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...the channel Vars in order to do that. In Latest Asterisk you can set Channel vars within the create command in the Body. Just set the PJSIP(add,PAI) as you would do it in Dialplan. https://blogs.asterisk.org/2020/07/22/ari-create-channel-with-variables/ BR Jöran On Fri, Aug 7, 2020 at 7:06 PM Dan Cropp <dan at amtelco.com> wrote: > An additional follow-up question, if I need to set the P-Asserted-Identity > on the create (originate), is there a way to do this with ARI? > > > > *From:* asterisk-users <asterisk-users-bounces at lists.digium.com> *On > Behalf Of *Da...

...for the name curl -v -u asterisk:asterisk -X POST http://asterisk:astersk at localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003 at 1003&app=hello-world&extension=1000&context=mycontext&priority=1&channelId=mycallerid.1&formats=ulaw&timeout=30&callerId=Dan Cropp<291> Here is an example of how we do this with AMI successfully. Action: Originate ActionID: S40 Channel: PJSIP/1003 at 1003 Exten: createcall Context: IS Priority: 1 Timeout: 60000 CallerID: Dan Cropp <291> Variable: CALLERID(num-pres)=allowed_passed_screen,TrunkAllocateId=1,Originate...

...0609726.3/play?media=sound:hello-world> --data '{ "endpoint": "SIP/Alice", "variables": { "CALLERID(name)": "Alice" , "PJSIP_HEADER(add,P-Asserted-Identity":"foobar"} }' BR Jöran On Mon, Aug 10, 2020 at 3:43 PM Dan Cropp <dan at amtelco.com> wrote: > Hi Jöran, > > > > Would it be possible to see an example using curl of how you are passing > the PAI Header through ARI create? > > > > Dan > > > > *From:* asterisk-users <asterisk-users-bounces at lists.digium.com&gt...

On Mon, Dec 15, 2014 at 3:33 PM, Dan Cropp <dan at amtelco.com> wrote: > > Hi George, > > > > Thank you for looking into this. > > This is behind a nat? > > > Just to be clear...both the pbx and local endpoints are behind the same NAT? > [global] > > type = global > > debug = yes &gt...

...m: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joshua Colp Sent: Wednesday, December 10, 2014 9:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PJSIP configuration question Kia ora, Dan Cropp wrote: > I'm working with a SIP provider to try and transition our sip > connection with them to PJSIP. I thought I had transitioned the > settings correctly, but whenever I attempt an Originate it never even > tries to send any PJSIP messages. What dial string are you providing...

...d> > --data '{ "endpoint": "SIP/Alice", "variables": { "CALLERID(name)": > "Alice" , "PJSIP_HEADER(add,P-Asserted-Identity":"foobar"} }' > > BR > Jöran > > > On Mon, Aug 10, 2020 at 3:43 PM Dan Cropp <dan at amtelco.com> wrote: > >> Hi Jöran, >> >> >> >> Would it be possible to see an example using curl of how you are passing >> the PAI Header through ARI create? >> >> >> >> Dan >> >> >> >> *From:* aste...

On Mon, Dec 15, 2014 at 3:54 PM, Dan Cropp <dan at amtelco.com> wrote: > > Yes, everything is behind the same NAT. > > > > For the application I?m working on, the only endpoint is the endpoint to > Vitelity. > > We use AMI to Originate calls from Asterisk endpoint through Vitelity to > phones. > > A...

...not in the ACK Is there a setting to indicate whether the Contact field should be sent as part of the ACK (response to the OK)? Have a great day! Dan -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Dan Cropp Sent: Thursday, December 11, 2014 9:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PJSIP configuration question This fixed the problem. Developer before me wrote some code to build up the dial string. Always thought that string appeared off, but it...

...the channel Vars in order to do that. In Latest Asterisk you can set Channel vars within the create command in the Body. Just set the PJSIP(add,PAI) as you would do it in Dialplan. https://blogs.asterisk.org/2020/07/22/ari-create-channel-with-variables/ BR Jöran On Fri, Aug 7, 2020 at 7:06 PM Dan Cropp <dan at amtelco.com<mailto:dan at amtelco.com>> wrote: An additional follow-up question, if I need to set the P-Asserted-Identity on the create (originate), is there a way to do this with ARI? From: asterisk-users <asterisk-users-bounces at lists.digium.com<mailto:asterisk-users-...

On Thu, Mar 28, 2019, at 11:10 AM, Dan Cropp wrote: > > Is there no one who knows if there is a way to turn off the norefersub setting? > > > Supported: norefersub > > > This happens in the TRYing, OK, and other commands in response to the INVITE. > > > For chan_sip, I noticed it does not send the nor...

An additional follow-up question, if I need to set the P-Asserted-Identity on the create (originate), is there a way to do this with ARI? From: asterisk-users <asterisk-users-bounces at lists.digium.com> On Behalf Of Dan Cropp Sent: Friday, August 7, 2020 11:51 AM To: 'asterisk-users at lists.digium.com' <asterisk-users at lists.digium.com> Subject: [asterisk-users] With ARI, is it possible to create (originate) a call and pass both the caller id name and number? I'm trying to transition from AMI to AR...

...g List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Message-ID: <AANLkTikV+32vKVSkAFmkDciOPn+rO=k3jYJmsZLNj1QS at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" 3 miliseconds... 2010/10/1 Danny Nicholas <danny at debsinc.com> [Dan Cropp] Thank you. I was originally using 30, but had the same problem. Dropped the timeout to 3 thinking I wouldn't have to wait 30 seconds to replicate. Guess that's what I get for looking at the Dial command (timeout in seconds) for most of the day then trying Originate (timeout in millisec...

On Mon, Dec 15, 2014 at 2:08 PM, Dan Cropp <dan at amtelco.com> wrote: > > Today, I tried the same behavior on Asterisk 13.1.0 and Asterisk 12.2.0. > > > > Same problem is happening with both of them. > > > > Could this be caused by PJPROJECT 2.3? > > > > Anyone have any suggestions for what I...

On Mon, Dec 15, 2014 at 9:48 PM, Dan Cropp <dan at amtelco.com> wrote: > > Thanks George. > > I will correct my local_net in the morning. > > Vitelity chan_sip settings I have working, do not have a fromuser. > sip.conf settings... > > I think you can actually specify anything, it just has to be populated wi...

...members to queues, although it doesn't specifically provide an example of using local channels in a queue: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ACD_id289508.html Basically, read that book, and if you get stuck ask for help. On Thu, Aug 27, 2015 at 3:08 PM, Dan Cropp <dan at amtelco.com<mailto:dan at amtelco.com>> wrote: Thanks Scott. I?m taking over for someone else?s code, so I must admit I?m still learning the Agent and Queue concepts. Local channels are something I have not used either. Would local channels essentially be an internal bridge?...

...Re: [asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet? Are you using this method of setting headers on PJSIP? https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Function_PJSIP_HEADER On Thu, Aug 27, 2015 at 4:54 PM, Dan Cropp <dan at amtelco.com<mailto:dan at amtelco.com>> wrote: Thanks Scott. I was able to get the basic concept to run. However, it seems PJSIP INVITE for the Dial also does not support added headers. The Local channel dial plan did have the channel variable values. I added them as SIP head...

On Mon, Aug 26, 2019, at 2:00 PM, Dan Cropp wrote: > Thank you Joshua. > > Out of curiosity, what is the maximum capacity you have heard for > simultaneous ConfBridges in a single box? (Looking at 3-4 channels per > ConfBridge) with recording. I don't really remember any specific values. 100? 200? -- Joshua C. Colp...

...than you can get with Pocketsphinx. We use Lumenvox with UniMRCP for most ASR use cases, but with 100,000 options it might very well be the only solution for you. Mind if I ask what the 100k options are for? Person names for a directory? Best regards, Luca On Wed, Feb 22, 2017 at 4:43 PM, Dan Cropp <dan at amtelco.com<mailto:dan at amtelco.com>> wrote: Is it correct that the unimrcp is the best approach for Asterisk and ASR/TTS? Could anyone provide pros/cons for the various ASR options for Asterisk? We need the ability for very large grammars (over 100,000 options). Because of...

...s at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of George Joseph Sent: Monday, December 15, 2014 4:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PJSIP configuration question On Mon, Dec 15, 2014 at 3:54 PM, Dan Cropp <dan at amtelco.com<mailto:dan at amtelco.com>> wrote: Yes, everything is behind the same NAT. For the application I?m working on, the only endpoint is the endpoint to Vitelity. We use AMI to Originate calls from Asterisk endpoint through Vitelity to phones. After that, we control the...

...Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joshua Colp Sent: Tuesday, July 14, 2015 2:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] pjsip.conf question Dan Cropp wrote: <snip> > > My pjsip.conf looks like... > > [transport1] > > type = transport > > bind = 0.0.0.0 > > protocol = udp > > [3400] > > type = aor > > max_contacts = 1 > > remove_existing = yes > > contact=sip:xxx.xxx.xxx.xxx >...