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Remote Dialtone

November 10, 2014, 7:22 pm
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Looking for a little help for the missing piece of the puzzle.
I have system A and system B -- both Switchvox systems.
I want system A to be able to place an outbound call over system B's PRI.
I have system A and system B connected via IAX provider setups with the Peer Settings Host type on System A set to provider since I took "Host Setting" to mean what type of host is system B.
On system B, I have Peer Settings Host Type set to client since I take "Host Setting" to mean what type of host is system A. Perhaps I have those two switched. I have tried it both ways.
And perhaps what I am trying to do here is not possible. I have trunking set to yes on both systems and have been tweaking some of the other settings on both systems and running tests. Thus far I am getting an error message that tries to spell out the name of the system. I have an outbound dial rule applied to system A that uses a special 3-digit code and then the remainder must be 8 digits which would be a 9 and the 7 digit local number I want to dial. The 3-digit code is stripped. Thus I am sending 9xxx-xxxx from System A to System B.

Again, perhaps this is not possible or perhaps I am missing some small piece of the puzzle. I'll keep trying and tweaking but thought someone might be able to point out the error of my ways or simply tell me that I can't do what I'm trying to do.

Thank You.

Statistics : Posted by jreed1949 • on Mon Nov 10, 2014 9:22 pm • Replies 2 • Views 144
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Inbound call from register => line does not go to extension

November 19, 2014, 5:18 pm
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Hi,

On Asterisk 1.8.13.1~dfsg1-3+deb7u3

I have in a standard sip.conf a line that says

register => number:password@proxy/extension

I see asterisk registering to my proxy server, and inbound calls reach the asterisk box. however they go to "number" and not to "extension". And I only created dial plan entry for "extension" of course.

I did create the proxy peer in the sip.conf as well so it does go to the right context.

What am I obviously missing ?

Any ideas ?

Statistics : Posted by xblurone • on Wed Nov 19, 2014 7:18 pm • Replies 0 • Views 1
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Troubleshooting: Sorry that is not valid extension...

November 20, 2014, 10:44 am
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Figured what was the problem Image
Have to say that I am beginner with asterisk (inherited when joined company)
Steps how I figured:
-Checked "database show" command which showed some additional entry for "problematic" extension:
Code: /CFI/1020                                         : 33
/FME/1020                                         : 000

-performed "grep CFI *" and "grep FME *in /etc/asterisk/ to see where are those entry coming from
-found that in extensions.conf I have _*7X.,n,Set(DB(FME/${CALLERID(num)})=${EXTEN:2}) and *6X.,2,Set(DB(CFI/${CALLERID(NUM)})=${EXTEN:2}) respectively
-tried functions from different phone *6 (*7) and got response that those are "follow me" and "forward" options.
-used *6 and *7 to remove feature from that extension and "voila"

Of course no body knows who and how they enabled this but that is another story

Cheers,

Statistics : Posted by spricer • on Thu Nov 20, 2014 12:44 pm • Replies 2 • Views 65
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Troubleshooting: Sorry that is not valid extension...

November 20, 2014, 10:44 am
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Logs? In particular, sip set debug on output.

Statistics : Posted by spricer • on Thu Nov 20, 2014 12:44 pm • Replies 2 • Views 65
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Troubleshooting: Sorry that is not valid extension...

November 20, 2014, 10:44 am
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I am having problem with calling particular extension from any phone where response is always:
"Sorry that is not a valid extension, please try again."
Version is Asterisk 1.4.26.3 (on Debian 5.0.3 - 2.6.26-2-686)
Protocol in use is SIP.
Other info:
-Tried different phones (SNOM 320) - same
-Phone registers properly and I CAN initiate call from that extension
-If I purposely call some non-existent extension I can see log ( /var/log/asterisk/messages) entry but no logs for problematic extension whatsoever.
-Extension is configured exactly the same as all other ones in sip.conf
-Extension can be seen when "sip show peers" is invoked

I am running out of ideas Image

I would appreciate any hint or advice how can I troubleshoot this.
Thanks!

sip.conf:
Code: [1020] ; Spare
type=friend
username=1020
secret=
callerid=SIP/1020
host=dynamic
context=from-sip
limitonpeers=yes
call-limit=100
canreinvite=yes
vmexten=VoiceMail
allowsubscribe=yes
mailbox=1020@employees


extensions.conf
Code: [general]
;------------------------------------------------------------------
[macro-CF]
;Call Forwarding
exten => s,1,Set(CF=${DB(CFI/${ARG1})})
exten => s,n,GotoIf(${CF}?yes:no)
exten => s,n(yes),Goto(from-sip,${CF},1)
exten => s,n(no),NoOp()

[macro-FM]
;Follow Me
;ARG1=EXTEN,ARG2=CALLERIDNUM
exten => s,1,Gotoif(${DB_EXISTS(FME/${ARG1})}?yes:no)
exten => s,n(yes),Set(CALLERID(num)=${ARG2})
exten => s,n,Goto(from-sip,${DB(FME/${ARG1})},1)
exten => s,n(no),Set(CALLERID(num)=${ARG2})



[globals]
;------------------------------------------------------------------
CMD_REBOOT_SNOMS=/etc/asterisk/rebootallsnoms.sh


[from-pstn]
;------------------------------------------------------------------
exten => s,1,Goto(from-pstn2,${EXTEN},1)

[from-pstn2]
include => ext-employees
include => ext-special
include => ext-management

include => welcome-holiday|*|*|24-31|dec
include => welcome-holiday|*|*|1-3|jan

include => welcome-opened|08:00-17:29|mon-fri|*|*

include => welcome-closed|17:30-07:59|mon-fri|*|*
include => welcome-closed|*|sat-sun|*|*

exten => i,1,Playback(invalid)
exten => i,n,Goto(main,1)

;External ringtone
exten => _10XX,2,SIPAddHeader("Alert-Info:<http://192.168.1.2/external.wav")

;All incoming calls enter this extension
exten => s,1,Goto(main,1)

exten => main,1,Answer()
exten => main,2,Gotoif(${DB_EXISTS(VM/CUSTOM)}?yes:no)
exten => main,3(yes),Background(welcome${DB(VM/CUSTOM)})
exten => main,4,Goto(7)
exten => main,6,Goto(7)
exten => main,7,WaitExten(60)
exten => main,n,Playback(invalid)
exten => main,n,Hangup()

exten => 0,1,VoiceMail(1099@employees)
exten => 0,n,Hangup()

exten => #,1,Directory(employees,ext-employees)
exten => #,n,Hangup()

[welcome-opened]
exten => main,5(no),Background(welcome1)
[welcome-closed]
exten => main,5(no),Background(welcome2)
[welcome-holiday]
exten => main,5(no),Background(welcome3)


[from-iax]
;------------------------------------------------------------------
exten => _.,1,Goto(from-iax2,${EXTEN},1)

[from-iax2]
include => ext-employees
include => ext-special
include => ext-outgoing

exten => i,1,Answer()
exten => i,n,Playback(invalid)
exten => i,n,Hangup()

;Internal ringtone for employees
exten => _10XX,2,SIPAddHeader("Alert-Info:<http://192.168.1.2/internal.wav")


[from-sip]
;------------------------------------------------------------------
;TEMP
exten => #66,1,Answer()
exten => #66,n,Set(TMP=${UNIQUEID})
exten => #66,n,Record(/rec/${TMP}.wav,5,300)
exten => #66,n,Playback(beep)
exten => #66,n,Playback(/rec/${TMP})
exten => #66,n,Hangup()
;--------------------
exten => _.,1,Goto(from-sip2,${EXTEN},1)

[from-sip2]
include => ext-employees        ;10XX
include => ext-special      ;1[2-4]XX
include => ext-management   ;15XX
include => ext-emergency   ;911
include => ext-outgoing         ;9.
include => pbx-features         ;*.
include => ext-iax      ;#.

exten => i,1,Answer()
exten => i,n,Playback(invalid)
exten => i,3,Hangup()

;Internal ringtone for employees
exten => _10XX,2,SIPAddHeader("Alert-Info:<http://192.168.1.2/internal.wav")



[ext-iax]
;-----------------------------------------------------------------
exten => _#XXX,1,Dial(IAX2/asterisk-ls/${EXTEN:1})
exten => _#XXX,n,Gotoif($["${DIALSTATUS}" = "CHANUNAVAIL"]?err:end)
exten => _#XXX,n(err),Playback(err-noiax)
exten => _#XXX,n(end),Hangup()

;SAS Paris
;exten => _#85XX,1...



[ext-employees]
;------------------------------------------------------------------
exten => _10XX,1,Macro(CF,${EXTEN})
;Priority 2 is ringtone, defined in some contexts which include this one
exten => _10XX,n,LDAPget(CALLERID(name)=cidname)
exten => _10XX,n,Dial(SIP/${EXTEN},30,tTwWkK)
exten => _10XX,n,Noop(Dialstatus is ${DIALSTATUS})
exten => _10XX,n,Gotoif($["${DIALSTATUS}" = "NOANSWER"]?follow)
exten => _10XX,n,Gotoif($["${DIALSTATUS}" = "BUSY"]?busy)
exten => _10XX,n,Gotoif($["${DIALSTATUS}" = "CHANUNAVAIL"]?invalid:unavailable)

exten => _10XX,n(follow),Macro(FM,${EXTEN},${CALLERID(num)})
exten => _10XX,n,Goto(unavailable)

exten => _10XX,n(busy),VoiceMail(${EXTEN}@employees,b)
exten => _10XX,n,Hangup()

exten => _10XX,n(unavailable),VoiceMail(${EXTEN}@employees,u)
exten => _10XX,n,Hangup()

exten => _10XX,n(invalid),Playback(invalid)
exten => _10XX,n,Hangup()



[ext-outgoing]
;------------------------------------------------------------------
exten => _9.,1,Dial(DAHDI/g0/${EXTEN:1})


[ext-emergency]
;------------------------------------------------------------------
exten => 911,1,Dial(DAHDI/g0/911)



[menu-wm]
;------------------------------------------------------------------
exten => i,1,Goto(s,1)

exten => s,1,Answer()
exten => s,n,Playback(entercode)
exten => s,n,Read(PIN,,4)
exten => s,n,Gotoif($["${PIN}" = "xxxx"]?menu-wm-main,s,1:end)
exten => s,n(end),Playback(invalid)
exten => s,n,Hangup()

[menu-wm-main]
exten => i,1,Goto(s,1)

exten => s,1,Background(menu-wm-main)
exten => s,n,WaitExten(60)
exten => s,n,Hangup()

exten => 1,1,Goto(menu-wm-play,s,1)

exten => 2,1,Goto(menu-wm-rec,s,1)

exten => 3,1,Goto(menu-wm-set,s,1)

[menu-wm-play]
exten => i,1,Goto(s,1)

exten => s,1,Background(menu-wm-qplay)
exten => s,n,Background(menu-wm-list)
exten => s,n,Background(menu-back)
exten => s,n,WaitExten(30)
extem => s,n,Playback(invalid)
exten => s,n,Hangup()

exten => *,1,Goto(menu-wm-main,s,1)

exten => 0,1,Goto(s,1)

exten => _[1-4],1,Playback(beep)
exten => _[1-4],n,Playback(welcome${EXTEN})
exten => _[1-4],n,Playback(beep)
exten => _[1-4],n,Goto(s,1)

[menu-wm-rec]
exten => i,1,Goto(s,1)

exten => s,1,Background(menu-wm-qrec)
exten => s,n,Background(menu-wm-list)
exten => s,n,Background(menu-back)
exten => s,n,WaitExten(30)
exten => s,n,Playback(invalid)
exten => s,n,Hangup()

exten => *,1,Goto(menu-wm-main,s,1)

exten => 0,1,Goto(s,1)

exten => _[1-4],1,Playback(sayafterbeep)
exten => _[1-4],n,Record(/var/lib/asterisk/sounds/welcome${EXTEN}.wav,5)
exten => _[1-4],n,Wait(1)
exten => _[1-4],n,Playback(messagerecorded)
exten => _[1-4],n,Goto(s,1)

[menu-wm-set]
exten => i,1,Goto(s,1)

exten => s,1,Background(menu-wm-setlist)
exten => s,n,Background(menu-back)
exten => s,n,WaitExten(30)
exten => s,n,Hangup()

exten => *,1,Goto(menu-wm-main,s,1)

exten => _[1-4],1,Set(DB(VM/CUSTOM)=${EXTEN})
exten => _[1-4],n,SayAlpha(OK)
exten => _[1-4],n,Goto(1)

exten => 0,1,Noop(${DB_DELETE(VM/CUSTOM)})
exten => 0,n,SayAlpha(OK)
exten => 0,n,Goto(s,1)



[pbx-features]
;------------------------------------------------------------------
;Call forwarding, activate
exten => _*6X.,1,Answer()
exten => _*6X.,2,Set(DB(CFI/${CALLERID(NUM)})=${EXTEN:2})
exten => _*6X.,n,Playback(cf-activated)
exten => _*6X.,n,SayDigits(${EXTEN:2})
exten => _*6X.,n,Hangup()

;Call forwaring, deactivate
exten => *6,1,Answer()
exten => *6,n,NoOp(${DB_DELETE(CFI/${CALLERID(num)})})
exten => *6,n,Playback(cf-deactivated)
exten => *6,n,Hangup()

;Follow me, activate
exten => _*7X.,1,Answer()
exten => _*7X.,n,Set(DB(FME/${CALLERID(num)})=${EXTEN:2})
exten => _*7X.,n,Playback(fm-activated)
exten => _*7X.,n,SayDigits(${EXTEN:2})
exten => _*7X.,n,Hangup()

;Follow me, deactivate
exten => *7,1,Noop(${DB_DELETE(FME/${CALLERID(num)})})
exten => *7,n,Playback(fm-deactivated)
exten => *7,n,Hangup()

;Pickup ringing extension
exten => _*8X.,1,Pickup(${EXTEN:2})



[ext-special]
;-------------------------------------------------------------------
;VoiceMail
exten => 1200,1,VoiceMailMain(@employees)
exten => 1200,n,Hangup()
exten => VoiceMail,1,Goto(1200,1)

;Conferences
exten => _130X,1,Answer()
exten => _130X,n,Wait(1)
exten => _130X,n,MeetMe(${EXTEN},DM1)
exten => _130X,n,Hangup()

;Parking
exten => _140Z,1,ParkedCall(${EXTEN})
exten => _140Z,n,Hangup()



[ext-management]
;-------------------------------------------------------------------
;Welcome message menu
exten => 1501,1,Goto(menu-wm,s,1)

;Reboot all Snoms
exten => 1502,1,Goto(reboot-snoms,s,1)



[reboot-snoms]
;-------------------------------------------------------------------
exten => s,1,Answer()
exten => s,n,Playback(enter-PIN)
exten => s,n,Read(PIN,,4)
exten => s,n,Gotoif($["${PIN}" = "xxxx"]?ok:end)
exten => s,n(ok),Noop(PIN is OK, will reset all Snom phones now)
exten => s,n,System(${CMD_REBOOT_SNOMS})
exten => s,n,SayAlpha(OK)
exten => s,n(end),Hangup()



[blf]
;-------------------------------------------------------------------
exten => 1000,hint,SIP/1000
exten => 1006,hint,SIP/1006
exten => 1007,hint,SIP/1007
exten => 1010,hint,SIP/1010
exten => 1012,hint,SIP/1012
exten => 1013,hint,SIP/1013
exten => 1023,hint,SIP/1023


Statistics : Posted by spricer • on Thu Nov 20, 2014 12:44 pm • Replies 2 • Views 65
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Incoming Numbers

November 20, 2014, 6:17 am
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He's got insecure=invite, which means he thinks that they are not authenticating (although more likely he copies it without understanding, as with the friend). I'm not sure why he has insecure=port.

Statistics : Posted by uk26 • on Thu Nov 20, 2014 8:17 am • Replies 3 • Views 86
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Incoming Numbers

November 20, 2014, 6:17 am
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I notice you are using type=friend, Does that mean your provider can authenticate with you with a username and password combo?

If so then you can set host=dynamic and add a permit and deny block to your peer in sip.conf and let they auth.

If they cant you could define a template and use it to create your peers and save some typing.

[incomingprovider](!)
context=fromoutside
type=peer
disallow=all
allow=ulaw,alaw
dtmfmode=rfc2833
insecure=port
deny=0.0.0.0/0.0.0.0
permit=77.240.54.0/255.255.254.0

[provider-54.10](incomingprovider)
host=77.240.54.10

[provider-54.20](incomingprovider)
host=77.240.54.20

[provider-54.30](incomingprovider)
host=77.240.54.30

Statistics : Posted by uk26 • on Thu Nov 20, 2014 8:17 am • Replies 3 • Views 86
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Incoming Numbers

November 20, 2014, 6:17 am
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Combine a good firewall with allowguest=yes.

Statistics : Posted by uk26 • on Thu Nov 20, 2014 8:17 am • Replies 3 • Views 86
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Incoming Numbers

November 20, 2014, 6:17 am
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Our provider deliver inbound numbers to the pbx directly by sip url

up until recently it was simple of adding to sip.conf

[incoming-77-240-54.10]
context=fromoutside
type=friend
host=77.240.54.10
disallow=all
allow=ulaw
allow=alaw
dtmfmode=rfc2833
insecure=port,invite

however the provider has now said we need to allow from 5 different /24 IP Ranges

how am i supposed to do this. create a duplicate of the above 1280 times Image

Statistics : Posted by uk26 • on Thu Nov 20, 2014 8:17 am • Replies 3 • Views 86
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host undefined

November 20, 2014, 8:54 am
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and how do for not experid? or how config the peer for renew registration?

Statistics : Posted by javierr_vv • on Thu Nov 20, 2014 10:54 am • Replies 2 • Views 57
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host undefined

November 20, 2014, 8:54 am
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The registration expired and the peer didn't renew it.

Statistics : Posted by javierr_vv • on Thu Nov 20, 2014 10:54 am • Replies 2 • Views 57
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host undefined

November 20, 2014, 8:54 am
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------phona A-------kamailio---------asterisk-----

Code: serverjavier-desktop*CLI> sip show peers
Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description                      Realtime
JavierTren/1000           152.74.21.12                             D  Yes        Yes            5060     Unmonitored                                  Cached RT
1 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 0 offline]
serverjavier-desktop*CLI> sip show peers
Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description                      Realtime
0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 offline]



in host appear sofphone's ip, but spend time and host is undefinid, therefore asterisk dont could route
call...
plz help Image

Statistics : Posted by javierr_vv • on Thu Nov 20, 2014 10:54 am • Replies 2 • Views 57
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Issue with Remote Extensions

November 21, 2014, 11:54 am
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I just finished setting up Asterisk for home usage and am having several issues with externally registered phones. Please see the details of my configuration below:

Working

Internal

Internal to SIP Provider

External to Internal

Not Working

Internal to External (does not ring on either end)

External to SIP Provider (calls ring and connect with no audio)

sip.conf

[general]
register => XXXXXXX:XXXXXXXX@sip-provider.com
registertimeout=20
externip=mydomainname.com
localnet=192.168.11.0/255.255.255.0
context=sipprovider-inbound
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=no
subscribecontext=from-sip
directmedia=no

[2004]
type=friend
host=dynamic
dtmfmode=rfc2833
nat=yes
username=2004
secret=secret
context=phones
canreinvite=no
#directmedia=no
callerid="John Doe"

Firewall Configuration

5060 Open to Asterisk RTP Open to match range in rtp.conf

Codecs

I have confirmed that all devices are using G711 ULAW

CLI Output

== Using SIP RTP CoS mark 5
-- Executing [2004@phones:1] Dial("SIP/2000-00000000", "SIP/2004") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/2004
[Nov 20 23:18:35] WARNING[18243]: chan_sip.c:4175 retrans_pkt: Retransmission timeout reached on transmission 119d17a8483dfe824f209828115ff597@192.168.11.20:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/ ... nsmissions
Packet timed out after 32000ms with no response
-- SIP/2004-00000001 is circuit-busy
[Nov 20 23:18:35] WARNING[18243]: chan_sip.c:4204 retrans_pkt: Hanging up call 119d17a8483dfe824f209828115ff597@192.168.11.20:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/ ... nsmissions).
== Everyone is busy/congested at this time (1:0/1/0)
-- Auto fallthrough, channel 'SIP/2000-00000000' status is 'CONGESTION'

Does anyone have any idea what may be going wrong? Any help would be greatly appriciated.

Statistics : Posted by cloudserv • on Fri Nov 21, 2014 1:54 pm • Replies 1 • Views 48
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route incoming call to my time condition

November 21, 2014, 2:14 pm
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You don't need to call Hangup, but if you do, the exten=> line needs to have valid syntax. Yours is missing the extension pattern field.

Examples of how to route by time using conditional contexts are included in the sample configuration files.

Statistics : Posted by itsupport5589 • on Fri Nov 21, 2014 4:14 pm • Replies 2 • Views 47
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route incoming call to my time condition

November 21, 2014, 2:14 pm
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GotoIfTime() — Conditionally branches, depending on the time and day
Synopsis

GotoIfTime(times,days_of_week,days_of_month,months?label)

Branches to the specified extension, if the current time matches the specified time. Each of the elements may be specified either as * (for always) or as a range.

The arguments to this application are:

times

Time ranges, in 24-hour format
days_of_week

Days of the week (mon, tue, wed, thu, fri, sat, sun)
days_of_month

Days of the month (1-31)
months

Months (jan, feb, mar, apr, etc.)

; If we're open, then go to the open context
; We're open from 9am to 6pm Monday through Friday
exten => s,1,GotoIfTime(09:00-17:59,mon-fri,*,*?open,s,1)
;
; We're also late on Tuesday and Thursday
exten => s,n,GotoIfTime(09:00-19:59,tue&thru,*,*?open,s,1)
;
; We're also open from 9am to noon on Saturday
exten => s,n,GotoIfTime(09:00-11:59,sat,*,*?open,s,1)
;
; Otherwise, we're closed
exten => s,n,Goto(closed,s,1)

http://www.asteriskdocs.org/en/2nd_Edit ... -B-91.html

Statistics : Posted by itsupport5589 • on Fri Nov 21, 2014 4:14 pm • Replies 2 • Views 47
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route incoming call to my time condition

November 21, 2014, 2:14 pm
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Hi,

I m trying to configure time conditions through cli root, im unable to do from GUI.

My Currently Inbound Route set in Extension.conf
exten=>_X.,1,Dial(sip/200)
exten=>n,hangup

above is my current setting if anyone dial to my landline goes to 200 ext

how do i route incoming call to my time condition, as follows

During Office Hours 8 AM to 5 PM, ->>should goes to 200 Ext.

and After Hours Call after 5 PM to 8 AM - >> Should goes to System Recording"We are currently not available, please leave your message, we will revert back to you"

How do I configure the above from CLI in Extension.conf.

Thanks in Advance.
Sam

Statistics : Posted by itsupport5589 • on Fri Nov 21, 2014 4:14 pm • Replies 2 • Views 47
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Clipping ${EXTEN}

November 21, 2014, 2:32 pm
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${EXTEN} is really a readlonly function, that reads a field in the channel data structure. You set extensions using Goto (or Gosub).

Statistics : Posted by jreed1949 • on Fri Nov 21, 2014 4:32 pm • Replies 2 • Views 44
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Clipping ${EXTEN}

November 21, 2014, 2:32 pm
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This is Asterisk Basic.

Quote:The ${EXTEN} variable properly has the syntax ${EXTEN:x:y}, where x is the starting position and y is the number of digits to return. Given the following dial string:

94169671111

we can extract the following digit strings using the ${EXTEN:x:y} construct:

${EXTEN:1:3} would contain 416

${EXTEN:4:7} would contain 9671111

${EXTEN:-4:4} would start four digits from the end and return four digits, giving us 1111

${EXTEN:2:-4} would start two digits in and exclude the last four digits, giving us 16967

${EXTEN:-6:-4} would start six digits from the end and exclude the last four digits, giving us 67

${EXTEN:1} would give us everything after the first digit, or 4169671111 (if the number of digits to return is left blank, it will return the entire remaining string)


http://www.asteriskdocs.org/en/3rd_Edit ... -book.html

Statistics : Posted by jreed1949 • on Fri Nov 21, 2014 4:32 pm • Replies 2 • Views 44
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Clipping ${EXTEN}

November 21, 2014, 2:32 pm
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I'm sure this is simple and I should be able to figure it out but thus far I haven't. Let's say that ${EXTEN} is 95556666. I want ${EXTEN} to be 5556666 without the 9. I thought it was as easy as Set(${EXTEN}=${EXTEN:2:7}) but that does not seem to be working. So help the old guy out here and tell me what I'm doing wrong. Thank You.

Statistics : Posted by jreed1949 • on Fri Nov 21, 2014 4:32 pm • Replies 2 • Views 44
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Problem with DID with trailing zeros

November 21, 2014, 4:03 pm
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First question is I have a main number that I want to you 415-XXX-4000 I have it in my Extensions.conf and yet it rejects it when dialed from an outside line. All other numbers like 415-XXX-4001 through 4099 work with no problem at all. Hope this helps.

Extensions.conf
Code: exten => 417,1,Dial(SIP/417,20)
exten => 417,2,VoiceMail(417@VoiceMail)
exten => 417,3,Hangup()
exten => 415XXX4017,1,Dial(SIP/417,20)



This is in my SIP.conf file is this the correct format for CallerID

Code: [417]
type=friend
host=dynamic
secret=fleetline
mailbox=417@VoiceMail
callerid= "417" <415XXX4017>




This is the error message I get when I call from an outside line to 415-XXX-4000
veryone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'DAHDI/i1/4154531262-3' status is 'CHANUNAVAIL'
-- Span 1: Channel 0/1 got hangup request, cause 16
-- Hungup 'DAHDI/i1/4154531262-3'

Statistics : Posted by aristech • on Fri Nov 21, 2014 6:03 pm • Replies 2 • Views 37
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