# Unable to get outbound call going from analog phone

**URL:** <https://community.asterisk.org/t/unable-to-get-outbound-call-going-from-analog-phone/654>\
**Category:** Asterisk Support\
**Created:** [July 8, 2005, 6:04pm UTC](https://community.asterisk.org/t/unable-to-get-outbound-call-going-from-analog-phone/654 "2005-07-08T18:04:25Z")\
**Posts on this page:** 7\
**Page:** 1

<div class="post-metadata">

**Author:** ![gventer](https://avatars.discourse-cdn.com/v4/letter/g/49beb7/32.png) [@gventer](https://community.asterisk.org/u/gventer)\
**Post date:** [July 8, 2005, 6:04pm UTC](https://community.asterisk.org/t/unable-to-get-outbound-call-going-from-analog-phone/654/1 "2005-07-08T18:04:25Z")

</div>

My setup is a TE110P card - channels 1-24, a TDM40B card - channels 25-28. I have an analog phone on channel 25. I am able to receive calls on the channel and I am able to make calls to SIP phones internally but I am not able to call an outside line. I get a busy signal right after I have pressed the first digit if it is anything other than a 2 or a 3. (Internal extensions start with 2 or 3). Note : The T1 line is in and outbound dialing - the TDM40B card has 4 FXS ports for four analog devices so they still dialout over the T1 line and not an analog line.

At first I thought it was my context setup that was wrong but I dont know. Here is my zapata.conf file:

context=international  
signalling=em\_w  
immediate=no  
rxgain=1.5  
txgain=1.0  
echocancel=128  
echocancelwhenbridged=yes  
echotraining=yes  
callerid=asreceived  
group=1  
callgroup=1  
pickupgroup=1  
channel =\> 1-24

context=international  
signalling=fxo\_ks  
immediate=no  
rxgain=1.5  
txgain=1.0  
echocancel=128  
echocancelwhenbridged=yes  
echotraining=yes  
callerid=asreceived  
roup=2  
channel =\> 25-28

The international context (and the long distance and local) is the one that comes with the sample dialplan when you install Asterisk.

Anybody has any idea?

Also it seems that Asterisk dont detect when I hangup on the analog phone.

Thanks  
Gerhard.

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<div class="post-metadata">

**Author:** ![gventer](https://avatars.discourse-cdn.com/v4/letter/g/49beb7/32.png) [@gventer](https://community.asterisk.org/u/gventer)\
**Post date:** [July 8, 2005, 6:17pm UTC](https://community.asterisk.org/t/unable-to-get-outbound-call-going-from-analog-phone/654/2 "2005-07-08T18:17:22Z")

</div>

Also do I need to modprobe again after I made changes to zapata.conf or can I simply do a reload from the CLI prompt?

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<div class="post-metadata">

**Author:** ![platinumrik](https://avatars.discourse-cdn.com/v4/letter/p/aeb1de/32.png) [@platinumrik](https://community.asterisk.org/u/platinumrik)\
**Post date:** [July 8, 2005, 6:56pm UTC](https://community.asterisk.org/t/unable-to-get-outbound-call-going-from-analog-phone/654/3 "2005-07-08T18:56:43Z")

</div>

can you post your dialplan? If you do a reload it should reload all of your config files.

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<div class="post-metadata">

**Author:** ![gventer](https://avatars.discourse-cdn.com/v4/letter/g/49beb7/32.png) [@gventer](https://community.asterisk.org/u/gventer)\
**Post date:** [July 8, 2005, 7:16pm UTC](https://community.asterisk.org/t/unable-to-get-outbound-call-going-from-analog-phone/654/4 "2005-07-08T19:16:02Z")

</div>

Sure.

[general]

static=yes

writeprotect=no

; Global variables used in the application

[globals]

TRUNK=Zap/g1 ; Trunk interface  
TRUNK2=Zap/g2 ; 2nd Trunk interface

TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)  
Reception = 7000

; Extension variables  
Ext\_01 = 3640   
Ext\_02 = 3661   
Ext\_03 = 3627   
Ext\_04 = 3624

Ext\_05 = 3610   
Ext\_06 = 3612   
Ext\_07 = 3645   
Ext\_08 = 3629

Ext\_09 = 3647

Ext\_10 = 3616   
Ext\_11 = 3625   
Ext\_12 = 3611   
Ext\_80 = 7000   
Ext\_81 = 3602   
Ext\_82 = 4001   
Ext\_83 = 4002

Ext\_20 = 3654   
Ext\_21 = 3655

[trunkint]

;

; International long distance through trunk

;

exten =\> \_9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})

exten =\> \_9011.,2,Congestion

[trunkld]

;

; Long distance context accessed through trunk

;

exten =\> \_91NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})  
exten =\> \_91NXXNXXXXXX,2,Congestion

;exten =\> \_1NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})  
;exten =\> \_1NXXNXXXXXX,2,Congestion

[trunklocal]

;

; Local seven-digit dialing accessed through trunk interface

;

exten =\> \_9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})  
exten =\> \_9NXXXXXX,2,Congestion

[trunktollfree]

;

; Long distance context accessed through trunk interface

;

exten =\> \_91800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})

exten =\> \_91800NXXXXXX,2,Congestion

exten =\> \_91888NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})

exten =\> \_91888NXXXXXX,2,Congestion

exten =\> \_91877NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})

exten =\> \_91877NXXXXXX,2,Congestion

exten =\> \_91866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})

exten =\> \_91866NXXXXXX,2,Congestion

[international]

;

; Master context for international long distance

;

ignorepat =\> 9

include =\> longdistance

include =\> trunkint

[longdistance]

;

; Master context for long distance

;

ignorepat =\> 9

include =\> local

include =\> trunkld

[local]

;

; Master context for local, toll-free, and iaxtel calls only

;

ignorepat =\> 9

include =\> default

include =\> parkedcalls

include =\> trunklocal

;include =\> iaxtel700

include =\> trunktollfree

;include =\> iaxprovider

;

; You can use an alternative switch type as well, to resolve

; extensions that are not known here, for example with remote

; IAX switching you transparently get access to the remote

; Asterisk PBX

;

; switch =\> IAX2/user:password@bigserver/local

[macro-stdexten];

;

; Standard extension macro:

; ${ARG1} - Extension (we could have used ${MACRO\_EXTEN} here as well

; ${ARG2} - Device(s) to ring

;  
exten =\> s,1,Answer  
exten =\> s,2,Ringing  
exten =\> s,3,Wait(2)

exten =\> s,4,Dial(${ARG2},20,tTr) ; Ring the interface, 20 seconds maximum

exten =\> s,5,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten =\> s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable, send to voicemail w/ unavail announce

exten =\> s-NOANSWER,2,Goto(default,s,1) ; If they press #, return to start

exten =\> s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send to voicemail w/ busy announce

exten =\> s-BUSY,2,Goto(default,s,1) ; If they press #, return to start

exten =\> \_s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer

exten =\> a,1,VoicemailMain(${ARG1}) ; If they press \*, send the user into VoicemailMain

[incoming]

;

; We start with what to do when a call first comes in.

; All standard extensions  
exten =\> ${Ext\_01},1,Macro(stdexten,${Ext\_01},SIP/Ext\_01)  
exten =\> ${Ext\_02},1,Macro(stdexten,${Ext\_02},SIP/Ext\_02)  
exten =\> ${Ext\_03},1,Macro(stdexten,${Ext\_03},SIP/Ext\_03)  
exten =\> ${Ext\_04},1,Macro(stdexten,${Ext\_04},SIP/Ext\_04)  
exten =\> ${Ext\_05},1,Macro(stdexten,${Ext\_05},SIP/Ext\_05)  
exten =\> ${Ext\_06},1,Macro(stdexten,${Ext\_06},SIP/Ext\_06)  
exten =\> ${Ext\_07},1,Macro(stdexten,${Ext\_07},SIP/Ext\_07)  
exten =\> ${Ext\_08},1,Macro(stdexten,${Ext\_08},SIP/Ext\_08)  
exten =\> ${Ext\_09},1,Macro(stdexten,${Ext\_09},SIP/Ext\_09)  
exten =\> ${Ext\_10},1,Macro(stdexten,${Ext\_10},SIP/Ext\_10)  
exten =\> ${Ext\_11},1,Macro(stdexten,${Ext\_11},SIP/Ext\_11)  
exten =\> ${Ext\_20},1,Macro(stdexten,${Ext\_20},SIP/Ext\_20)  
exten =\> ${Ext\_21},1,Macro(stdexten,${Ext\_21},SIP/Ext\_21)

;Reception,Sales and Support  
exten =\> ${Ext\_80},1,Goto(reception,s,1)  
exten =\> ${Ext\_81},1,Goto(support-direct,s,1)

;Conference rooms  
exten =\> ${Ext\_82},1,Macro(stdexten,${Ext\_82},SIP/Ext\_82)  
exten =\> ${Ext\_83},1,Macro(stdexten,${Ext\_83},SIP/Ext\_83)

;Fax  
exten =\> 2383,1,Dial(Zap/25,20)  
exten =\> 3695,1,Dial(Zap/26,20)  
exten =\> t,1,Hangup

[reception]  
exten =\> s,1,Answer  
exten =\> s,2,Ringing  
exten =\> s,3,Wait(2)

exten =\> s,4,Dial(SIP/Ext\_05,8,tTr)

exten =\> s,5,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

;temp testing

;exten =\> s,4,Goto(mainmenu,s,1)

exten =\> s-NOANSWER,1,Goto(mainmenu,s,1) ; If unavailable ring the sales group

exten =\> s-BUSY,1,Goto(mainmenu,s,1)  
exten =\> t,1,Goto(mainmenu,s,1) ;time out

[sales]  
exten =\> s,1,Answer  
exten =\> s,2,Background(custom/thank-you)  
exten =\> s,3,Background(custom/one-moment-plz)  
exten =\> s,4,Ringing  
exten =\> s,5,Wait(2)  
exten =\> s,6,Dial(SIP/Ext\_10&SIP/Ext\_06,20,tTr)   
exten =\> s,102,Voicemail(u${Ext\_80})   
exten =\> t,1,Voicemail(u7000)

[support]

exten =\> s,1,Answer  
exten =\> s,2,Background(custom/thank-you)  
exten =\> s,3,Background(custom/one-moment-plz)  
exten =\> s,4,Ringing  
exten =\> s,5,Wait(2)  
exten =\> s,6,Dial(SIP/Ext\_02&SIP/Ext\_08&SIP/Ext\_07&SIP/Ext\_11,20,tTr)   
exten =\> s,102,Voicemail(u${Ext\_81}) ;put them into voicemail  
exten =\> t,1,Voicemail(u3602)

[support-direct]

exten =\> s,1,Answer  
exten =\> s,2,Wait(1)  
exten =\> s,3,Background(custom/dm-support-1st-announcement)  
exten =\> s,4,Ringing  
exten =\> s,5,Wait(2)  
exten =\> s,6,Dial(SIP/Ext\_02&SIP/Ext\_08&SIP/Ext\_07&SIP/Ext\_11,20,tTr)   
exten =\> s,102,Voicemail(u${Ext\_81}) ;put them into voicemail  
exten =\> t,1,Voicemail(u3602) ;timeout into voicemail

[mainmenu]  
include =\> incoming  
exten =\> s,1,Answer  
exten =\> s,2,Ringing  
exten =\> s,3,Wait(1sss)  
exten =\> s,4,Background(custom/main-menu) ;for sales press 1 for support press 2  
exten =\> 1,1,Goto(sales,s,1)  
exten =\> 2,1,Goto(support,s,1)  
;exten =\> 3,1,Directory(default)  
;exten =\> 2,5,Wait(3)

;exten =\> 2,6,Goto(s,4) ;no response repeat options again

exten =\> t,1,Hangup

[default]

include =\> incoming  
include =\> parkedcalls

exten =\> 8000,1,Answer  
exten =\> 8000,2,DigitTimeout(2)

exten =\> 8000,3,ResponseTimeout(10)

exten =\> 8000,4,Goto(mainmenu,s,1)

exten =\> 7000,1,Answer  
exten =\> 7000,2,DigitTimeout(2)

exten =\> 7000,3,ResponseTimeout(10)  
exten =\> 7000,4,Goto(reception,s,1)

; recording of phrases

exten =\> 205,1,Goto(phrase-menu,s,1)

exten =\> 206,1,Goto(recording,s,1)

; conference rooms (you’ll need to edit meetme.conf to enable this room)

exten =\> 8601,1,Meetme(8601)  
exten =\> 8602,1,Meetme(8602)

; Intercom

exten =\> 6000,1,Dial,console/dsp

; Monitor an extension

exten =\> 8159,1,ZapBarge

exten =\> 8159,2,Hangup

; Call Pickup

exten =\> \*8,1,Pickup

; Give voicemail at extension 9999  
exten =\> 3900,1,Answer  
exten =\> 3900,2,Wait(1)

exten =\> 3900,3,VoicemailMain

exten =\> 3900,4,Hangup

; #=hangup

exten =\> #,1,Background(custom/goodbye)

exten =\> #,2,Hangup

; hangup if caller does nothing

exten =\> t,1,Hangup

; Invaild

exten =\> i,1,Playback(invalid)

; Repeats Previous Menu

exten =\> \*,1,Goto(s,1)

[phrase-menu]

exten =\> s,1,Answer ; Answer the line  
exten =\> s,2,Wait(1)

exten =\> s,3,DigitTimeout,3 ; Set Digit Timeout to 5 seconds

exten =\> s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds

exten =\> s,5,BackGround(custom/phrase-menu) ; Play main menu.

; Phrase Recording

exten =\> 1,1,Wait(1)

exten =\> 1,2,Read(PHRASEID|custom/enter-phrase-num)

exten =\> 1,3,Wait(2) ; give yourself 2 secs to take a breath and wait for beep

exten =\> 1,4,Record(custom/${PHRASEID}:gsm)

exten =\> 1,5,Wait(2)

exten =\> 1,6,Playback(custom/${PHRASEID})

exten =\> 1,7,Wait(1)

exten =\> 1,8,Goto(s,2)

; Phrase review

exten =\> 2,1,Wait(1)

exten =\> 2,2,Read(PHRASEID|custom/enter-phrase-num)

exten =\> 2,3,Wait(1)

exten =\> 2,4,Playback(custom/${PHRASEID})

exten =\> 2,5,Wait(1)

exten =\> 2,6,Goto(s,2)

; #=hangup

exten =\> #,1,Background(custom/goodbye)

exten =\> #,2,Hangup

exten =\> t,1,Hangup

exten =\> i,1,Playback(invalid)

exten =\> i,2,Goto(s,2)

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<div class="post-metadata">

**Author:** ![platinumrik](https://avatars.discourse-cdn.com/v4/letter/p/aeb1de/32.png) [@platinumrik](https://community.asterisk.org/u/platinumrik)\
**Post date:** [July 8, 2005, 7:28pm UTC](https://community.asterisk.org/t/unable-to-get-outbound-call-going-from-analog-phone/654/5 "2005-07-08T19:28:03Z")

</div>

you include iaxprovider but i don’t see it in your dial plan at all? i think you might be missing some of your context’s

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<div class="post-metadata">

**Author:** ![gventer](https://avatars.discourse-cdn.com/v4/letter/g/49beb7/32.png) [@gventer](https://community.asterisk.org/u/gventer)\
**Post date:** [July 9, 2005, 5:09pm UTC](https://community.asterisk.org/t/unable-to-get-outbound-call-going-from-analog-phone/654/6 "2005-07-09T17:09:57Z")

</div>

No, the iaxprovider is commented out.

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<div class="post-metadata">

**Author:** ![platinumrik](https://avatars.discourse-cdn.com/v4/letter/p/aeb1de/32.png) [@platinumrik](https://community.asterisk.org/u/platinumrik)\
**Post date:** [July 9, 2005, 6:08pm UTC](https://community.asterisk.org/t/unable-to-get-outbound-call-going-from-analog-phone/654/7 "2005-07-09T18:08:16Z")

</div>

yeah , sorry bout that. do your 1800 numbers work? try dialing from the console see what that yelds