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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. > > After that, we contr...

....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 call through AMI to...

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 > > > > [t...

...s, 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? How would I ?Regis...

...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 this, my initial th...

...terisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Is it possible to have two endpoints to the same IP address where one uses IP based authentication and the other requires asterisk to register to that system? On Mon, Dec 18, 2017 at 9:04 AM, Dan Cropp <dan at amtelco.com<mailto:dan at amtelco.com>> wrote: Thanks George I originally didn?t have the 1002@ for the identify. Changed that when things were not working. I changed it back. Unfortunately, the system I am connecting with doesn?t seem to support the line support. Looking at the SIP pack...

...ter Local/number at agent in the queue on behalf of the agent (replace number with the agent's extension number) In dialplan [agent], wild card match the number, add the header, and then Dial(PJSIP/{$EXTEN}) to send the call to the agent. On Thu, Aug 27, 2015 at 1:40 PM, Dan Cropp <dan at amtelco.com<mailto:dan at amtelco.com>> wrote: I have a call coming in. I need to add a SIP Header to the channel. Then, I need to send the call to the Queue so it is sent to the Agent. The SIP header I added, I need to have appear in the INVITE sent to the Agent. It works in chan_sip. I send t...

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 can try? > >...

....com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of George Joseph Sent: Monday, December 15, 2014 4:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PJSIP configuration question On Mon, Dec 15, 2014 at 3:33 PM, Dan Cropp <dan at amtelco.com<mailto: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 [transport1] type = transport bind = 0.0.0.0 protocol = udp [outbo...

...ers] 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 headers, then Dial(PJSI...

...om [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of George Joseph Sent: Wednesday, December 10, 2014 2:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PJSIP configuration question On Wed, Dec 10, 2014 at 1:27 PM, Dan Cropp <dan at amtelco.com<mailto:dan at amtelco.com>> wrote: Not sure why, but Vitelity changed the settings to IP based authentication on me. Here's the new sip.conf settings they sent me. type=friend dtmfmode=auto host=64.2.142.93 allow=all nat=yes canreinvite=no trustrpid=yes sendrpid=yes When I use t...

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 with something other...

I have an endpoint with multiple phones registered as aor contacts. When I attempt to originate a call it will only ring one of the phones. Is it possible to ring multiple phones as a single endpoint. First phone to answer wins the call and all others terminated? Again, using AMI. Dan -------------- next part -------------- An HTML attachment was scrubbed... URL:

...m.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of George Joseph Sent: Monday, December 15, 2014 11:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PJSIP configuration question On Mon, Dec 15, 2014 at 9:48 PM, Dan Cropp <dan at amtelco.com<mailto: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 with something other than a su...

On Wed, Aug 9, 2023 at 3:20 PM Dan Cropp <dcropp at amtelco.com> wrote: > I have a customer who just encountered a crash while running Asterisk > 18.17.1 version. > > > > I’m trying to adapt to the changes so not sure where best to look or how > to possibly report this. > > > > I started by going through > https://git...

We are looking to migrate from AMI to ARI. We currently rely heavily on ConfBridges for multiple party support. Is it possible to add more than 2 channels? If so, is there a limit? Or a way to configure the limit? Have a great day! Dan -------------- next part -------------- An HTML attachment was scrubbed... URL:

The PJSIP endpoint is configured for ulaw only. Not sure how or why we are seeing the g729 on calls for this endpoint. Would this be a case that asterisk detects the rtp stream is g729 even though it's negotiated as ulaw? Why would asterisk change the format to g729 when disallow = all and allow = ulaw are the endpoint settings? [121] type = endpoint context = IS transport = transport1 aors

...dia=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> *On > Behalf O...

On Mon, Dec 15, 2014 at 7:34 PM, Dan Cropp <dan at amtelco.com> wrote: > > I am not sure if I entered the correct settings for the transport > information. > > For the local_net, I entered my local ip address, but no mask. I will > check with the network admin so he can verify the settings I entered. > > > You need the networ...

...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 *Dan Cropp > *Sent:...