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Well since it dumps them back to the main menu they can always hit zero, however the counter is a good idea!

Here is what I build and it seems to work perfectly: Menu option 2?? (all my individual extensions are in the 230-269) Generic command Set CPxx Value; Data Variable 0; With Value $KEY Test Variable $CP0 against Extn Variable Extension linked to Transfer by Extn (Destination $KEY) Variable No...

Absolutely awesome, thank you so much!

Okay so I found the Test Variable and placed it between 2?? and the Transfer call. I selected the Varible to Match as $KEY and under "select the parameter to test the variable against I checked Extn. I then linked Extension to transfer and no match back to the Menu. However after saving and...

I have a VMPro module setup with Menu option 2?? this links to Destination $KEY Source $CLI This transfers just fine as long as the extension exists. If the extension does not exist it just silenty drops the call. Is there a way to bounce them back into the menu if the extension doesn't exist?

Thank you, I have the 8_0_8_29 but looks like they have 8.0.4002 now

Sweet, I like to be sure. I've downloaded 56 just havn't installed it yet, guess I'll patch it up and see where it goes. About how long does it normaly take to apply the patch and reboot?

8.0?

amriddle: twice seemed to be when making SIP calls, once something else. Westi: I thought that might be the answer, I need to obtain an update licence to upgrade correct?

Damn thing has rebooted 3 times in the lasthour, here are all the traces: 18447412mS CMCallEvt: 0.5883.0 -1 BaseEP: NEW CMEndpoint f4e0904c TOTAL NOW=8 CALL_LIST=2 18447412mS PRN: Begin Stack Trace 18447412mS PRN: pc=81630000 18447412mS PRN: lr=f043d580 18447412mS PRN: findfunc...

This is an IP500 running 8.0(18) We use VMPro however the CompactFlash is mounted (holdmusic.wav being pulled from there) It has a Mix of SIP Trunks and Analog Trunks inbound with Analog/Digial and VOIP extensions.

Does this mean anything to anyone: 3512823mS PRN: Begin Stack Trace 3512823mS PRN: pc=81630000 3512823mS PRN: lr=f043d580 3512823mS PRN: findfunc f043d52c f043e414 f043e974 f031b1b4 f03241c8 f0324544 f0a06d14 f06f2834 f06fbe20 f06fbfc0 3512823mS PRN: findfunc f06fc498 f06fc67c...

Here is how I set it up: When they press transfer and dial 9 it triggers an ARS that matches N; to N Line group ID 7 Line 18 (SIP) Channel 1 is outgoing line group 7 (the only one) its local URI, Contact and display name are all the same; 8665516300 (number changed) If the operator was to start...

Anyone? I guess this isn't an issue?

Anybody else have an issue transferring CID blocked calls? If not can you show me the trace so I can see the difference, perhaps we can discuss URI settings?

Yes, I spent about an hour on the phone with them yesterday only to have them confirm that the "anonymous@anonymous.invalid" was causing the issue and they couldn't correct it. I was hoping for some sort of 'Don't used called # when trasfering call' option :)

IP500 8.0(18) When a call comes in and the CID is presented we are able to press transfer, dial 9 then an off switch number and transfer the call without issue. However if a call comes in marked "Anonymous" (thru dialing *67 or carrier blocking) and we try to transfer it we get a call rejected...

Wow, I was wicked confused!

That is doable, the problem is since they can't list two SIP endpoints correctly we would need to wait for them to swap the sip bindings and restart the object. Our entire goal is seamless failover with just an email from the sonicwall notifying us of WAN failure.

Their SBC couldn't handle two SIP bindings that go to the same endpoint. We ended up with one way audio or Audio up one pipe and down another.