Dead Audio for the calls transferred from Unity Connection to an Extension

[Abdul Jaseem]

Hi Experts,

We have an Auto Attendant configured on Cisco Unity Connection, once I dial the main line number I'm getting the greetings. It will say "Press 1" for HR Department and call get transferred to a DN. But when the DN answered, there is no audio.

PSTN Caller >> SIP Gateway >> CUCM >> CUC >> Auto Attendant (DN:1000) >> Greetings >> Transfer to DN: 1003

CUCM and CUC integrated via SIP.

Helps are much appreciated.

[Aseem Anand]

Hi,

Check the following:

(a) What kind of transfer method are you using on CUC? I would suggest using release to switch and then check if you get the same result.

(b). What happens when you press the message button the phone with extension 1003, you get the audio or not?

Aseem

[Abdul Jaseem]

Hi Aseem, Thanks for your response.

(a) I have used Release to Switch transfer

(b) 1003 can able to access VM by hitting VM button.

I tried internal calls as well, same issue.

DN:1004 >> Unity Connection AA DN:1000 >> Press 1 >> Transfer to DN:1003

Dead Audio for this situation too.

1004 able to dial 1003 and speak without any issue. Only calls transferred from Unity having the issue.

[Aseem Anand]

Hi,

Can you check the region settings between the CUC and the extension 1003 and make sure the codec is consistent and there is no need of Xcoder?

You can find the codecs supported by CUC under:

CUC >>>> telephony Integrations >>>> Port Group >>>> codec advertising

Also check in CUCM traces if the trunk is invoking an MTP and if yes then make sure the region settings between the trunk and the MTP and between MTP and the IP phone is correct keeping in mind that call manager based MTP resources only support G711.

Aseem

[Abdul Jaseem]

Hi, 

I really appreciate your help. Still I'm facing the issue.

I verified the configurations and all fine.

Here I will provide simulated the Lab scenario, please check if there is any issues.

Call Handler DID : 512-555-1000

Translation rules to convert 512-555-1000 to 1000 [rule 1 /^512555\(10..\)/ /\1/]

CTI Route Point DN: 1000 >> Forward all to Unity

Unity Call Handler DN : 1000

Call Handler Greetings: Thank you for calling ....., Press 1 for 1001 press 2 for 1002

Situation 1:

From 1001 I can dial Call Handler and press 2, the call get transferred to 1002 without any issues.

Scenario 2:

If a PSTN caller dial 512-555-100, he can able to listen the greetings, but when he hit 1, the phone 1001 will ring without a ring back to caller and also if 1001 picked up, there is no audio.

Configurations:

ITSP_Router

dial-peer voice 2 pots
 description PSTN PHONE 2
 destination-pattern 2148576980
 port 2/1
dial-peer voice 3 voip
 description ACCEPT CALL FROM UC COLLAB
 session protocol sipv2
 incoming called-number .
 dtmf-relay rtp-nte
dial-peer voice 4 voip
 description UC COLLAB DID
 destination-pattern 51255510..
 session protocol sipv2
 session target ipv4:142.100.64.253
 dtmf-relay rtp-nte

TEXAS_VOICE_GW

dial-peer voice 2 voip
 description SEND CALL TO CUCM
 destination-pattern 10..
 session protocol sipv2
 session target ipv4:142.100.64.11
 dtmf-relay rtp-nte
dial-peer voice 3 voip
 translation-profile incoming PSTN-INCOMING
 session protocol sipv2
 incoming called-number .
 dtmf-relay rtp-nte
dial-peer voice 4 voip
 destination-pattern .T
 session protocol sipv2
 session target ipv4:142.100.64.252

All devices are in same device pool.

[Aseem Anand]

Hi,

Can you make sure the trunk has an MRGL which contains an annunciator ?

Also, make sure you have the correct locale under the device pool assigned to the trunk?

Aseem

[Abdul Jaseem]

Hi Aseem,

I verified those and everything in place!

But still issue not yet resolved.

I really appreciate your interest on this issue.

[Evgeny Izetov]

It may have something to do with the fact that G.729 is the default codec for dial-peers in IOS. Try configuring all of your dial-peers with G.711 and see if that resolves it.

Also, what interface(s) are you binding your media to?

[YAARUB F. MILAD]

I have had a similar issue and it was because I mistakenly enabled digest authentication on SIP security profile. 

Only discovered that after collecting call traces on cucm.