Re: [asterisk-users] tel URI
Hi Matt, Thanks for prompt reply. Le 30/01/2019 à 22:38, Matthew Fredrickson a écrit : > Right now, chan_pjsip does not properly handle tel: URIs. If you need > them you might need to use chan_sip. ok, I'm back on chan_sip, but I still do not see how I can send outgoing calls with tel: uri scheme. Is it supported? Thanks, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française https://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.797.527 signature.asc Description: OpenPGP digital signature -- _ -- 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] Dailplan with playtones
Hello I use this dial paln: [o2-in] exten => o2,1,Answer exten => o2,n,Playback(hello-world) exten => o2,n,Ringing exten => o2,n,Dial(SIP/10/20/s@no-op,25,rt) exten => o2,n,Playtones(425/1000,0/4000) exten => o2,n,Wait(30) exten => o2,n,Hangup() All is fine. Hello world is Playback and I hear a ring tone. If I remove the Playback hello-world. No ring tone is hearing anymore. Also not when my softphone is ring. What I want is: - Play caller a ring tone when softphone is dial or even if Subscriber absent - han up after a while I try this https://www.voip-info.org/asterisk-cmd-playtones Best Regards -- _ -- 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] Dailplan with playtones
With softphone I mean linphone csipsimple or whatever. How should a dialplan lokks like? On 31.01.19 11:26, Antony Stone wrote: > On Thursday 31 January 2019 at 10:59:01, basti wrote: > >> Hello I use this dial paln: >> >> [o2-in] >> exten => o2,1,Answer >> exten => o2,n,Playback(hello-world) >> exten => o2,n,Ringing >> exten => o2,n,Dial(SIP/10/20/s@no-op,25,rt) >> exten => o2,n,Playtones(425/1000,0/4000) >> exten => o2,n,Wait(30) >> exten => o2,n,Hangup() >> >> All is fine. Hello world is Playback and I hear a ring tone. > > That seems a most odd thing to want the caller to hear. > >> If I remove the Playback hello-world. No ring tone is hearing anymore. > > I can't say I'm surprised, given that you've already Answer()ed the call. > >> Also not when my softphone is ring. > > Please be more specific what you mean by that? > >> What I want is: >> >> - Play caller a ring tone when softphone is dial or even if Subscriber >> absent >> - han up after a while > > Well, simply don't Answer() the incoming call until the subsequent Dial() has > succeeded. > >> I try this https://www.voip-info.org/asterisk-cmd-playtones > > Is there a good reason why you are Answer()ing the call (and possibly thereby > causing the caller to incur charges) before knowing the outcome of the Dial()? > > > Antony. > -- _ -- 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] T-38 re-invite issue
On 7/3/18 3:57 PM, D'Arcy Cain wrote: > On 2018-06-13 07:45 AM, D'Arcy Cain wrote: >> On 2018-06-13 07:20 AM, James Cloos wrote: D'Arcy Cain writes: >>> > Ie after both sides select t38, until they agree on the t38 terms. >>> OK, so does that mean that setting it to 25000 should leave time for the re-invite or does the timeout start after that. >>> >>> As I wrote above, after that. After the sip/sdp. >> >> So, how do I increase the timeout before the re-invite then? Source? > > Does anyone have any ideas on this? We now have another number that is > failing the same way. If someone can point me to the area in the source > where this other timeout is I will try to make the change there. > Perhaps I can even send a patch to control it. I don't want to be a total pest but I can't be the only one who has been tripped up by this. Please, does anyone have any clue about this? -- D'Arcy J.M. Cain Vybe Networks Inc. http://www.VybeNetworks.com/ IM:da...@vex.net VoIP: sip:da...@vybenetworks.com -- _ -- 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] Dailplan with playtones
On Thursday 31 January 2019 at 11:36:05, basti wrote: > With softphone I mean linphone csipsimple or whatever. I know what you mean by "a softphone"; I just wasn't sure how you were calling your softphone and what you were saying (didn't) happen. > How should a dialplan lokks like? Have you tried: [o2-in] exten => o2,1,Dial(SIP/10/20/s@no-op,25,rt) same => n, Hangup() Also, are you certain that "o2" is a sensible extension to be expecting? Maybe it is, but I would have more expected to see "s" or even "_X." as the extension. It depends on what is feeding into you "o2-in" context, of course. Antony. > On 31.01.19 11:26, Antony Stone wrote: > > On Thursday 31 January 2019 at 10:59:01, basti wrote: > >> Hello I use this dial paln: > >> > >> [o2-in] > >> exten => o2,1,Answer > >> exten => o2,n,Playback(hello-world) > >> exten => o2,n,Ringing > >> exten => o2,n,Dial(SIP/10/20/s@no-op,25,rt) > >> exten => o2,n,Playtones(425/1000,0/4000) > >> exten => o2,n,Wait(30) > >> exten => o2,n,Hangup() > >> > >> All is fine. Hello world is Playback and I hear a ring tone. > > > > That seems a most odd thing to want the caller to hear. > > > >> If I remove the Playback hello-world. No ring tone is hearing anymore. > > > > I can't say I'm surprised, given that you've already Answer()ed the call. > > > >> Also not when my softphone is ring. > > > > Please be more specific what you mean by that? > > > >> What I want is: > >> > >> - Play caller a ring tone when softphone is dial or even if Subscriber > >> absent > >> - han up after a while > > > > Well, simply don't Answer() the incoming call until the subsequent Dial() > > has succeeded. > > > >> I try this https://www.voip-info.org/asterisk-cmd-playtones > > > > Is there a good reason why you are Answer()ing the call (and possibly > > thereby causing the caller to incur charges) before knowing the outcome > > of the Dial()? > > > > > > Antony. -- Normal people think "If it ain't broke, don't fix it". Engineers think "If it ain't broke, it doesn't have enough features yet". Please reply to the list; please *don't* CC me. -- _ -- 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] Dailplan with playtones
On Thursday 31 January 2019 at 10:59:01, basti wrote: > Hello I use this dial paln: > > [o2-in] > exten => o2,1,Answer > exten => o2,n,Playback(hello-world) > exten => o2,n,Ringing > exten => o2,n,Dial(SIP/10/20/s@no-op,25,rt) > exten => o2,n,Playtones(425/1000,0/4000) > exten => o2,n,Wait(30) > exten => o2,n,Hangup() > > All is fine. Hello world is Playback and I hear a ring tone. That seems a most odd thing to want the caller to hear. > If I remove the Playback hello-world. No ring tone is hearing anymore. I can't say I'm surprised, given that you've already Answer()ed the call. > Also not when my softphone is ring. Please be more specific what you mean by that? > What I want is: > > - Play caller a ring tone when softphone is dial or even if Subscriber > absent > - han up after a while Well, simply don't Answer() the incoming call until the subsequent Dial() has succeeded. > I try this https://www.voip-info.org/asterisk-cmd-playtones Is there a good reason why you are Answer()ing the call (and possibly thereby causing the caller to incur charges) before knowing the outcome of the Dial()? Antony. -- This sentence contains exacly three erors. Please reply to the list; please *don't* CC me. -- _ -- 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] tel URI
Hi list, Using Asterisk 16.1.1, with PJSIP, I'm asked to build a SIP trunk to a system that uses exclusively tel: uri on inbound and outbound calls. I could not find documentation or sample config about tel:uri. Is this doable? If not possible with PJSIP, is chan_sip a better option? Any pointer would be greatly appreciated. Thanks, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française https://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.797.527 -- _ -- 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] tel URI
On Thu, Jan 31, 2019, 9:24 AM Jean-Denis Girard Hi list, > > Using Asterisk 16.1.1, with PJSIP, I'm asked to build a SIP trunk to a > system that uses exclusively tel: uri on inbound and outbound calls. I > could not find documentation or sample config about tel:uri. Is this > doable? If not possible with PJSIP, is chan_sip a better option? Any > pointer would be greatly appreciated. > Right now, chan_pjsip does not properly handle tel: URIs. If you need them you might need to use chan_sip. Matthew Fredrickson > > Thanks, > -- > Jean-Denis Girard > > SysNux Systèmes Linux en Polynésie française > https://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.797.527 > > -- > _ > -- 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
[asterisk-users] features.conf disconnect and local channels
Asterisk 16.1
This statement appears in the features.conf doc: "Note that the DTMF
features listed below only work when two channels have answered and are
bridged together. They can not be used while the remote party is ringing
or in progress. If you require this feature you can use chan_local in
combination with Answer to accomplish it."
I need attended transfer and disconnect from features.conf to work.
Below is what I came up with that seems to work fine. Is there a better
way? This seems a bit verbose.
[InternalSets]
exten =>298,1,NoOp()
same =>n,Dial(Local/M${EXTEN}@InternalSets/nj,,H)
exten =>M298,1,NoOp()
same =>n,Answer()
same =>n,GoSub(sub-voicemail,start,1(${MITCHIPHONE},${EXTEN:1}))
exten =>299,1,NoOp()
same =>n,Dial(Local/M${EXTEN}@InternalSets/nj,,H)
exten =>M299,1,NoOp()
same =>n,Answer()
same =>n,GoSub(sub-voicemail,start,1(${MLCX450},${EXTEN:1}))
[sub-voicemail]
do some checks and then Dial or send to voicemail.
--
Mitch
--
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Re: [asterisk-users] Dailplan with playtones
Thanks antony. Now it works. Just for the docu: [o2-in] exten => o2,1,Ringing exten => o2,n,Playtones(425/1000,0/4000) exten => o2,n,Dial(SIP/10/20/s@no-op,130,rt) exten => o2,n,StopPlaytones() exten => o2,n,Hangup() [no-op]; just hang up exten => s,1,Hangup(130) On 31.01.19 11:40, Antony Stone wrote: > On Thursday 31 January 2019 at 11:36:05, basti wrote: > >> With softphone I mean linphone csipsimple or whatever. > > I know what you mean by "a softphone"; I just wasn't sure how you were > calling > your softphone and what you were saying (didn't) happen. > >> How should a dialplan lokks like? > > Have you tried: > > [o2-in] > exten => o2,1,Dial(SIP/10/20/s@no-op,25,rt) > same => n, Hangup() > > Also, are you certain that "o2" is a sensible extension to be expecting? > > Maybe it is, but I would have more expected to see "s" or even "_X." as the > extension. It depends on what is feeding into you "o2-in" context, of course. > > > Antony. > >> On 31.01.19 11:26, Antony Stone wrote: >>> On Thursday 31 January 2019 at 10:59:01, basti wrote: Hello I use this dial paln: [o2-in] exten => o2,1,Answer exten => o2,n,Playback(hello-world) exten => o2,n,Ringing exten => o2,n,Dial(SIP/10/20/s@no-op,25,rt) exten => o2,n,Playtones(425/1000,0/4000) exten => o2,n,Wait(30) exten => o2,n,Hangup() All is fine. Hello world is Playback and I hear a ring tone. >>> >>> That seems a most odd thing to want the caller to hear. >>> If I remove the Playback hello-world. No ring tone is hearing anymore. >>> >>> I can't say I'm surprised, given that you've already Answer()ed the call. >>> Also not when my softphone is ring. >>> >>> Please be more specific what you mean by that? >>> What I want is: - Play caller a ring tone when softphone is dial or even if Subscriber absent - han up after a while >>> >>> Well, simply don't Answer() the incoming call until the subsequent Dial() >>> has succeeded. >>> I try this https://www.voip-info.org/asterisk-cmd-playtones >>> >>> Is there a good reason why you are Answer()ing the call (and possibly >>> thereby causing the caller to incur charges) before knowing the outcome >>> of the Dial()? >>> >>> >>> Antony. > -- _ -- 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
