[Asterisk-Users] Call forwarding...

[Kanuri, Seshu (Company IT)]

Try changing SetCIDNum SetCallerID and use to SetCIDName as under:

Ex:
---
exten => s, 1, SetCallerID(${CALLERIDNUM})
exten => s, 2, SetCIDName(${CALLERIDNAME})
exten => s, 3, Dial(${ARG2}/${ARG1},${RINGSECS})
exten => s, 4, Voicemail(u${ARG1})
exten => s, 5, Hangup
exten => s, 101, Voicemail(b${ARG1})
exten => s, 102, Hangup
 
Seshu Kanuri


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Mark
Benson
Sent: Wednesday, May 18, 2005 6:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Call forwarding...

Sorry for posting this again, but it seems to have become attached to
another thread. Guess I replied to another message instead of starting a
new one...

Hi,

I'm trying to setup a call forwarding rule so that when an extention
doesn't answer the call is forwarded to my mobile.

I'm using voiptalk.org for incoming and outgoing calls and SIP phones
for extentions (so all IP based - no real phone lines).

I tried this (from voip-info.org wiki)...

exten => 1234,1,dial(sip/1234,20)
exten => 1234,2,playback(pls-wait-connect-call)
exten => 1234,3,Setvar(NewCaller=${CALLERIDNUM})
exten => 1234,4,SetCIDNum(0${CALLERIDNUM}) exten =>
1234,5,dial(${TRUNK}c/9871234321,20,r)
exten => 1234,6,SetCIDNum(${NewCaller})
exten => 1234,7,voicemail2(u1234 at default) exten =>
1234,101,voicemail2(b1234 at default)
exten => 1234,102,hangup

Mine looks like this...

exten => 08700688nnn,1,Dial(SIP/operator,1,t)
exten => 08700688nnn,2,playback(pls-wait-connect-call)
exten => 08700688nnn,3,Setvar(NewCaller=${CALLERIDNUM})
exten => 08700688nnn,4,SetCIDNum(0${CALLERIDNUM})
exten => 08700688nnn,5,Dial(${TRUNK}/07961106nnn,20,r)
exten => 08700688nnn,6,SetCIDNum(${NewCaller})
exten => 08700688nnn,7,Voicemail(u100)
exten => 08700688nnn,8,Hangup()
exten => 08700688nnn,101,Voicemail(b100) exten =>
08700688nnn,102,Hangup()

(where nnn is a real number)
The sip channel is set to time out quickly for testing.
And I don't appear to have the pls-wait-connect-call audio file - but
that isn't an issue for the time being...
The IAX2/0870nnnnn is the extention/device that calls go out on via
voiptalk... (my call provider)...
If I include the c/ in the TRUNK line I get...

   -- Executing Dial("IAX2/08700688nnn at 217.14.132.nnn:4569-1",
"c/07961106nnn|20|r") in new stack
May 18 10:37:40 WARNING[24416]: channel.c:1957 ast_request: No channel
type registered for 'c'
May 18 10:37:40 NOTICE[24416]: app_dial.c:927 dial_exec_full: Unable to
create channel of type 'c' (cause 66)

Asterisk shows this from the moment the sip channel is considered not to
have answered (1 sec)...

   -- Nobody picked up in 1000 ms
   -- Executing Playback("IAX2/08700688nnn at 217.14.132.nnn:4569-1",
"pls-wait-connect-call") in new stack
May 18 10:20:26 WARNING[24416]: file.c:486 ast_openstream_full: File
pls-wait-connect-call does not exist in any format May 18 10:20:26
WARNING[24416]: file.c:790 ast_streamfile: Unable to open
pls-wait-connect-call (format ilbc): No such file or directory May 18
10:20:26 WARNING[24416]: app_playback.c:83 playback_exec: 
ast_streamfile failed on IAX2/08700688nnn at 217.14.132.nnn:4569-1 for
pls-wait-connect-call
   -- Executing SetVar("IAX2/08700688nnn at 217.14.132.nnn:4569-1",
"NewCaller=01202843nnn") in new stack
   -- Executing SetCIDNum("IAX2/08700688nnn at 217.14.132.nnn:4569-1",
"001202843nnn") in new stack
   -- Executing Dial("IAX2/08700688nnn at 217.14.132.nnn:4569-1",
"/07961106nnn|20|r") in new stack
May 18 10:20:26 WARNING[24416]: channel.c:1957 ast_request: No channel
type registered for ''
May 18 10:20:26 NOTICE[24416]: app_dial.c:927 dial_exec_full: Unable to
create channel of type '' (cause 66)  == Everyone is busy/congested at
this time (1:0/0/1)
   -- Executing SetCIDNum("IAX2/08700688nnn at 217.14.132.nnn:4569-1",
"01202843nnn") in new stack
   -- Executing VoiceMail("IAX2/08700688nnn at 217.14.132.nnn:4569-1",
"u100") in new stack
   -- Playing '/var/spool/asterisk/voicemail/default/100/unavail' 
(language 'en')

Again - I'm not worried about the audio file warning - I can fix that
later... I guess this is the important bit...

   -- Executing Dial("IAX2/08700688nnn at 217.14.132.nnn:4569-1",
"/07961106nnn|20|r") in new stack
May 18 10:20:26 WARNING[24416]: channel.c:1957 ast_request: No channel
type registered for ''
May 18 10:20:26 NOTICE[24416]: app_dial.c:927 dial_exec_full: Unable to
create channel of type '' (cause 66)  == Everyone is busy/congested at
this time (1:0/0/1)

The call then drops into voicemail...

I've tried various permuations but still no call is made to the mobile
number. Any ideas?

Cheers,

Mark

I should mention that I have tried using the call forward function of
the sip phones, but a) this means configuring the phones and some are
remote and behind firewalls and b) It doesn't work... 
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