[asterisk-users] How can get user inputs from called party after dial?
Hi, I want to dial a party, play him a message and wait for his input, i.e. DTMF digits and use them to control the rest of the dial plan. How do I do it? If I use Dial it will not return until the end of the call, isn't it? Thanks, Eyal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How can get user inputs from called party after dial?
Thanks, but I'm missing something here, the dial command is where?
I need to do something like:
Dial(1234)
Read(1 digit)
DoSomthing(based on digit from 1234)
And as far as I understand the Dial start the call and only come back (ig
you use the g option) after call finished.
Eyal
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce bruce
Sent: Saturday, July 10, 2010 9:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How can get user inputs from called party
after dial?
You need read():
http://www.voip-info.org/wiki/view/Asterisk+cmd+Read
It's as easy as:
exten = s,n,Read(variable,,11)
exten = s,n,NoOp(${variable})
Above will take up to 11 digits input by user and will display it back in
NoOP on Asterisk CLI.
-Bruce
On Sat, Jul 10, 2010 at 2:16 PM, eyal goltzman egoltz...@gmail.com wrote:
Hi,
I want to dial a party, play him a message and wait for his input, i.e. DTMF
digits and use them to control the rest of the dial plan.
How do I do it?
If I use Dial it will not return until the end of the call, isn't it?
Thanks,
Eyal
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Re: [asterisk-users] How can get user inputs from called party after dial?
Thank you Bruce,
I think we are not on the same page.
I have an AMI script that issue an originate command, after one channel is
connected I'm in my dialplan at extensions_custom.conf (I use FreePBX).
Now I'm issuing a Dial command to the another party that when he pick up the
phone I play for him a message (using the A option in the Dial command) and
then want to wait for his input, this is the case.
Eyal
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce bruce
Sent: Saturday, July 10, 2010 9:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How can get user inputs from called party
after dial?
You need to do some reading :-)
I will give you a quick teach here. At the end of file
/etc/asterisk/extensions_custom.conf (if you are running FreePBX) OR in
/etc/asterisk/extensions.conf (if you are running vanilla Asterisk) add
this:
[first-Dialplan]
exten = s,1,Answer
exten = s,n,Playback(Welcome)
exten = s,n,Read(numb,,10)
exten = s,n,NoOp(${numb})
And send your inbound route to context first-Dialplan so that it's triggered
when a call comes in. Then on terminal do a asterisk -r and you
will see the NoOp show the DTMF number entered. From there on you can do
anything you want with the variable ${numb}
If any part of above is unclear to you, you must consult your friend,
google, for examples of Asterisk dialplan.
-Bruce
On Sat, Jul 10, 2010 at 2:38 PM, Eyal Goltzman egoltz...@gmail.com wrote:
Thanks, but I'm missing something here, the dial command is where?
I need to do something like:
Dial(1234)
Read(1 digit)
DoSomthing(based on digit from 1234)
And as far as I understand the Dial start the call and only come back (ig
you use the g option) after call finished.
Eyal
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce bruce
Sent: Saturday, July 10, 2010 9:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How can get user inputs from called party
after dial?
You need read():
http://www.voip-info.org/wiki/view/Asterisk+cmd+Read
It's as easy as:
exten = s,n,Read(variable,,11)
exten = s,n,NoOp(${variable})
Above will take up to 11 digits input by user and will display it back in
NoOP on Asterisk CLI.
-Bruce
On Sat, Jul 10, 2010 at 2:16 PM, eyal goltzman egoltz...@gmail.com wrote:
Hi,
I want to dial a party, play him a message and wait for his input, i.e. DTMF
digits and use them to control the rest of the dial plan.
How do I do it?
If I use Dial it will not return until the end of the call, isn't it?
Thanks,
Eyal
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No virus found in this incoming message.
Checked by AVG - www.avg.com
Version: 9.0.830 / Virus Database: 271.1.1/2991 - Release Date: 07/10/10
09:36:00
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09:36:00
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Re: [asterisk-users] How can get user inputs from called party after dial?
Thank you Bruce, In the below example you sent the dialplan will stop after
Dial.
I found the solution to my problem in the M option of the Dial command that
let you run a macro BEFORE the parties are connected and continue the
dialplan based on the MACRO_RESULT.
Thanks for your help,
Eyal
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce bruce
Sent: Saturday, July 10, 2010 9:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How can get user inputs from called party
after dial?
For dial you do this:
[first-Dialplan]
exten = s,1,Answer
exten = s,n,Dial(SIP/provider/111222)
exten = s,n,Playback(Welcome)
exten = s,n,Read(numb,,10)
exten = s,n,NoOp(${numb})
-Bruce
On Sat, Jul 10, 2010 at 2:51 PM, bruce bruce bruceb...@gmail.com wrote:
You need to do some reading :-)
I will give you a quick teach here. At the end of file
/etc/asterisk/extensions_custom.conf (if you are running FreePBX) OR in
/etc/asterisk/extensions.conf (if you are running vanilla Asterisk) add
this:
[first-Dialplan]
exten = s,1,Answer
exten = s,n,Playback(Welcome)
exten = s,n,Read(numb,,10)
exten = s,n,NoOp(${numb})
And send your inbound route to context first-Dialplan so that it's triggered
when a call comes in. Then on terminal do a asterisk -r and you
will see the NoOp show the DTMF number entered. From there on you can do
anything you want with the variable ${numb}
If any part of above is unclear to you, you must consult your friend,
google, for examples of Asterisk dialplan.
-Bruce
On Sat, Jul 10, 2010 at 2:38 PM, Eyal Goltzman egoltz...@gmail.com wrote:
Thanks, but I'm missing something here, the dial command is where?
I need to do something like:
Dial(1234)
Read(1 digit)
DoSomthing(based on digit from 1234)
And as far as I understand the Dial start the call and only come back (ig
you use the g option) after call finished.
Eyal
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce bruce
Sent: Saturday, July 10, 2010 9:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How can get user inputs from called party
after dial?
You need read():
http://www.voip-info.org/wiki/view/Asterisk+cmd+Read
It's as easy as:
exten = s,n,Read(variable,,11)
exten = s,n,NoOp(${variable})
Above will take up to 11 digits input by user and will display it back in
NoOP on Asterisk CLI.
-Bruce
On Sat, Jul 10, 2010 at 2:16 PM, eyal goltzman egoltz...@gmail.com wrote:
Hi,
I want to dial a party, play him a message and wait for his input, i.e. DTMF
digits and use them to control the rest of the dial plan.
How do I do it?
If I use Dial it will not return until the end of the call, isn't it?
Thanks,
Eyal
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No virus found in this incoming message.
Checked by AVG - www.avg.com
Version: 9.0.830 / Virus Database: 271.1.1/2991 - Release Date: 07/10/10
09:36:00
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No virus found in this incoming message.
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Version: 9.0.830 / Virus Database: 271.1.1/2991 - Release Date: 07/10/10
09:36:00
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[asterisk-users] SIP Trunk configuration problem - fromdomain
Hello, I'm trying to register to my provider sip trunk, I got from him an host IP (a.b.c.d) to connect to and my provider recognize me based on the fixed IP (x.y.z.w) he gave me (no need for username and password) In the sip.conf I add: [mytrunk] type=friend insecure=no host=a.b.c.d fromdomain=x.y.z.w qualify=3600 nat=no ; change to yes if you are behind NAT bindport=5060 bindaddr=0.0.0.0 context=default disallow=all allow=ulaw allow=alaw Now, my asterisk resides in my internal network (10.100.101.107) and in the SIP requests that sent to the provider I can see (via a sniffer) that the From and Contact fields have - sip:aster...@10.100.101.107 and not the x.y.z.w I expected to see as a result of the fromdomain=x.y.z.w. Any idea? Thanks, Eyal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to change the IP in the SIP contact header
Hello, I'm trying to use a SIP trunk service and the provider ask me to have the IP address of the contact header as my public IP and not as my private one, how can I do it? See attached the SIP invite where a.b.c.d is the SIP server IP and x.y.z.w is my public address: sipINVITE sip:144@ a.b.c.d SIP/2.0 Via: SIP/2.0/UDP 10.100.101.107:5060;branch=z9hG4bK76d52819;rport Max-Forwards: 70 From: Polycom sip:100@ x.y.z.w;tag=as7435100b To: sip:144@ a.b.c.d Contact: sip:1...@10.100.101.107 Call-ID: 08116cf06661dc091de10c1b3315d...@84.94.96.110 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Mon, 05 Jul 2010 15:49:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 292 v=0 o=root 1812163927 1812163927 IN IP4 10.100.101.107 s=Asterisk PBX 1.6.1.20 c=IN IP4 10.100.101.107 t=0 0 m=audio 18848 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to change the IP in the SIP contact header
Yes, I tried and it did not solve the problem, Thanks From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jamie A. Stapleton Sent: Monday, July 05, 2010 9:05 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] How to change the IP in the SIP contact header Have you tried setting externip= In the [general] of your sip.conf? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eyal Goltzman Sent: Monday, July 05, 2010 1:58 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] How to change the IP in the SIP contact header Hello, I'm trying to use a SIP trunk service and the provider ask me to have the IP address of the contact header as my public IP and not as my private one, how can I do it? See attached the SIP invite where a.b.c.d is the SIP server IP and x.y.z.w is my public address: sipINVITE sip:144@ a.b.c.d SIP/2.0 Via: SIP/2.0/UDP 10.100.101.107:5060;branch=z9hG4bK76d52819;rport Max-Forwards: 70 From: Polycom sip:100@ x.y.z.w;tag=as7435100b To: sip:144@ a.b.c.d Contact: sip:1...@10.100.101.107 Call-ID: 08116cf06661dc091de10c1b3315d...@84.94.96.110 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Mon, 05 Jul 2010 15:49:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 292 v=0 o=root 1812163927 1812163927 IN IP4 10.100.101.107 s=Asterisk PBX 1.6.1.20 c=IN IP4 10.100.101.107 t=0 0 m=audio 18848 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv No virus found in this incoming message. Checked by AVG - www.avg.com Version: 9.0.830 / Virus Database: 271.1.1/2978 - Release Date: 07/05/10 09:36:00 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using AMI Originate to call 2 outside numbers and connect them
Thanks Tzafrir, If I use the other channel as the extension parameter it will do the work for me also? Eyal -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Saturday, July 03, 2010 2:15 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Using AMI Originate to call 2 outside numbers and connect them On Sat, Jul 03, 2010 at 01:33:25AM +0300, eyal goltzman wrote: Hello, Can I use AMI Originate to call 2 outside numbers (SIP) and connect them? How? Originate one channel to the application Dial to dial to the other channel? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG - www.avg.com Version: 9.0.830 / Virus Database: 271.1.1/2978 - Release Date: 07/02/10 21:35:00 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using AMI Originate to call 2 outside numbers and connect them
Hello, Can I use AMI Originate to call 2 outside numbers (SIP) and connect them? How? Thanks Eyal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] unwanted entries created in dialplan from users.conf, how can I get rid of it?
Hello,
When I call dialplan reload I can see the following lines:
== Parsing '/etc/asterisk/extensions.conf': == Found
-- Registered extension context 'default' (0x8a72410) in local table
0x8a679d0; registrar: pbx_config
-- Added extension '_1XX' priority 1 to default (0x8a72410)
.
.
.
== Parsing '/etc/asterisk/users.conf': == Found
-- Added extension '100' priority -1 to default (0xb731b4e8)
-- Added extension '100' priority 1 to default (0xb731b4e8)
-- Added extension '101' priority -1 to default (0xb731b4e8)
-- Added extension '101' priority 1 to default (0xb731b4e8)
That result in a dialplan that look like that:
[ Context 'default' created by 'pbx_config' ]
'100' = hint: SIP/100IAX2/100 [pbx_config]
1. Dial(${HINT})[pbx_config]
'_1XX' = 1. Playback(digits/4) [pbx_config]
I don't want those 2 first line to be there only the _1XX. How do I get rid
of it? Why it is added to the dial plan automatically?
extention.conf look like this:
[default]
exten = _1XX,1,Playback(digits/4)
users.cong look like this:
[100]
username = 100
transfer = yes
mailbox = 100
call-limit = 100
type = peer
fullname = Polycom
registersip = no
host = dynamic
callgroup = 1
type = peer
context = default
cid_number = 100
hasvoicemail = no
threewaycalling = no
hasdirectory = no
callwaiting = no
hasmanager = no
hasagent = no
hassip = yes
hasiax = no
secret = 100
nat = yes
canreinvite = no
dtmfmode = rfc2833
insecure = no
pickupgroup = 1
autoprov = no
macaddress =
linenumber = 1
LINEKEYS = 1
disallow = all
allow = ulaw,gsm
Thanks,
Eyal
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[asterisk-users] Where can I find a minimal set of empty configuration files (SIP only)?
Hello, After installing and learning Asterisk I found myself with a need for a minimal set of empty configuration files with only the must have stuff in order to setup a SIP only machine, is there a place to find it? Thanks, Eyal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is there a default dial plan that is not in extention.conf?
Hi, I have a trivial peace of dialplan for exten 100. I try to change it to _1XX and the asterisk act according to a different (Default??) dial plan and not the one I want? Is that possible? Where is the other dialplan sits? In my extention.conf I can't see something that look like what asterisk is dialing. How can I trace\debug my dialplan? Thanks, Eyal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a default dial plan that is not in extention.conf?
Thank you all,
This is what I see after CLI dialplan show 1...@default :
'100' = hint: SIP/100IAX2/100
[pbx_config]
1. Dial(${HINT})
[pbx_config]
'_1XX' = 1. Playback(digits/4)
[pbx_config]
From where come the 2 first lines?? I only have the third one as the only
one under my default context at extention.conf.
And what is [pbx_config]?
Thanks
Eyal
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
Sent: Friday, June 25, 2010 4:05 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Is there a default dial plan that is not in
extention.conf?
On Fri, Jun 25, 2010 at 02:25:38PM +0300, Eyal Goltzman wrote:
Hi,
I have a trivial peace of dialplan for exten 100. I try to change it to
_1XX
and the asterisk act according to a different (Default??) dial plan and
not
the one I want? Is that possible? Where is the other dialplan sits? In my
extention.conf I can't see something that look like what asterisk is
dialing.
How can I trace\debug my dialplan?
To see where it comes from, run in the Asterisk CLI:
dialplan show context
or:
dialplan show exten@context
Here is a partial output from 'dialplan show' here, that shows all of
them (but is normally overly long)
[ Context 'app_queue_gosub_virtual_context' created by 'app_queue' ]
's' =1. NoOp() [app_queue]
[ Context 'parkedcalls' created by 'features' ]
'700' = 1. Park() [features]
[ Context 'app_dial_gosub_virtual_context' created by 'app_dial' ]
's' =1. NoOp() [app_dial]
[ Context 'from-pstn' created by 'pbx_config' ]
'_X.' = 1. Answer() [pbx_config]
2. Playback(demo-instruct)[pbx_config]
3. Hangup() [pbx_config]
[ Context 'ael-dundi-e164' created by 'pbx_ael' ]
's' =1. MSet(LOCAL(exten)=${ARG1}) [pbx_ael]
2. Goto(${exten},1) [pbx_ael]
3. Return() [pbx_ael]
'pbx_config' is dialplan that was generated from your extensions.conf.
'pbx_ael' is dialplan that was generated from extensions.ael.
Various other modules include their own minor dialplan snippets.
'dialplan show exten@context' also resolves various 'include='
directives.
If you had:
[local]
include = phones
exten = 120,1,Dial(SIP/trunk/123456)
[phones]
exten = 100,1,Dial(SIP/phone1)
the 'dialplan show local' would show the equivalent of
include = phones
exten = 120,1,Dial(SIP/trunk/123456)
whereas 'dialplan show 1...@local would show the actual (equivalent of)
exten = 100,1,Dial(SIP/phone1)
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
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No virus found in this incoming message.
Checked by AVG - www.avg.com
Version: 9.0.830 / Virus Database: 271.1.1/2961 - Release Date: 06/24/10
21:35:00
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New to Asterisk? Join us for a live introductory webinar every Thurs:
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