CUC transfer call to PSTN external number issue on E1(Cause i = 0x80AF - Resource unavailable, unspecified)

[mehrdadelectron]

Dear Friends,

I have a 4 digit E1 link  on Cisco router 2800 and use CUC 11 for IVR and CUCM 11 as call manager,

my problem is that when I set an external PSTN number for a User Input Digit in Call-Handler on CUC, router call the destination number but the call drop after of-hook the external PSTN phone ,

for example I set digit 8 for call my mobile phone and set "Transfer to Alternate Contact Number" with prefix of 9 for outgoing call, but as I test ,I hear the MOH for a second and then hear nothing but my mobile phone start ringing, then when I pick up the mobile, the call became disconnect.

Note:

-CSS and partition is OK because the call always place.

-I attach the all debug that I think you may need.

-I think my problem is related to codec(as you can see in "show voice dsp voice" log(no codec define for outgoing transferd call) but if I change the Region and Device pool of trunk and Voice Mail port and also LineCodec of CUC but no success.

thanks a lot.

[Mohammed al Baqari]

Where are you calling the CUC from? Is it local desk or pstn. ? You need to
look at the codec between calling party and called party. CUCM will avoid
allocating DSP by using same codec for calling and called legs.

Also, from the error message is sent to mobile pstn or received from mobile
pstn. You might be out E1 channels

[mehrdadelectron]

Dear Mohammed,

first of all thank your for fast reply.

I call from PSTN to CUC(with analog telephon or mobile) and the E1 router use DID and dial-peer(H.323 gateway) for reach to CUCM that is integrated with CUC with SCCP, then cuc try to do outgoing call to transfer me to that destination and in this step call will drop. 

when I test from local voip phone to do same test all the things works fine.

as you mention I am agree with you that the problem is in codec but I created special "Region(related with all other Region by g711alaw preference) that assigned to--> Device pool(PUB_SUB_G711alow)" and put H.323 router gateway and Voice mail ports in that Device Pool but no success!

[Mohammed al Baqari]

What is the codec configured in your h323 dialpeer. If you didn't configure
one it points to g729 by default

[mehrdadelectron]

Dear Mohammed,

based on my configuration attachment(first post),I set g711 codec  for all h.323 gateways.

I have restarted all of my trunks , voice ports and all the related part and call problem solved but problem exist,I use a pstn number for FAX and now when a customer dial the company E1 number, and with CUC IVR transfer to FAX number, the FAX machine cannot revived and the customer gets "Communication Error" or "Error 388" on that, I draw the diagram for clarify the flow.

   

[mehrdadelectron]

no advice? :(

[R0g22]

I checked the logs, so the initial call gets complete but it fails post entering code which does not make sense. I am suspecting H.245 is not completing and once the timer expires the call is dropped. Grab the following in a single file, simultaneously for a single failed call -

debug h225 asn
debug h245 asn
debug voip ccapi inout
debug isdn q931

[mehrdadelectron · Accepted solution]

Dear Nupus,

I had two different problems,

1- when I called CUC number from outside with my mobile phone and the CUC transfer me to external number, the call dropped.

Resolvation :

as I check the "show voice dsp voice", & I found out that the problem is in codec and because of diffrent incomming (g711alaw) and outgoing(g711ulaw) in router and no transcoding, so I force the CUCM , CUC and Router to us g711alaw codec with Region, Advertised Codec and "codec command" and then restart all trunks in cucm, this problem solved.

2- when I use  2 FAX PSTN machine instead of mobile phone and the destination number, the machine give me the communication Error and FAX not revived in this situation.

Reservation:

I check the "show voice dsp voice" in this scenario too, and see the below result:

----------------------------FLEX VOICE CARD 0 ------------------------------
                           *DSP VOICE CHANNELS*

CURR STATE : (busy)inuse (b-out)busy out (bpend)busyout pending 
LEGEND     : (bad)bad    (shut)shutdown  (dpend)download pending

DSP   DSP                 DSPWARE CURR  BOOT                         PAK   TX/RX
TYPE  NUM CH CODEC        VERSION STATE STATE   RST AI VOICEPORT TS ABRT PACK COUNT
===== === == ========= ========== ===== ======= === == ========= == ==== ============
C5510 001 01 voice      24.3.3 busy  idle      0  0 0/0/0:15  31    0   10276/9935
C5510 001 02 voice      24.3.3 busy  idle      0  0 0/0/0:15  30    0    3397/2290

so I read the "Fax Relay Troubleshooting" Document and add "fax protocol pass-though g711alaw" command in related dial-peer and now in codec filed it shows "g711alaw" and the problem solved.

Thank you all.

https://www.cisco.com/c/en/us/support/docs/voice/fax-modem-over-ip/20227-faxrelay-tsguide.html