similar to: asterisk-users Digest, Vol 75, Issue 2

When calling Originate Action, it rings my phone. If I do not answer, I receive a Channel Event: Hangup, followed by receiving an OriginateResponse event with a Failure Response, Reason 3. My phone continues to ring. If I answer the phone at this point, it attempts to answer, but does not succeed. Looking at the asterisk debug, it appears to destroy the SIP dialog for the call. It also

Hi Dan, as far as PPI and PAI Header, we use 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

I'm trying to transition from AMI to ARI. Running into a small hiccup when I try to create (originate a call) with the caller id name and number I can pass the Name and Number if the name has no spaces in it and it shows up in my PhonerLite application. curl -v -u asterisk:asterisk -X POST http://asterisk:astersk at localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003 at

Dear R-help, I have fitted a glm logistic function to dichotomous forced choices responses varying according to time interval between two stimulus. x values are time separation in miliseconds, and the y values are proportion responses for one of the stimulus. Now I am trying to extrapolate x values for the y value (proportion) at .25, .5, and .75. I have tried several predict parameters, and they

Hi Dan, i would do something like this (it is not a copy of what we are doing but an example of how i would do it) Important here is the "--data" and "-H" Option as well as the "variables" section within the Body. I added the callerid function here as well as it is the sample in the asterisk wiki. curl -v -H "Content-Type: application/json" -u

Im trying to improve my internal apache proxy. It has to deliver a lot of little/medium sized files. But every transfer starts with the usual small window size. While this is good for internet connections it is not as good for only internal connections where the environment is sane. I have tryed to tune initial window size via /proc/sys/net/ipv4/tcp_congestion_control tryed already

How does one use the vcut from Vorbis-tools package? I tried to enter samples, miliseconds, seconds, h:mm:ss as cut point, but nothing worked (I even tried to enter 1 as cutpoint, but I always got "Cutpoint not within stream." message)... I'm doing this on Windows, if it matters... -- Jernej Simoncic, jernej.simoncic@guest.arnes.si http://www2.arnes.si/~sopjsimo/ ICQ: 26266467

>>>>> "DC" == D'Arcy Cain <darcy at VybeNetworks.com> writes: DC> Perhaps someone can explain what t38timeout is supposed to do A 'git grep t38timeout' on the src leads one to res/res_fax.c, where one case see that it is the number of miliseconds to permit for t38 negotiation to complete once it starts. Ie after both sides select t38, until they

Hello, I want to run an r script that contains code for a gui (rgtk) on the command line (windows 2000, 32 bits) using R2.10.1, but the Gui disappears a few miliseconds after I started the program. What switch should I use to prevent this? I tried r.exe, rterm.exe and rscript.exe with various combinations of switches, but none of them works.   TIA Cheers!! Albert-Jan

Dear R-list, I have a plot with y-axis corresponding to wind measurments and x-axis with date-time information. When I want to identify some extrem wind events in the wind-curve, I use locator() to get the exact date-information, by clicking in the points in graph I?m interested in. I get in the R console the x and y coordinates. The x coordinates are not in a POSIXct format, I guess R is

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 > > > > [transport1] > > type = transport

Thank you for the speedy reply. My originate string is something like the following where xxxxx is really the sip provider's supplied IP address 1234567890 is really the phone number I am dialing PJSIP/outbound.vitelity.net/1234567890 In the chan_sip based solution, it's... SIP/outbound.vitelity.net/1234567890 Have a great day! Dan -----Original Message----- From:

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 control the call through AMI to perform the

Ok, it didn't quite solve everything. There is one slight issue. When I answer the call on my cell phone, Asterisk sees it as answered. I can play audio, send dtmfs, etc and hear it on my phone. However, a short while later, Vitelity tears down that call and Asterisk is never notified about it. I tell Asterisk to hang up the call (via AMI) and it is removed from Asterisk. I gather the

what-he-said On 08/07/2013 06:48 AM, Basil Mohamed Gohar wrote: > On 08/06/2013 07:40 PM, Jorge N??ez wrote: >> Hi I am trying to use icecast to broadcast a realtime conference from >> freeswitch. But I am having a delay like 20 seconds then I reduced it to >> 12s. But I don't know if somebody can help me how to reduce it as lower >> as possible. >> >>

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 norefersub. As a result, > Cisco then sends NOTIFY

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? > > > > My boss is giving me until

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 than a sub-account username. >

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(PJSIP/Agent). The INVITE for the Dial on PJSIP continues to not include the SIP Headers I added. For chan_sip, I have no problem with this. Even the

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 Digium - A Sangoma Company | Senior Software Developer