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<br>@Sherwood, <br><br>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...?<br><br>-Satish <br><br>> Date: Mon, 28 Mar 2011 12:58:28 -0500<br>> From: sherwood.mcgowan@gmail.com<br>> To: asterisk-users@lists.digium.com<br>> Subject: Re: [asterisk-users] s extension not working<br>> <br>> Uhm....<br>> <br>> That's because you're being passed 7527 as the extension, so it won't<br>> match "s"<br>> <br>> On 3/28/2011 11:38 AM, satish patel wrote:<br>> > If i use 's' then i got following error. This scenario is back to<br>> > back asterisk connected on PRI line (T1). for testing purpose i<br>> > calling from one asterisk to other and i want to land call on 's'<br>> > extension.<br>> ><br>> > shirley*CLI><br>> > -- Extension '7527' in context 'from-pstn' from '7623' does not<br>> > exist. Rejecting call on channel 0/1, span 1<br>> ><br>> ><br>> ><br>> ><br>> > If i use _XXX then it working with following output.<br>> ><br>> > shirley*CLI><br>> > -- Accepting call from '7623' to '7527' on channel 0/1, span 1<br>> > -- Executing [7527@from-pstn:1] Answer("DAHDI/i1/7623-10", "") in<br>> > new stack<br>> > -- Executing [7527@from-pstn:2] Playback("DAHDI/i1/7623-10",<br>> > "hello-world") in new stack<br>> > -- <DAHDI/i1/7623-10> Playing 'hello-world.ulaw' (language 'en')<br>> > -- Executing [7527@from-pstn:3] Hangup("DAHDI/i1/7623-10", "") in<br>> > new stack<br>> > == Spawn extension (from-pstn, 7527, 3) exited non-zero on<br>> > 'DAHDI/i1/7623-10'<br>> > -- Hungup 'DAHDI/i1/7623-10'<br>> ><br>> ><br>> ><br>> > ------------------------------------------------------------------------<br>> > From: danny@debsinc.com<br>> > To: asterisk-users@lists.digium.com<br>> > Date: Mon, 28 Mar 2011 11:08:57 -0500<br>> > Subject: Re: [asterisk-users] s extension not working<br>> ><br>> > ------------------------------------------------------------------------<br>> ><br>> > *From:*asterisk-users-bounces@lists.digium.com<br>> > [mailto:asterisk-users-bounces@lists.digium.com] *On Behalf Of *satish<br>> > patel<br>> > *Sent:* Monday, March 28, 2011 11:04 AM<br>> > *To:* asterisk-users<br>> > *Subject:* [asterisk-users] s extension not working<br>> ><br>> > <br>> ><br>> > Hey Guys!<br>> ><br>> > I have asterisk 1.8.x and somehow my 's' extension not picking up any<br>> > incoming calls..<br>> ><br>> > Not working<br>> ><br>> > [from-pstn]<br>> > exten => s,1,Answer()<br>> > same => n,Playback(hello-world)<br>> > same => n,Hangup()<br>> ><br>> ><br>> ><br>> ><br>> > Working...<br>> ><br>> > [from-pstn]<br>> > exten => _XXXX,1,Answer()<br>> > same => n,Playback(hello-world)<br>> > same => n,Hangup()<br>> ><br>> ><br>> > -S<br>> ><br>> > <br>> ><br>> > Ok Satish. I assume sip.conf or dahdi.conf has a context of<br>> > from-pstn. The key to actually solving this will be for you to give<br>> > us say 10 lines of CLI output.<br>> ><br>> ><br>> > --<br>> > _____________________________________________________________________<br>> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>> > New to Asterisk? Join us for a live introductory webinar every Thurs:<br>> > http://www.asterisk.org/hello asterisk-users mailing list To<br>> > UNSUBSCRIBE or update options visit:<br>> > http://lists.digium.com/mailman/listinfo/asterisk-users<br>> ><br>> ><br>> > --<br>> > _____________________________________________________________________<br>> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>> > New to Asterisk? Join us for a live introductory webinar every Thurs:<br>> > http://www.asterisk.org/hello<br>> ><br>> > asterisk-users mailing list<br>> > To UNSUBSCRIBE or update options visit:<br>> > http://lists.digium.com/mailman/listinfo/asterisk-users<br>> <br>> -- <br>> Sherwood McGowan <sherwood.mcgowan@gmail.com><br>> Carrier, ITSP, Call Center, and PBX Solutions Consultant<br>> <br>> <br>> --<br>> _____________________________________________________________________<br>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>> http://www.asterisk.org/hello<br>> <br>> asterisk-users mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> http://lists.digium.com/mailman/listinfo/asterisk-users<br> </body>
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