11-19-2008 01:11 AM - edited 03-15-2019 02:38 PM
Hi,
I am facing a call transfer problem between a site to site call and then transfered to a PSTN call.
Setup: site A:ISDN->UC520----------Site B:UC520->FXO
When a call is made From site A IP Phone to Site B ip phone, call is successfuly established. Then if the call is transfered to the PSTN line at siteB then call disconnects immediately after pressing the Transfer button on the IP Phone.
If the first call is made to the PSTN at the site B and then if the call is forwarded to the Site A IP Phone then it works.
When i did the "debug voice ccapi inout" in site B. It shows
"ssCTreRoutingNotSupported=FALSE" and then call is disconnected.
Pls Note: problem is only on the site B. If site B calls site A and if the call is transfered to the Site A PSTN Then it works perfectly.
Pls help...
Thanks
Manoj
11-19-2008 05:18 AM
Manoj,
Do you have the following on the Site B UC520?
telephony-service
transfer-pattern .T
Brandon
11-19-2008 05:28 PM
Hi Brandon,
Yes i have transfer-pattern .T in both the sites.
Does it create any problem? FYI, site A call transfer is working fine.
11-20-2008 05:47 AM
Can you post the config of both UC520s?
Brandon
11-20-2008 06:32 PM
11-21-2008 06:58 AM
Manoj,
It looks like you have a transcoder configured at Site A, but not at Site B. This is why calls success from B to A. You are sending calls across the WAN g.729 (this is the default for voip dial peers), so they would need to be transcoded to be sent out to the PSTN. You can either configure a transcoder at site B or change your config as follows to send calls from A to B as g.711:
Site A
dial-peer voice 86 voip
destination-pattern 1..
session target ipv4:10.86.10.1
dtmf-relay h245-alphanumeric h245-signal
codec g711ulaw
Site B
dial-peer voice 4 voip
incoming called-number 1..
codec g711ulaw
Hope this helps.
Brandon
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