Re: [asterisk-users] How to correctly set REDIRECTING to indicate diversion reason
On Mon, Nov 20, 2017 at 8:43 AM, Benoit Panizzonwrote: > Hello List > > Next question where google did not spit out an unsable answer. > > When redirecting a call with Transfer, I would like to correctly > indicate the reason. > > I did try this: > > exten => XX,1,NoOp(Call to ${EXTEN} from ${CALLERID(all)}) > exten => XX,n,Dial(SIP/ZZ) > exten => XX,n,set(REDIRECTING(reason)=cfb) > exten => XX,n,Transfer(SIP/YY) > > I did try with 'reason' 'orig-reason' I added cfb with and only quotes, > I did try cfnr. > > But the 302 message generated this way allways contains reason=unknown > in the diversion header. > > Any hint welcome. > You need to set more redirecting information [1]. In sip.conf send_diversion=yes needs to be in effect. You also need to setup the from party id information (at least the from number) to indicate where you are redirecting from. You should also increment the redirecting count. chan_pjsip has the same requirements. pjsip.conf send_diversion=yes needs to be in effect and you also need to setup the from party id information. Richard [1] https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information -- _ -- 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] Ringing (180) no SDP to progress(183) with SDP transition => no audio.
On Mon, Nov 20, 2017 at 7:31 AM, Benoit Panizzonwrote: > Dear List > > I am testing various early audio scenarios with different voice IC's, > phones and pbxes. > > In Switzerland, when you operate a value added number, you have to > announce the price of the call, usually in early audio, before the call > is established. > > In 'dialplan' terms this would be: > > exten => XX,1,Ringing > exten => XX,n,Wait(15) > exten => XX,n,Progress > exten => XX,n,Playback(price-announce,noanswer) > exten => XX,n,Wait(5) > exten => XX,n,Answer > > I see the asterisk playing the early announcement audio in the rtp > stream. Some devices (arris EMTA) calling the asterisk also do play it > to the caller. > > But! > > Most other devices I have tested just keep playing the locally generated > ringtone despite getting an 183 with SDP and the announcement is never > to be heard by the caller. > > If I do to force inband ringback tone, this works with all devices I > have tested so far. > > exten => XX,1,Progress > exten => XX,n,Ringing > exten => XX,n,Wait(15) > exten => XX,n,Playback(price-announce,noanswer) > exten => XX,n,Wait(5) > exten => XX,n,Answer > > Is anything wrong with the transition of ringing without SDP (to have > the local device generating ringback tone) and then start sending early > audio with 183? > Both orderings of Ringing and Progress are valid. It is up to the calling device to handle it. As you have seen, there is quite a difference in how devices handle it. I have even seen where the calling device needs Ringing before Progress to handle the call correctly. I think that case was because the device was converting ISDN to SIP. I do think that the devices that don't stop local ringback in favor of the incoming RTP stream following the 183 are broken. Unfortunately it is something that is out of your control. Richard -- _ -- 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] How to correctly set REDIRECTING to indicate diversion reason
Hello List Next question where google did not spit out an unsable answer. When redirecting a call with Transfer, I would like to correctly indicate the reason. I did try this: exten => XX,1,NoOp(Call to ${EXTEN} from ${CALLERID(all)}) exten => XX,n,Dial(SIP/ZZ) exten => XX,n,set(REDIRECTING(reason)=cfb) exten => XX,n,Transfer(SIP/YY) I did try with 'reason' 'orig-reason' I added cfb with and only quotes, I did try cfnr. But the 302 message generated this way allways contains reason=unknown in the diversion header. Any hint welcome. -Benoît Panizzon- -- I m p r o W a r e A G-Leiter Commerce Kunden __ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 PrattelnFax +41 61 826 93 01 Schweiz Web http://www.imp.ch __ -- _ -- 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] Ringing (180) no SDP to progress(183) with SDP transition => no audio.
Dear List I am testing various early audio scenarios with different voice IC's, phones and pbxes. In Switzerland, when you operate a value added number, you have to announce the price of the call, usually in early audio, before the call is established. In 'dialplan' terms this would be: exten => XX,1,Ringing exten => XX,n,Wait(15) exten => XX,n,Progress exten => XX,n,Playback(price-announce,noanswer) exten => XX,n,Wait(5) exten => XX,n,Answer I see the asterisk playing the early announcement audio in the rtp stream. Some devices (arris EMTA) calling the asterisk also do play it to the caller. But! Most other devices I have tested just keep playing the locally generated ringtone despite getting an 183 with SDP and the announcement is never to be heard by the caller. If I do to force inband ringback tone, this works with all devices I have tested so far. exten => XX,1,Progress exten => XX,n,Ringing exten => XX,n,Wait(15) exten => XX,n,Playback(price-announce,noanswer) exten => XX,n,Wait(5) exten => XX,n,Answer Is anything wrong with the transition of ringing without SDP (to have the local device generating ringback tone) and then start sending early audio with 183? -Benoît Panizzon- -- I m p r o W a r e A G-Leiter Commerce Kunden __ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 PrattelnFax +41 61 826 93 01 Schweiz Web http://www.imp.ch __ -- _ -- 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] How to supervise a Voicemail box with a BLF button ? What does "State:Unavailable" exactly means ?
Hello, I'm trying to supervise an existing Voicemail box with a BLF button on Debian's asterisk 13.14.1 system. I mostly found this [1] document. I added in a context a line like: exten = *7000,hint,MWI:31@default With "core show hints", I can read this: *7000@subs : MWI:31@defaultState:Unavailable Presence:not_set Watchers 1 My questions are: 1. Is this "exten = *7000,hint,MWI:31@default" statement correct ? 2. What does "State:Unavailable" exactly means ? To it means "Asterisk is unable to find any MWI:31@default state". 3. Which Asterisk version introduced this MWI/BLF/hint feature ? Was it supported in 1.6, for instance ? Best regards [1] https://community.asterisk.org/t/hint-mwi-help-i-am-not-able-to-make-it-works/72112 -- _ -- 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