Hi list,

I experiencing a strange behaviour when transferring a call. The use case is
like this:
- Incoming call from Zap/1-1
- Routed to SIP phone SIP/1001
- The called user (SIP/1001) wants to redirect the call and presses "#"
- IVR (default setup) says "Transfer" and user gets dial tone
- User dials 1002
- IVR says "No such extension - please try again"
???

It seems that the 1st digit gets canceled out? Debugging the server output I
get (tried twice):

snip
--------------------------
Goto (incoming,s,70)
   -- Executing Goto("Zap/1-1", "sip_incoming|s|1") in new stack
   -- Goto (sip_incoming,s,1)
   -- Executing Dial("Zap/1-1", "SIP/1001||rtT") in new stack
   -- Called 1001
ioctl(ZT_LOADZONE) failed: Inappropriate ioctl for device
Failed to register zone 'United States / North America': No data available
   -- SIP/1001-08d8c668 is ringing
   -- SIP/1001-08d8c668 answered Zap/1-1
   -- Started music on hold, class 'default', on channel 'Zap/1-1'
   -- Playing 'pbx-transfer' (language 'en')
   -- Stopped music on hold on Zap/1-1
   -- Unable to find extension '' in context 'local_extensions'
   -- Playing 'pbx-invalid' (language 'en')
   -- parse_srv: SRV mapped to host alpha2.callcentric.com, port 5060
   -- Started music on hold, class 'default', on channel 'Zap/1-1'
   -- Playing 'pbx-transfer' (language 'en')
   -- Stopped music on hold on Zap/1-1
   -- Unable to find extension '001' in context 'local_extensions'
--------------------------
snip

The extension.conf:
snip
--------------------------
[local_extensions]
include => outgoing
; Local extensions
exten => 1001,1,Dial(SIP/1001,20,rtT)
exten => 1002,1,Dial(SIP/1002,20,rtT)
exten => 1003,1,Dial(SIP/1003,20,rtT)
--------------------------
snip
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to