEthernet0/0

sccp ccm 172.16.108.21 identifier 2 priority 2 version 7.1

sccp ccm 172.16.116.21 identifier 1 priority 1 version 7.1

sccp

!

sccp ccm group 2

associate ccm 1 priority 1

associate ccm 2 priority 2

associate profile 2 register VG1-CONF

!

sccp ccm group 1

associate ccm 1 priority 1

associate ccm 2 priority 2

associate profile 1 register VG1-XCODE

!

dspfarm profile 1 transcode 

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

codec g729br8

codec g729r8

maximum sessions 12

associate application SCCP

!

dspfarm profile 2 conference 

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

codec g729r8

codec g729br8

maximum sessions 3

associate application SCCP

!

!

Pls. suggest

[Jonathan Schulenberg]

Questions:

1. Does the H323 gateway have the MRGL as well? You mention all the phones do but not the gateway.
2. Is the H323 gateway using an MTP to achieve fast start; or, normal slow start behavior? Sometimes an MTP plus a transocder on the same call gets complicated.
3. When does the consult transfer to Phone C fail: immediately when you dial the extension or after some ringing when the phone goes off-hook to answer? What do you hear: the annunciator or reorder?
4. Why are you using a transcoder? Cisco IP phones support both G.711 and G.729 natively. The gateway just needs to include it during the H.245 OpenLogicalChannel sequence. You would add a voice class codec to the CUCM-facing H.323 dial-peers instead of hard-coding the codec directly on the dial-peer itself.

voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g729r8

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[rajesh.kumar]

Hi  Jonathan Schulenberg

1.H323 GW is using MRGL-Hardware (same as phone MRGL)

2.MTP is unchecked in H323 GW

3.Transfer is consulted transfer. When I transfer call, PSTN called hear MOH tone. Moment I press trasfer again call get disconnect.(Trasfer across WAN link with codec g729)

same thing happens during Conference call across the WAN using PSTN.

4.We are using transcoder, becoz, we have PRI at MainOffice with DID range.

MainOffice VG to MainOffice phones are using g711 codec.

MainOffice VG to Branch Office using g729 codec.

MainOffice to Branch IP phone use g729 codec.

We had issue of conference call with mixed codec calls.

In voice gateway,

!

voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g729r8

codec preference 3 g729br8

This is used for voip dial-peer.

Any other suggestion? Is there any hardware MTP required ? as of now we dont have MOH issue for any calls during hold. Only Conference and call transfer with PSTN call.

Note: If Branch IP phone call MainOffice IP phone, call establishes with g729 codec.

then If I try to add another PSTN call from Branch ip phone (which uses main office h323 VG) to the call.

moment I press conf. button again, pstn call disconnects.

Pls. let me know any hint.

Thanks

Rgds

Rajesh

[Ayodeji Okanlawon]

Rajesh,

Can you do a test transfer call and send....

1. Your sh run
2. Debug VoIP ccapi input
3. Debug h225 asn1
4. Debug h245 asn1

Please let us know what the calling, called and xfered number is

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[rajesh.kumar]

Hi  aokanlawon

I have attached the shor run and debug voice ccapi inout.

Calling from pstn number 0502252586 and called internal DID 8225

call established. But when I transfer call to extension 8240 which is across the WAN, then call failed.

from 8225 i can transfer the pstn call to 8240 (across wan g729) extn 8240 rings, but call disconnect the moment i pick up remote phone extn 8240.

pls. suggest

rgds

rajesh

[Ayodeji Okanlawon]

You need to send..these debugs also..Please send them

Debug h225 asn1

Debug h245 asn1

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[rajesh.kumar]

Hi

I could not take

Debug h225 asn1

Debug h245 asn1

becoz, router become too slow when i enable it.

I have no problem in call transfer from pstn to ip phone extn within main office ip phones.

All uses same g711 codec.

I have collected CUCM SDL log. and attached here.

[Ayodeji Okanlawon]

Rajesh,

I have looked at the logs and there is not much I can see other than the call to 8225 uses codec g711ullaw..Because the transfer happened between phones in CUCM, we will need to see the CUCM SDI traces....If you can send CUCM SDI traces we can look at it further..The CUCM trace you sent is not what is needed now

Apr 21 14:03:50.803: //716395/1E869EAAB2AC/CCAPI/cc_api_caps_ack:

   Destination Interface=0x49602EB4, Destination Call Id=716394, Source Call Id=716395,

   Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_VOICE(0x2), Vad=OFF(0x1),

   Modem=ON(0x2), Codec Bytes=160, Signal Type=2, Seq Num Start=1385)

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

[Ayodeji Okanlawon]

Sorry  I didnt know you have attached the SDI logs, let me have a look

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

[rajesh.kumar]

some more collected through rtmt

[Ayodeji Okanlawon]

Rajesh..I have looked at the logs in detail..Wao what an interesting setup you have...

+++Call comes in over h323 gateway to a SIP phone (9975) on extension 8225...CUCM invokes an MTP for this call leg.+++

Here is the RTP call flow for the first call leg. Both extension 8225 and caller 0502252586 sends their RTP stream to the MTP.

90502252586----g711ulaw----MTP---g711ulaw-----8225

Next thing is extension 8225 attempts to transfer the call to a SCCP phone on extension 8240. After the call succeeds, cucm attempts to complete the transfer to extension 8240..

Here we then see the issue...

The region between the MTP and Gateway is set to default and the region between the MTP and sip phone 8225 is set to default too..So we see that the call used g711ulaw as I showed in the diagram...

But from the logs below:

17:42:35.077 |MediaManager(10045857)::preCheckCapabilities, region1=Default, region2=DF-AAB, capCount1=6, capCount2=11|1,100,50,1.465260426^172.16.200.125^SEP001F9E245A81

17:42:35.077 |RegionsServer::MatchCapabilities -- kbps=8, capACount=6, capBCount=11

We can see that the SCCP phone is in region DF-AAB and the codec between this region and default region is G729.

Now the call look slike this..

90502252586----g711ulaw----MTP---g729-----8240..

Hence CUCM attempts to invoke a xcoder..

7:42:35.077 |MediaManager(10045857)::prepareConnectionsWithOneSavedConnectionWithParty1, savedConnRes=MTP xcoderReqd=1 xcodingSide=1|1,100,50,1.465260426^172.16.200.125^SEP001F9E245A81

But there CUCM couldnt find any xcoder configured..

17:42:35.078 |MRM::getXcodeDeviceGivenMrgl GETTING XCODE FROM DEFAULT LIST|1,100,50,1.465260426^172.16.200.125^SEP001F9E245A81

17:42:35.078 |MediaResourceManager::sendAllocationResourceErr - ERROR - no transcoder device configured|1,100,50,1.465260426^172.16.200.125^SEP001F9E245A81

Action plan..

ensure that you have a xcoder in the MRGL of the SCCP phone SEP001F9E245A81..

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[rajesh.kumar]

Hi

Here I have attached the Phone MRGL config, Hardware Xcode, conf and registration status.

I have specifically created mrgl-test for the testing phone, instead of disturbing other operational phones.

under mrgl-test, I have added all the hardware xcode and conf bridge. Also attached screenshot that those xcode status is registered with cucm.

Pls. suggest.

Rgds

Rajesh kumar

[Ayodeji Okanlawon]

Rajesh,

Have you dont the tests I suggested. I have just seen your MRGL config and here are my observations

1. Your MTP and Xcoder are in the same MRG..This is not right..You s hould configure a different MRG for MTP and Xcoder..In your MRGL ensure the MRG for MTP is at the top..

2. Your xcoder is in the default device pool so I assume its using the default region too...Hence the region between the phone 8240 (df-aab) and xcoder (default) = G729...You should put your xcoder in a seperate region and have the region matrix set to use G711 with the sccp phone

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[Ayodeji Okanlawon]

Can you do two things.

1. Put the sccp phone in the default region that the MTP and gateway ate in..test

2. Put the phone back in its df-aab region, ensure the region between the xcoder and the sccp phone (8240) is seto g711..test again


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[rajesh.kumar]

Hi Aokanlawon

I have configured as you suggested. Some reason it was not working. After I restart CUCM cluster Transfer and Conference between PSTN, Main Office and Branch Office started working.

Now the problem is Transcoder is being used for all the calls from PSTN to Branch Office.

PSTN--------->VG------------------->CUCM------------------->WAN LINK---------->(Phone C)

                                    (Phone A)(Phone B)

If PSTN caller Call DID number of Phone "C" directl,  which is located at Branch office. VG pvdm Transcoder is utilised.

VG pvdm is now fully utilised for normal calls, conference and trsnfer are running out of resources.

Main Office VG is in "Default" device pool and Branch office phones are in different DP. Region also different. g729 is the codec between "Default" DP region to Branh office IP phone region.

Pls. suggest.

Rgds

Rajesh kumar