upendra
2012-Jul-13 09:42 UTC
[asterisk-users] How to set SIP to auto answer in the dial plan .
Hi, I am trying to write dial plan for sip to auto answer (auto attend) the incoming call to the sip phone. - If i call from sip1 to sip2 then sip2 should automatically answer the call and play some sound file. I am trying to do this but as new to the asterisk dial plan configuration , so not able Todo this. help me if anyone already done this setup. Regards Upendra. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120713/4921de8c/attachment-0001.htm>
Zohair Raza
2012-Jul-13 09:45 UTC
[asterisk-users] How to set SIP to auto answer in the dial plan .
try with SipAddHeader(uri=answer-after=0) check syntax for Addheader Regards, Zohair Raza On Fri, Jul 13, 2012 at 1:42 PM, upendra <uppi.me at gmail.com> wrote:> Hi, > > > I am trying to write dial plan for sip to auto answer (auto attend) the > incoming call to the sip phone. > > - If i call from sip1 to sip2 then sip2 should automatically answer the call > and play some sound file. > I am trying to do this but as new to the asterisk dial plan configuration , > so not able Todo this. > help me if anyone already done this setup. > > > > Regards > Upendra. > > -- > _____________________________________________________________________ > -- 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
upendra
2012-Jul-13 09:50 UTC
[asterisk-users] How to set SIP to auto answer in the dial plan .
Hi, thanks , i need to put this in the sip context...???? regards Upendra. On Fri, Jul 13, 2012 at 3:15 PM, Zohair Raza <engineerzuhairraza at gmail.com>wrote:> try with SipAddHeader(uri=answer-after=0) > > check syntax for Addheader > > Regards, > Zohair Raza > > > > > On Fri, Jul 13, 2012 at 1:42 PM, upendra <uppi.me at gmail.com> wrote: > > Hi, > > > > > > I am trying to write dial plan for sip to auto answer (auto attend) the > > incoming call to the sip phone. > > > > - If i call from sip1 to sip2 then sip2 should automatically answer the > call > > and play some sound file. > > I am trying to do this but as new to the asterisk dial plan > configuration , > > so not able Todo this. > > help me if anyone already done this setup. > > > > > > > > Regards > > Upendra. > > > > -- > > _____________________________________________________________________ > > -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120713/3ba8262f/attachment.htm>
Zohair Raza
2012-Jul-13 09:56 UTC
[asterisk-users] How to set SIP to auto answer in the dial plan .
In dialplan http://www.voip-info.org/wiki/view/Asterisk+cmd+SipAddHeader Regards, Zohair Raza On Fri, Jul 13, 2012 at 1:50 PM, upendra <uppi.me at gmail.com> wrote:> Hi, > > thanks , i need to put this in the sip context...???? > > regards > Upendra. > > > On Fri, Jul 13, 2012 at 3:15 PM, Zohair Raza <engineerzuhairraza at gmail.com> > wrote: >> >> try with SipAddHeader(uri=answer-after=0) >> >> check syntax for Addheader >> >> Regards, >> Zohair Raza >> >> >> >> >> On Fri, Jul 13, 2012 at 1:42 PM, upendra <uppi.me at gmail.com> wrote: >> > Hi, >> > >> > >> > I am trying to write dial plan for sip to auto answer (auto attend) the >> > incoming call to the sip phone. >> > >> > - If i call from sip1 to sip2 then sip2 should automatically answer the >> > call >> > and play some sound file. >> > I am trying to do this but as new to the asterisk dial plan >> > configuration , >> > so not able Todo this. >> > help me if anyone already done this setup. >> > >> > >> > >> > Regards >> > Upendra. >> > >> > -- >> > _____________________________________________________________________ >> > -- 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 > > > > -- > _____________________________________________________________________ > -- 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
Jared Baxley
2012-Jul-13 16:27 UTC
[asterisk-users] How to set SIP to auto answer in the dial plan .
You also have to send the alert info you particular phone needs to make it autoanswer. On Jul 13, 2012 4:53 AM, "upendra" <uppi.me at gmail.com> wrote:> Hi, > > thanks , i need to put this in the sip context...???? > > regards > Upendra. > > On Fri, Jul 13, 2012 at 3:15 PM, Zohair Raza <engineerzuhairraza at gmail.com > > wrote: > >> try with SipAddHeader(uri=answer-after=0) >> >> check syntax for Addheader >> >> Regards, >> Zohair Raza >> >> >> >> >> On Fri, Jul 13, 2012 at 1:42 PM, upendra <uppi.me at gmail.com> wrote: >> > Hi, >> > >> > >> > I am trying to write dial plan for sip to auto answer (auto attend) the >> > incoming call to the sip phone. >> > >> > - If i call from sip1 to sip2 then sip2 should automatically answer the >> call >> > and play some sound file. >> > I am trying to do this but as new to the asterisk dial plan >> configuration , >> > so not able Todo this. >> > help me if anyone already done this setup. >> > >> > >> > >> > Regards >> > Upendra. >> > >> > -- >> > _____________________________________________________________________ >> > -- 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 >> > > > -- > _____________________________________________________________________ > -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120713/5e4f8e05/attachment.htm>
upendra
2012-Jul-14 05:50 UTC
[asterisk-users] How to set SIP to auto answer in the dial plan .
Hi, its not working for me ....! let me know anyone having sample dialplan so that i can use for test 1 sip call answer. regards Upendra On Fri, Jul 13, 2012 at 9:57 PM, Jared Baxley <jared.baxley at gmail.com>wrote:> You also have to send the alert info you particular phone needs to make it > autoanswer. > On Jul 13, 2012 4:53 AM, "upendra" <uppi.me at gmail.com> wrote: > >> Hi, >> >> thanks , i need to put this in the sip context...???? >> >> regards >> Upendra. >> >> On Fri, Jul 13, 2012 at 3:15 PM, Zohair Raza < >> engineerzuhairraza at gmail.com> wrote: >> >>> try with SipAddHeader(uri=answer-after=0) >>> >>> check syntax for Addheader >>> >>> Regards, >>> Zohair Raza >>> >>> >>> >>> >>> On Fri, Jul 13, 2012 at 1:42 PM, upendra <uppi.me at gmail.com> wrote: >>> > Hi, >>> > >>> > >>> > I am trying to write dial plan for sip to auto answer (auto attend) the >>> > incoming call to the sip phone. >>> > >>> > - If i call from sip1 to sip2 then sip2 should automatically answer >>> the call >>> > and play some sound file. >>> > I am trying to do this but as new to the asterisk dial plan >>> configuration , >>> > so not able Todo this. >>> > help me if anyone already done this setup. >>> > >>> > >>> > >>> > Regards >>> > Upendra. >>> > >>> > -- >>> > _____________________________________________________________________ >>> > -- 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 >>> >> >> >> -- >> _____________________________________________________________________ >> -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120714/bbcff687/attachment.htm>
Larry Moore
2012-Jul-14 07:30 UTC
[asterisk-users] How to set SIP to auto answer in the dial plan .
I have the following in my intercom macro in extensions.ael; SIPAddHeader(Alert-Info: Ring Answer); SIPAddHeader(Call-Info:\;Answer-After=0); SIPAddHeader(P-Auto-Answer: normal); If memory serves me, respectively they are for the following vendors; Grandstream Linksys/Cisco SPA Yealink Larry. On 14/07/2012 1:50 PM, upendra wrote:> Hi, > > its not working for me ....! let me know anyone having sample dialplan > so that i can use for test 1 sip call answer. > > > > regards > Upendra > > On Fri, Jul 13, 2012 at 9:57 PM, Jared Baxley <jared.baxley at gmail.com > <mailto:jared.baxley at gmail.com>> wrote: > > You also have to send the alert info you particular phone needs to > make it autoanswer. > > On Jul 13, 2012 4:53 AM, "upendra" <uppi.me at gmail.com > <mailto:uppi.me at gmail.com>> wrote: > > Hi, > > thanks , i need to put this in the sip context...???? > > regards > Upendra. > > On Fri, Jul 13, 2012 at 3:15 PM, Zohair Raza > <engineerzuhairraza at gmail.com > <mailto:engineerzuhairraza at gmail.com>> wrote: > > try with SipAddHeader(uri=answer-after=0) > > check syntax for Addheader > > Regards, > Zohair Raza > > > > > On Fri, Jul 13, 2012 at 1:42 PM, upendra <uppi.me at gmail.com > <mailto:uppi.me at gmail.com>> wrote: > > Hi, > > > > > > I am trying to write dial plan for sip to auto answer > (auto attend) the > > incoming call to the sip phone. > > > > - If i call from sip1 to sip2 then sip2 should > automatically answer the call > > and play some sound file. > > I am trying to do this but as new to the asterisk dial > plan configuration , > > so not able Todo this. > > help me if anyone already done this setup. > > > > > > > > Regards > > Upendra. > > > > -- > > > _____________________________________________________________________ > > -- 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 > > > > -- > _____________________________________________________________________ > -- 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 > > > > > -- > _____________________________________________________________________ > -- 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 >
Doug Lytle
2012-Jul-14 12:01 UTC
[asterisk-users] How to set SIP to auto answer in the dial plan .
Larry Moore wrote:> If memory serves me, respectively they are for the following vendors;And Polycom: exten => s,n,SIPAddHeader(Alert-Info: Ring Answer) Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."
Ron Bergin
2012-Jul-14 19:23 UTC
[asterisk-users] How to set SIP to auto answer in the dial plan .
upendra wrote:> Hi, > > > I am trying to write dial plan for sip to auto answer (auto attend) the > incoming call to the sip phone. > > - If i call from sip1 to sip2 then sip2 should automatically answer the > call and play some sound file. > I am trying to do this but as new to the asterisk dial plan configuration > , > so not able Todo this. > help me if anyone already done this setup. > > > > Regards > Upendra. > --Unless I'm misunderstanding your needs, wouldn't this do what you want? exten => 1234,1,Answer exten => 1234,n,Playback(soundfile) exten => 1234,n,Dial(SIP/1234,60,m) ; caller hears music on hold ; instead of ringtone -- Ron Bergin