Uisng SPA504G+SPA500S with Asterisk 1.6.2.17.3

Hello,

I have SPA504G with SPA500S(expansion module). I am using Asterisk 1.6.2.17.3. I am little confused when I get more than one call on SPA504G. I am unable to transfer a call to a particular destination when using the sidecar SPA500S.

Here is a simple scenario…

I pickup 1st call on SPA504G talking to the caller. I now get 2nd call on the same phone, I hit “answer” and pick up the call. This will put 1st call on hold. I am talking to caller on 2nd call and want to transfer the call to an extension(internal) say ext 1124. I see these option on the screen… ConfLX, xferLX, conf, xfer. I hit xfer and dial the extension number 1124(using the keybad of SPA504G). Extension 1124 picks up, I now see these options… ConfLX, xferLX, conf, xfer. I then hit xfer and voila the 2nd call is now connected to ext 1124. All ends well.

But if I do the same transfer using the buttons on the sidecar, it won’t work. Here is the scenario…

I pickup 1st call on SPA504G talking to the caller. I now get 2nd call on the same phone, I hit “answer” and pick up the call. This will put 1st call on hold. I am talking to caller on 2nd call and want to transfer the call to an extension(internal) say ext 1124. I see these option on the screen… ConfLX, xferLX, conf, xfer. I hit xfer and hit the button(configured as ext 1124) on the sidecar(SPA500S). Extension 1124 picks up, I now see these options… ConfLX, xferLX, conf, xfer. I then hit xfer and instead of connecting the call it asks me to “Enter the Number”.

This is where I dont know how to proceed.

Note: button on the sidecar for extension 1124 is configured as follows:

fnc=sd+blf+cp;sub=1124@xxx.xxx.xxx.xxx;nme=P Jo

Any ideas? I checked with the cisco agent and they asked to check with Asterisk :frowning:

Here is the log file when I try transfer using side car…Not sure why I see the highlighted error below…

CC_switchOnAudDev(1)
CC_switchOnAudDev(1)
PHN_setAudioPath(1)
~[cb] speed dial: 1124
[CMXHTTP] force stop wav
[CMXHTTP] force stop wav
[CMXHTTP] force stop wav
[CMXHTTP] force stop wav

Calling:1124@x.x.x.x:0
[2:2]AUD ALLOC CALL (port=16426)
AUD_startRtpRx(). [2:0]. Port 16426. Ip 0x0. Tx txPayloadType -1.
CC_eventProc:event = 90, lid=2, par=518
[0:5060]<<x.x.x.x:5060
[0:5060]<<x.x.x.x:5060
NOTIFY sip:151@10.1.0.x:5060 SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK2a2a28e4;rport
Max-Forwards: 70
From: sip:1124@x.x.x.x;tag=as3889efef
To: “Operator1” sip:151@x.x.x.x;tag=b6f929e259b24d2a
Contact: sip:1124@x.x.x.x
Call-ID: a9a19242-dd3e004a@10.1.0.x
CSeq: 112 NOTIFY
User-Agent: FPBX-2.9.0rc1(1.6.2.17.3)
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 221

I have a client using 15 of these phones experiencing the same problem. They are unable to transfer via the attendant console for second calls. Any resolution?

Not yet… I will be speaking to the Cisco rep this Friday. will post if something useful.

I am still not successfull at this. I spoke to cisco techie and he said I will have to make some configurational changes in Asterisk which they don’t support.

Any idea?