Re: [asterisk-users] AGICommand_exec remove my double quotation
sr guys, but I found it out, commands like SET VARIABLE also have the same problem, strip off my quotations. solution is to escape the quotation with double slash \\. *EXEC CELGenUserEvent DIAL_DATA,abc\\"xyz.* It has been noted in res_agi.c rgds, Loi Dang On Wed, May 3, 2017 at 2:36 PM Lợi Đặng <loi.dangth...@gmail.com> wrote: > Good day, guys > As the subject, I'm sending this command for executing CELGenUserEvent via > AGI > *EXEC CELGenUserEvent DIAL_DATA,abc"xyz* > in asterisk cli with agi debug mode, I see that I successfully sent my > param *abc"xyz *to asterisk, but it removes my double quotation in > collecting the options to execute > cli: > > *AGI Rx << EXEC CELGenUserEvent > DIAL_DATA,abc"xyz-- AGI Script Executing Application: (CELGenUserEvent) > Options: (DIAL_DATA,abcxyz)* > > It leads to wrong data generated. > I tried other commands like *SET VARIABLE, *but my quotations don't get > removed. Single quotations are not removed in AGICommand_exec as well. > > I'm not sure what I missed, helps are appreciated. > rgds, > Loi Dang > > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGICommand_exec remove my double quotation
Good day, guys As the subject, I'm sending this command for executing CELGenUserEvent via AGI *EXEC CELGenUserEvent DIAL_DATA,abc"xyz* in asterisk cli with agi debug mode, I see that I successfully sent my param *abc"xyz *to asterisk, but it removes my double quotation in collecting the options to execute cli: *AGI Rx << EXEC CELGenUserEvent DIAL_DATA,abc"xyz-- AGI Script Executing Application: (CELGenUserEvent) Options: (DIAL_DATA,abcxyz)* It leads to wrong data generated. I tried other commands like *SET VARIABLE, *but my quotations don't get removed. Single quotations are not removed in AGICommand_exec as well. I'm not sure what I missed, helps are appreciated. rgds, Loi Dang -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] codec negotiation or transcoding issue
Asterisk might be unable to transcode rtp type from downstream to upstream, or vice versa. There's a bug reported here, for asterisk 12 or above, using chan_sip. https://issues.asterisk.org/jira/browse/ASTERISK-25676 It says that you could avoid the bug by using chan_pjsip, but you still encounter it? Turn `core set debug 5` to see whether you have `Unsupported payload type received` like I once did? rgds, On Wed, Mar 15, 2017 at 1:40 AM Faheem Muhammadwrote: > Hi, > I'm facing strange issue while establishing inbound calls from SIP trunks. > Provider A is sending (G729,Alaw,uLaw) offer and asterisk dial the peer > with its preferred codec order(G729,aLaw, uLaw). The peer's phone send the > codec list as (uLaw, speex) in 200 OK replay. The Peer's phone has selected > only uLaw and speed in this case. > > Ideally Asterisk should establish the call on uLaw codec, but Asterisk > establish the call with two codec for this call. For downstream RTP is > established with G729 and for upstream RTP is established with uLaw codec. > This behavior cause the one way audio for some phones like Eyebeam 1.5.9 > but Phonerlite latest version allow it and there is no audio issue. > > Is it normal SIP RFC 3261 behavior or there is something wrong with codec > negotiation or transcoding? > > I'm using Asterisk 13.14.0 with realtime chan_pjsip compiled with bundled > pjproject on centos 6.8_x64. I have tested it with Asterisk 11.x with > chan_sip and it works fine. > > Please advise me how can I setup the call based on late negotiation > mechanism? > > Thank you! > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users