Re: [asterisk-users] {s} - extension
On Wed, Mar 5, 2008 at 10:12 AM, Daniel Suleyman <[EMAIL PROTECTED]> wrote: > but when I use next construction(As I understand it is used to allow > to process any extension dialed by user) > > exten => s,1,Answer; > exten => s,2,Playback(hello-world,skip); > exten => s,3,Hangup; AFAIK, "s" extension is used in analogue PSTN incoming calls, as the call itself doesn't contain the extension (public telephone number) it tries to reach. If want to catch "any" extension dialed by the user you should use something like this: exten => _.,1,Answer; exten => _.,2,Playback(hello-world,skip); exten => _.,3,Hangup; In any case, I cannot understand why you would like to use it. You should define your extensions and use the invalid extension (i) to catch calls sent to any number not detailed in the dialplan. -- Andres Jimenez GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] {s} - extension
On Wed, Mar 05, 2008 at 02:12:47PM +0400, Daniel Suleyman wrote: > Dear all, I have small question > > in sip.conf I added > > [service] > type=friend > ;username= > ;secret= > qualify=900 > host=X.X.X.X > dtmfmode = rfc2833 > disallow=all > ;allow=g729 > allow=gsm > allow=alaw > allow=ulaw > > and I can proccess incoming call from soft phone only I calling on > number that is used in extensions.conf(in example below it is 1) > > exten => 1,1,Answer; > exten => 1,2,Playback(hello-world,skip); > exten => 1,3,Hangup; > > but when I use next construction(As I understand it is used to allow > to process any extension dialed by user) > > exten => s,1,Answer; > exten => s,2,Playback(hello-world,skip); > exten => s,3,Hangup; > > Asterisk says call rejected due to no extension. Chances are you got the wrong context. Have you set a context in sip.conf? Or do you use the defualt one? What is the output of: dialplan show default -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] {s} - extension
The idea is that the person connecting and dial anything he want and
the script is deciding to proceed the call or to terminate it(I think
it will be easy to manage extensions.conf - no need to create
extensions).
You know {i} doesent work
exten => 1,1,Answer;
exten => 1,2,Playback(hello-world,skip);
exten => 1,3,Hangup;
exten => i,1,Answer;
exten => i,2,Playback(welcome,skip);
exten => i,3,Hangup;
as I thought when i will dial wrong number it will play welcome message but
asterisk promt -> Call from 7007 to extension 700786 rejected because
extension not found.
to Tzafrir Cohen
in curent config i didnt set context in sip(unsing default) but it
doesent matter if I set up context the same thing s and i doesn't work
now dial plan show next
Context 'default' created by 'pbx_config' ]
'7007' => 1. Answer()
[pbx_config]
2. Playback(hello-world|skip)
[pbx_config]
3. Hangup()
[pbx_config]
'i' => 1. Answer()
[pbx_config]
2. Playback(welcome|skip)
[pbx_config]
3. Hangup()
[pbx_config]
-= 2 extensions (6 priorities) in 1 context. =-
2008/3/5, Andres Jimenez <[EMAIL PROTECTED]>:
> On Wed, Mar 5, 2008 at 10:12 AM, Daniel Suleyman <[EMAIL PROTECTED]> wrote:
>
> > but when I use next construction(As I understand it is used to allow
> > to process any extension dialed by user)
> >
> > exten => s,1,Answer;
> > exten => s,2,Playback(hello-world,skip);
> > exten => s,3,Hangup;
>
> AFAIK, "s" extension is used in analogue PSTN incoming calls, as the
> call itself doesn't contain the extension (public telephone number) it
> tries to reach.
>
> If want to catch "any" extension dialed by the user you should use
> something like this:
>
> exten => _.,1,Answer;
> exten => _.,2,Playback(hello-world,skip);
> exten => _.,3,Hangup;
>
> In any case, I cannot understand why you would like to use it.
> You should define your extensions and use the invalid extension (i) to
> catch calls sent to any number not detailed in the dialplan.
>
> --
> Andres Jimenez
>
> GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED]
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] {s} - extension
On Wed, Mar 05, 2008 at 04:04:00PM +0400, Daniel Suleyman wrote: > in curent config i didnt set context in sip(unsing default) but it > doesent matter if I set up context the same thing s and i doesn't work > > now dial plan show next > > Context 'default' created by 'pbx_config' ] > '7007' => 1. Answer() >[pbx_config] > 2. Playback(hello-world|skip) > [pbx_config] > 3. Hangup() >[pbx_config] > 'i' => 1. Answer() > [pbx_config] > 2. Playback(welcome|skip) > [pbx_config] > 3. Hangup() >[pbx_config] > > -= 2 extensions (6 priorities) in 1 context. =- So indeed you have no extension 's' in the context 'default' Can you post your extensions.conf ? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] {s} - extension
On Wed, Mar 5, 2008 at 12:04 PM, Daniel Suleyman <[EMAIL PROTECTED]> wrote:
> The idea is that the person connecting and dial anything he want and
> the script is deciding to proceed the call or to terminate it(I think
> it will be easy to manage extensions.conf - no need to create
> extensions).
It is easy. I meant you have to configure "extensions" (what to do
when a number is dialed) and handle "unknown extensions" with the "i"
extensions.
> You know {i} doesent work
> exten => i,1,Answer;
> exten => i,2,Playback(welcome,skip);
> exten => i,3,Hangup;
>
> as I thought when i will dial wrong number it will play welcome message but
> asterisk promt -> Call from 7007 to extension 700786 rejected because
> extension not found.
Have you reloaded your extensions.conf file?
in asterisk CLI "extensions reload"
--
Andres Jimenez
GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED]
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Re: [asterisk-users] {s} - extension
On Wed, Mar 05, 2008 at 12:41:23PM +, Andres Jimenez wrote: > On Wed, Mar 5, 2008 at 12:04 PM, Daniel Suleyman <[EMAIL PROTECTED]> wrote: > > The idea is that the person connecting and dial anything he want and > > the script is deciding to proceed the call or to terminate it(I think > > it will be easy to manage extensions.conf - no need to create > > extensions). > > It is easy. I meant you have to configure "extensions" (what to do > when a number is dialed) and handle "unknown extensions" with the "i" > extensions. This is not needed. If the extension is not found, there is a fallthrough to 's' (Right? Or is it chan_zap-specific)? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] {s} - extension
On Wed, Mar 5, 2008 at 1:36 PM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: > This is not needed. If the extension is not found, there is a > fallthrough to 's' (Right? Or is it chan_zap-specific)? I would say it's chan_zap-specific. From http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf "For some kinds of connections — such incoming calls from an outside telephone line — the user has not dialed an extension. In that case, Asterisk behaves as if the user had dialed a special extension named "s" (for Start). Asterisk will look for an extension "number" s in the definition of the context for that channel for instructions about what it should do to handle the call. " The key factor is that "s" is used when NO EXTENSION has been specified (when the call is not clearly directed to an specific number). As far as I know, that's the way analog lines behave. The line just receives the call, but no information says to which number the call was sent. -- Andres Jimenez GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] {s} - extension
Thank you all for answers. As I understand s - i and others is device specific. I will not need them in my SIP configuration. 2008/3/5, Andres Jimenez <[EMAIL PROTECTED]>: > On Wed, Mar 5, 2008 at 1:36 PM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: > > > This is not needed. If the extension is not found, there is a > > fallthrough to 's' (Right? Or is it chan_zap-specific)? > > I would say it's chan_zap-specific. > > From > http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf > > "For some kinds of connections — such incoming calls from an outside > telephone line — the user has not dialed an extension. In that case, > Asterisk behaves as if the user had dialed a special extension named > "s" (for Start). Asterisk will look for an extension "number" s in the > definition of the context for that channel for instructions about what > it should do to handle the call. " > > The key factor is that "s" is used when NO EXTENSION has been > specified (when the call is not clearly directed to an specific > number). As far as I know, that's the way analog lines behave. The > line just receives the call, but no information says to which number > the call was sent. > > > -- > Andres Jimenez > > GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED] > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] {s} - extension
Hi - > Thank you all for answers. As I understand s - i and others is device > specific. > I will not need them in my SIP configuration. The "s" extension is not zap-specific. You can use it for any type of device. It's just the generic extension that a call will go to when no other matching extensions are present. As Tzafrir pointed out, you had no "s" extension in the default context, and your sip device was in the default context. Therefore, you were only able to dial extensions that you had explicitly declared. To access the "s" extension from your sip device, you'd either need to add your sip device to the context where your "s" extension is, or include that context in the default context. NOTE: Andres' example using "_." will work, too (but you should make sure you put in at the end of a context if you want to put other extensions in that context as it will match all calls). - Noah > > > 2008/3/5, Andres Jimenez <[EMAIL PROTECTED]>: > > > > On Wed, Mar 5, 2008 at 1:36 PM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: > > > > > This is not needed. If the extension is not found, there is a > > > fallthrough to 's' (Right? Or is it chan_zap-specific)? > > > > I would say it's chan_zap-specific. > > > > From > http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf > > > > "For some kinds of connections — such incoming calls from an outside > > telephone line — the user has not dialed an extension. In that case, > > Asterisk behaves as if the user had dialed a special extension named > > "s" (for Start). Asterisk will look for an extension "number" s in the > > definition of the context for that channel for instructions about what > > it should do to handle the call. " > > > > The key factor is that "s" is used when NO EXTENSION has been > > specified (when the call is not clearly directed to an specific > > number). As far as I know, that's the way analog lines behave. The > > line just receives the call, but no information says to which number > > the call was sent. > > > > > > -- > > Andres Jimenez > > > > GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED] > > ___ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 's' extension Asterisk 1.2.18
Never mind the 1.2.18 messed and did not recognize the s extension any more so I just upgrade to 1.2.21.1 and fixed the problem,.weird. otis OCOSA ListAcct wrote: > how can I fix this just started .. > > Jul 14 14:32:35 NOTICE[4983]: chan_zap.c:6223 ss_thread: Got event 18 > (Ring Begin)... > == Starting Zap/1-1 at bell,s,1 failed so falling back to exten 's' > == Starting Zap/1-1 at bell,s,1 still failed so falling back to > context 'default' > Jul 14 14:32:35 WARNING[4983]: pbx.c:2377 __ast_pbx_run: Channel > 'Zap/1-1' sent into invalid extension 's' in context 'default', but no > invalid handler > -- Hungup 'Zap/1-1' > -- Starting simple switch on 'Zap/1-1' > > Otis > > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 's' extension Asterisk 1.2.18
Maybe a hardware problem? What does zttool and ztcfg -vvv say? Is Zaptel running? -- Original Message -- From: OCOSA ListAcct <[EMAIL PROTECTED]> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion Date: Sat, 14 Jul 2007 14:56:33 -0500 > >how can I fix this just started .. > >Jul 14 14:32:35 NOTICE[4983]: chan_zap.c:6223 ss_thread: Got event 18 >(Ring Begin)... > == Starting Zap/1-1 at bell,s,1 failed so falling back to exten 's' > == Starting Zap/1-1 at bell,s,1 still failed so falling back to >context 'default' >Jul 14 14:32:35 WARNING[4983]: pbx.c:2377 __ast_pbx_run: Channel >'Zap/1-1' sent into invalid extension 's' in context 'default', but no >invalid handler >-- Hungup 'Zap/1-1' >-- Starting simple switch on 'Zap/1-1' > >Otis > > > >___ >--Bandwidth and Colocation Provided by http://www.api-digital.com-- > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > Sent via the WebMail system at rockynet.com Sent via the WebMail system at rockynet.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 's' extension Asterisk 1.2.18
Yeah thats what I thought I found everything running so I just upgraded and fixed the problem. Otis Anthony Francis wrote: > Maybe a hardware problem? What does zttool and ztcfg -vvv say? Is Zaptel > running? > -- Original Message -- > From: OCOSA ListAcct <[EMAIL PROTECTED]> > Reply-To: Asterisk Users Mailing List - Non-Commercial > Discussion > Date: Sat, 14 Jul 2007 14:56:33 -0500 > > >> how can I fix this just started .. >> >> Jul 14 14:32:35 NOTICE[4983]: chan_zap.c:6223 ss_thread: Got event 18 >> (Ring Begin)... >> == Starting Zap/1-1 at bell,s,1 failed so falling back to exten 's' >> == Starting Zap/1-1 at bell,s,1 still failed so falling back to >> context 'default' >> Jul 14 14:32:35 WARNING[4983]: pbx.c:2377 __ast_pbx_run: Channel >> 'Zap/1-1' sent into invalid extension 's' in context 'default', but no >> invalid handler >>-- Hungup 'Zap/1-1' >>-- Starting simple switch on 'Zap/1-1' >> >> Otis >> >> >> >> ___ >> --Bandwidth and Colocation Provided by http://www.api-digital.com-- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> > > > > > > > Sent via the WebMail system at rockynet.com > > > > > > > > > > Sent via the WebMail system at rockynet.com > > > > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] s extension not working
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Monday, March 28, 2011 11:04 AM To: asterisk-users Subject: [asterisk-users] s extension not working Hey Guys! I have asterisk 1.8.x and somehow my 's' extension not picking up any incoming calls.. Not working [from-pstn] exten => s,1,Answer() same => n,Playback(hello-world) same => n,Hangup() Working... [from-pstn] exten => _,1,Answer() same => n,Playback(hello-world) same => n,Hangup() -S Ok Satish. I assume sip.conf or dahdi.conf has a context of from-pstn. The key to actually solving this will be for you to give us say 10 lines of CLI output. -- _ -- 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] s extension not working
If i use 's' then i got following error. This scenario is back to back
asterisk connected on PRI line (T1). for testing purpose i calling from one
asterisk to other and i want to land call on 's' extension.
shirley*CLI>
-- Extension '7527' in context 'from-pstn' from '7623' does not exist.
Rejecting call on channel 0/1, span 1
If i use _XXX then it working with following output.
shirley*CLI>
-- Accepting call from '7623' to '7527' on channel 0/1, span 1
-- Executing [7527@from-pstn:1] Answer("DAHDI/i1/7623-10", "") in new stack
-- Executing [7527@from-pstn:2] Playback("DAHDI/i1/7623-10", "hello-world")
in new stack
-- Playing 'hello-world.ulaw' (language 'en')
-- Executing [7527@from-pstn:3] Hangup("DAHDI/i1/7623-10", "") in new stack
== Spawn extension (from-pstn, 7527, 3) exited non-zero on 'DAHDI/i1/7623-10'
-- Hungup 'DAHDI/i1/7623-10'
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Mon, 28 Mar 2011 11:08:57 -0500
Subject: Re: [asterisk-users] s extension not working
From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel
Sent: Monday, March 28, 2011 11:04
AM
To: asterisk-users
Subject: [asterisk-users] s
extension not working
Hey Guys!
I have asterisk 1.8.x and somehow my 's' extension not picking up any incoming
calls..
Not working
[from-pstn]
exten => s,1,Answer()
same => n,Playback(hello-world)
same => n,Hangup()
Working...
[from-pstn]
exten => _,1,Answer()
same => n,Playback(hello-world)
same => n,Hangup()
-S
Ok Satish. I assume sip.conf or
dahdi.conf has a context of from-pstn. The key to actually solving this will
be for you to give us say 10 lines of CLI output.
--
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Re: [asterisk-users] s extension not working
Uhm
That's because you're being passed 7527 as the extension, so it won't
match "s"
On 3/28/2011 11:38 AM, satish patel wrote:
> If i use 's' then i got following error. This scenario is back to
> back asterisk connected on PRI line (T1). for testing purpose i
> calling from one asterisk to other and i want to land call on 's'
> extension.
>
> shirley*CLI>
> -- Extension '7527' in context 'from-pstn' from '7623' does not
> exist. Rejecting call on channel 0/1, span 1
>
>
>
>
> If i use _XXX then it working with following output.
>
> shirley*CLI>
> -- Accepting call from '7623' to '7527' on channel 0/1, span 1
> -- Executing [7527@from-pstn:1] Answer("DAHDI/i1/7623-10", "") in
> new stack
> -- Executing [7527@from-pstn:2] Playback("DAHDI/i1/7623-10",
> "hello-world") in new stack
> -- Playing 'hello-world.ulaw' (language 'en')
> -- Executing [7527@from-pstn:3] Hangup("DAHDI/i1/7623-10", "") in
> new stack
> == Spawn extension (from-pstn, 7527, 3) exited non-zero on
> 'DAHDI/i1/7623-10'
> -- Hungup 'DAHDI/i1/7623-10'
>
>
>
>
> From: da...@debsinc.com
> To: asterisk-users@lists.digium.com
> Date: Mon, 28 Mar 2011 11:08:57 -0500
> Subject: Re: [asterisk-users] s extension not working
>
>
>
> *From:*asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *satish
> patel
> *Sent:* Monday, March 28, 2011 11:04 AM
> *To:* asterisk-users
> *Subject:* [asterisk-users] s extension not working
>
>
>
> Hey Guys!
>
> I have asterisk 1.8.x and somehow my 's' extension not picking up any
> incoming calls..
>
> Not working
>
> [from-pstn]
> exten => s,1,Answer()
> same => n,Playback(hello-world)
> same => n,Hangup()
>
>
>
>
> Working...
>
> [from-pstn]
> exten => _,1,Answer()
> same => n,Playback(hello-world)
> same => n,Hangup()
>
>
> -S
>
>
>
> Ok Satish. I assume sip.conf or dahdi.conf has a context of
> from-pstn. The key to actually solving this will be for you to give
> us say 10 lines of CLI output.
>
>
> --
> _
> -- 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
>
>
> --
> _
> -- 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
--
Sherwood McGowan
Carrier, ITSP, Call Center, and PBX Solutions Consultant
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Re: [asterisk-users] s extension not working
@Sherwood,
I was also thinking about that But then how 's' extension match any unknown
number ? Like when call coming from PSTN then how IVR picked up...?
-Satish
> Date: Mon, 28 Mar 2011 12:58:28 -0500
> From: sherwood.mcgo...@gmail.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] s extension not working
>
> Uhm
>
> That's because you're being passed 7527 as the extension, so it won't
> match "s"
>
> On 3/28/2011 11:38 AM, satish patel wrote:
> > If i use 's' then i got following error. This scenario is back to
> > back asterisk connected on PRI line (T1). for testing purpose i
> > calling from one asterisk to other and i want to land call on 's'
> > extension.
> >
> > shirley*CLI>
> > -- Extension '7527' in context 'from-pstn' from '7623' does not
> > exist. Rejecting call on channel 0/1, span 1
> >
> >
> >
> >
> > If i use _XXX then it working with following output.
> >
> > shirley*CLI>
> > -- Accepting call from '7623' to '7527' on channel 0/1, span 1
> > -- Executing [7527@from-pstn:1] Answer("DAHDI/i1/7623-10", "") in
> > new stack
> > -- Executing [7527@from-pstn:2] Playback("DAHDI/i1/7623-10",
> > "hello-world") in new stack
> > -- Playing 'hello-world.ulaw' (language 'en')
> > -- Executing [7527@from-pstn:3] Hangup("DAHDI/i1/7623-10", "") in
> > new stack
> > == Spawn extension (from-pstn, 7527, 3) exited non-zero on
> > 'DAHDI/i1/7623-10'
> > -- Hungup 'DAHDI/i1/7623-10'
> >
> >
> >
> >
> > From: da...@debsinc.com
> > To: asterisk-users@lists.digium.com
> > Date: Mon, 28 Mar 2011 11:08:57 -0500
> > Subject: Re: [asterisk-users] s extension not working
> >
> >
> >
> > *From:*asterisk-users-boun...@lists.digium.com
> > [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *satish
> > patel
> > *Sent:* Monday, March 28, 2011 11:04 AM
> > *To:* asterisk-users
> > *Subject:* [asterisk-users] s extension not working
> >
> >
> >
> > Hey Guys!
> >
> > I have asterisk 1.8.x and somehow my 's' extension not picking up any
> > incoming calls..
> >
> > Not working
> >
> > [from-pstn]
> > exten => s,1,Answer()
> > same => n,Playback(hello-world)
> > same => n,Hangup()
> >
> >
> >
> >
> > Working...
> >
> > [from-pstn]
> > exten => _,1,Answer()
> > same => n,Playback(hello-world)
> > same => n,Hangup()
> >
> >
> > -S
> >
> >
> >
> > Ok Satish. I assume sip.conf or dahdi.conf has a context of
> > from-pstn. The key to actually solving this will be for you to give
> > us say 10 lines of CLI output.
> >
> >
> > --
> > _
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> >
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> --
> Sherwood McGowan
> Carrier, ITSP, Call Center, and PBX Solutions Consultant
>
>
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Re: [asterisk-users] s extension not working
On 3/28/2011 1:33 PM, satish patel wrote: > > @Sherwood, > > I was also thinking about that But then how 's' extension match > any unknown number ? Like when call coming from PSTN then how IVR > picked up...? > > -Satish The 's' extension does not match anything other than 's'. If your sip registrations are configured without a trailing /, it gets sent to the 's' extension on your default context. However, if you want to match *any* number, you'd want '_X.', which matches any number. -- Sherwood McGowan Carrier, ITSP, Call Center, and PBX Solutions Consultant -- _ -- 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] s extension don't work on TDM40B
To execute the "s" extension automatically when you pick up the phone, you need to put that channel in immediate mode. (I'd tell you how to do it, but I can't remember the syntax off the top of my head.) Jared Smith On Tue, 2003-06-10 at 09:57, Thomas Haeger wrote: > Hi all, > > > i have read in the * whitepaper the following: > > "s: The "start" extension. A call which does not have digits associated with > it (for > example, a loopstart analog line) begins at the "s" extension." > > I think this means the s extension will be execute when the phone is picked > up. > > But when i pick up the phone the s extension will be never executed. > > Whats wrong ? > > > Thanks for Help, > > Thomas. > > > > *** > beroNet technologies GmbH > Dipl.- Ing. Thomas Häger > Potsdamer Str. 18 A > 14513 Teltow > > FON:+49 (0) 3328 3077731 > FAX:+49 (0) 3328 334779 > Email: [EMAIL PROTECTED] > *** > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] s extension don't work on TDM40B
That's a local phone. if u what a local phone to exec 's' extensions, put immediate=yes into zapata.conf . Otherwise, you'll get a dialtone waiting for a exten input. Matteo. Il mar, 2003-06-10 alle 17:57, Thomas Haeger ha scritto: > Hi all, > > > i have read in the * whitepaper the following: > > "s: The "start" extension. A call which does not have digits associated with > it (for > example, a loopstart analog line) begins at the "s" extension." > > I think this means the s extension will be execute when the phone is picked > up. > > But when i pick up the phone the s extension will be never executed. > > Whats wrong ? > > > Thanks for Help, > > Thomas. > > > > *** > beroNet technologies GmbH > Dipl.- Ing. Thomas Häger > Potsdamer Str. 18 A > 14513 Teltow > > FON:+49 (0) 3328 3077731 > FAX:+49 (0) 3328 334779 > Email: [EMAIL PROTECTED] > *** > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] s extension don't work on TDM40B
the channel has to be in immediate mode to work as you describe with s otherwise nothing happens until you type some digits that match something in the context the phone starts in. At 05:57 PM 6/10/2003 +0200, you wrote: Hi all, i have read in the * whitepaper the following: "s: The "start" extension. A call which does not have digits associated with it (for example, a loopstart analog line) begins at the "s" extension." I think this means the s extension will be execute when the phone is picked up. But when i pick up the phone the s extension will be never executed. Whats wrong ? Thanks for Help, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] "s" extension doesn't work with ata
Hi, You have well formed your file zapata.conf? Carlos Rojas On Apr 8, 2005 9:40 PM, Drew Einhorn <[EMAIL PROTECTED]> wrote: > The ATA generates it's own dialtone, and waits for > the user to dial a number, before sending anything > to the * box. So one of the first examples in the > in the Brief Introduction to Dialplans from > Vol. 1 of the Asterisk Documentation Project. > >[incoming] >exten => s,1,Answer() >exten => s,2,Playback(goodbye) >exten => s,3,Hangup() > > does not work. The ATA generates a Dialtone > and waits for the user to dial, then as soon > as the user presses some keys. The ATA sends > that extension was not found in [incoming] > > This example is elaborated into a simple example > IVR. > > But how do we get the intial prompt to play > on an ATA? > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] "s" extension doesn't work with ata
On April 8, 2005 10:59 pm, Carlos Rojas wrote: > You have well formed your file zapata.conf? Why would an ATA use anything in zapata.conf? An ATA typically takes an analogue interface and converts it to an IAX or SIP device. I'd suggest looking at his iax.conf or sip.conf, depending on the unit. I think there'd be basic documentaion with the unit to help with this. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] "s" extension doesn't work with ata
Drew Einhorn wrote: The ATA generates it's own dialtone, and waits for the user to dial a number, before sending anything to the * box. So one of the first examples in the in the Brief Introduction to Dialplans from Vol. 1 of the Asterisk Documentation Project. [incoming] exten => s,1,Answer() exten => s,2,Playback(goodbye) exten => s,3,Hangup() does not work. The ATA generates a Dialtone and waits for the user to dial, then as soon as the user presses some keys. The ATA sends that extension was not found in [incoming] This example is elaborated into a simple example IVR. But how do we get the intial prompt to play on an ATA? In MY extensions.conf I have a comment above [incoming] that says something like "Calls without a destination number land here, usually from the PSTN". "s" is ONLY EVER called when Asterisk doesn't know what number was dialed. This (generally) only happens if a call is coming in on an ANALOG port, or if the call is coming in on a T-1/E-1 port that does not have DID/DDI service on it. An IP Phone or ATA normally send the number dialed to Asterisk and therefore if you dial 5551212 then the ATA will send the call to exten => 5551212,1,Blah( Now if your ATA is not sending the correct numbers or not waiting for you to finish dialing then the problem is with ATA and NOT Asterisk. You didn't bother to tell us what ATA you are using, so I can't really give you any more advice. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] "s" extension doesn't work with ata
On Apr 8, 2005, at 9:40 PM, Drew Einhorn wrote: ...But how do we get the intial prompt to play on an ATA? On many ATAs you can have it do a "hot-line" dial -- start a call when the phone is picked up. Perhaps you can have your ATA dial "@servername" (no phone number, just the @ sign and the server name). If you do this, this makes the phone act like a regular phone -- you pick up the phone and it is connected to the asterisk server, rather than the primary way most ATAs work -- you dial a number and it connects to the server after you are finished dialing. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
