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SIP incoming call with G722-64 codec not working

n.agtual12345
Level 1
Level 1

Hi, Guys.

Have setup cube sip trunk to ITSP, incoming and outgoing calls are working. Except for an incoming call with g722 codec and video h263 (just need voice call). The called number does not even ring. The caller informed that his using polycom phone.

Also, itsp provided 10 numbers for testing in which we can assigned to our phones but only the main number is working. When doing an incoming call, (dialing the other numbers except from the main number) can see always on the logs that itsp is always feeding the main number. I think it was because of the configuration under the sip-ua (register the maint number to a registrar)  but itsp informed that it was also their setup for other clients and is working. Appreciate your help on these.

 

Thanks

 

 

27 Replies 27

G'Day,

 

Thanks.

 

Everything is now working fine. Except for incoming and outgoing fax. Been able to ring the fax machines from each end and just disconnect. Not going to fax tone.

 

 

 

Please attach FXS port config , dial-peer config, debug ccsip message and debug isdn q931 for an incoming fax call.

 

Thanks

Manish

Hi,

 

Please see attachments for the dial-peer config, debugs, and simple diag.

 

1. xxxxxx_050714_01.txt

    - SG number to (61861514100/0861514100)

2. xxxxxx_050714_02.txt

    - (61861514100/0861514100) to (0865570007/0061865570007)

3. xxxxxx_050714_03.txt

    - 0865570007 to (61861514100/0861514100)

 

Thanks,

 

Can you configure this on H323 gateway.

 

Voice service voip
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw

 

voice-port 0/1/1
 cptone SG
 timeouts interdigit 6
 station-id number 61861514100

 

dial-peer voice 1001 pots
huntstop
destination-pattern 61861514100
port 0/1/1
forward-digits all

 

If that didnt work send accross "debug voice ccapi inout" only from H323 GW.

 

Thanks

Manish

Hi,

 

Below is already configured. Forgot to show it on the configs yesterday.

 

Voice service voip
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw

 

Have configured below. Still same unable to make incoming and outgoing fax.

 

voice-port 0/1/1
 cptone SG
 timeouts interdigit 6
 station-id number 61861514100

 

dial-peer voice 1001 pots
huntstop
destination-pattern 61861514100
port 0/1/1
forward-digits all

 

I think something to do with this (debugs from yesterday).

 

May  7 06:49:48.562 UTC: //39/91ECDDF58028/CCAPI/ccCallDisconnect:
   Cause Value=65, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
May  7 06:49:48.562 UTC: //39/91ECDDF58028/CCAPI/ccCallDisconnect:
   Cause Value=65, Call Entry(Responsed=TRUE, Cause Value=65)
May  7 06:49:48.562 UTC: //39/91ECDDF58028/CCAPI/cc_api_get_transfer_info:
   Transfer Number Is Null
May  7 06:49:48.562 UTC: //40/91ECDDF58028/CCAPI/ccCallDisconnect:
   Cause Value=65, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=65)
May  7 06:49:48.562 UTC: //40/91ECDDF58028/CCAPI/ccCallDisconnect:
   Cause Value=65, Call Entry(Responsed=TRUE, Cause Value=65)
May  7 06:49:48.562 UTC: //40/91ECDDF58028/CCAPI/cc_api_get_transfer_info:
   Transfer Number Is Null
May  7 06:49:48.586 UTC: //40/91ECDDF58028/CCAPI/cc_api_get_transfer_info:
   Transfer Number Is Null
May  7 06:49:48.590 UTC: //40/91ECDDF58028/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x48C05ECC, Tag=0x0, Call Id=40,
   Call Entry(Disconnect Cause=65, Voice Class Cause Code=0, Retry Count=0)
May  7 06:49:48.590 UTC: //40/91ECDDF58028/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent

 

Does "voice class codec 1 " contains g711ulaw ?

If yes then apply that on CUBE dial-peer 2001 and test.

 

Thanks

Manish

G'Day, Manish.

 

Yes voice class codec contains g711ulaw. Have applied and still does not work.

 

Just to check with you, do we need the 3 voip call legs must have the same codec in order to send/receive fax. So correct me if i'm wrong, so call leg 1 (on attached diagram) must be g729r8 also for fax to work. (The FoIP method we use is T.38)

 

Regards,

 

Noel


do we need the 3 voip call legs must have the same codec in order to send/receive fax - Yes

so call leg 1 (on attached diagram) must be g729r8 also for fax to work - If your telco supports g729r8 then you can use it or you can modify call leg 2 and 3 to use g711alaw as primary codec.

 

Hi, Manish.

 

Thanks.  Currenlty coordinating with are ITSP if they can fix the codec to g729r8. Also, tried to search on cisco for a document stating that same codec is need to t38 fax to work and I could not find one. Can you share the link if you have one.

 

Thanks you

I could not find document that you are looking for but here is one thread where Deji explained how fax did not works with transcoders.

 

Do rate useful posts.

 

I have looked at your logs and I can see that your call is invoking a xcoder. You cant have a xcoder with fax. The reason this is happening is because your call legs use different codecs. The leg to the ITSP is using G711alaw and the leg to cucm is using G729.

As Manish suggested, apply voice class codec 1 to the dial-peer 2001 and ensure the region setting between your sip trunk to cucm and the h323 gateway where the fax is terminated is set to G711.

Once you have done this, do another test call and send us the ff

1. debug ccsip messages from the cube

2. From the h323 gateway

Debug h225 asn1

debug h245 asn1

debug voip vtsp all

debug fax relay t30 all-level-1

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G'Day, Ayodeji.

I see, so xcoder does not work with fax. So, it means the 3 voip call legs must have the same codec in order to send/receive fax. So correct me if i'm wrong, so call leg 1 (on attached diagram) must be g729r8 also for fax to work. (The FoIP method we use is T.38)

 

Appreciate if you can provide me a cisco doc for this.

 

 

Tha'ts my over sight. Yes the call will not work because you need fast start and early-offer for h323-sip call to work if you you enable early-offer on CUBE.

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