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Trouble phone2phone Cisco phone / trunk H323 / to SIP 3partie phones

danjor
Level 1
Level 1

Hi,

I do have on 2 different sites:

Site A :

Cisco phones 79xx and Softphone

2811 with CME7.1 IOS 12.4.24(T3)

Site B:

Cisco Phones 79xx and SIP third partie phones (DECT) using G711ulaw

1861 with CME7.1 IOS 15.0(1)M3

Calls between Site A (Softphone for testing over VPN IPsec Tunnel) and Site B from Cisco to Cisco works fine

Calls in site B from Cisco to SIP phones works fine

Calls from Site A Cisco softphone to  SIP phones does call but I not hear ring the call on the softphone, it just stays off hook and notting happens more, until I hangup the call.

When I do retry the call, an error message is given on site B (Log)

  %CALL_CONTROL-6-CALL_LOOP: The incoming call has a global identfier already present in the list of currently handled calls. It is being refused

First I was expecting a transcoding issue but I found on the forum that from phone to phone, there is no need to trancode.

Site A dialpeer

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

supplementary-service h450.12

h323    

  call start slow

sip 

  registrar server expires max 1200 min 300

dial-peer voice 18 voip

destination-pattern 5[34].

session target ipv4:192.168.92.1

dtmf-relay h245-alphanumeric

Site B Extract

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

supplementary-service h450.12

h323  

    call start slow

sip  

   registrar server expires max 1200 min 300

dial-peer voice 11 voip

destination-pattern [579]..

session target ipv4:192.168.71.1

dtmf-relay h245-alphanumeric

Any idea what happens?

Thanks for a short answer,

Daniel

Hi,

As I did not get an answer her, I did open a TAC case.

Lot of work but now it works...

Solution 1 site SIP third party:

Definition from the voice in alaw/ulaw

Solution multisite SIP third party:

Changing also the trunk to the sites as alaw/ulaw or add txcoder to the SIP sites

1 Reply 1

danjor
Level 1
Level 1

Hi,

just for information (as I didn't get an answer her...):

It's solved by a TAC case.

solution 1 site:

Change the communication on SIP side to ulaw (or alaw depending on the third party) for all communication.

Solution multisite multi third party:

Changind all the trunk connecting to the SIP site also in ulaw/alaw or use txcoder localy where the sip-phones are.