search for: vinzen

...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 Of *Jöran Vinzens > *Sent:* Friday, August 7, 2020 12:10 PM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion < > asterisk-users at lists.digium.com> > *Subject:* Re: [asterisk-users] With ARI, is it possible to create > (originate) a call and pass both the caller id name and numb...

...asterisk systems are sending this reInvites out parallel. While an invite is pending on a system it is not accepting another incoming reInvite from peer. With chan_SIP canreinvite=no solved the issue. But it seems there is nothing similar in PJSIP. any help would be much appreciated! -- Jöran Vinzens - vinzens at sipgate.de sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.co.uk -------------- next part -------------- An HTML attachment was...

...uot;endpoint": "SIP/Alice", "variables": { "CALLERID(name)": "Alice" , "PJSIP_HEADER(add,P-Asserted-Identity)":"foobar"} }' there was a bracket missing after the function of PJSIP_HEADER BR On Mon, Aug 10, 2020 at 3:57 PM Jöran Vinzens <vinzens at sipgate.de> wrote: > 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 wit...

...It seems it has very little effect. Does anyone have similar issues or a solution? Is there anyone who calls AGI several times during call establishment? any hin and help would be very much appreciated! I am happy to share more config and information if it helps to find a solution. -- Jöran Vinzens - vinzens at sipgate.de sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.co.uk -------------- next part -------------- An HTML attachment was...

...> > > If you are using PHP, then I’m sure that the above still applies, but PHP > is not my area of expertise. > > > > Regards; > > John V. > > > > > > *From:* asterisk-users <asterisk-users-bounces at lists.digium.com> *On > Behalf Of *Jöran Vinzens > *Sent:* Friday, September 20, 2019 12:47 PM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion < > asterisk-users at lists.digium.com> > *Subject:* [asterisk-users] Load issues using AGI > > > > Hi all, > > > > we have just upgraded from Ast...

...tps://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 -- Jöran Vinzens - vinzens at sipgate.de Telefon: +49 211-63 55 56-21 Telefax: +49 211-63 55 55-22 sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.co.uk -----...

...far as i understood you Josh, there is no way to prohibit this kind of reInvite? It is not about route Optimization just for some more options for the A Party. BR Jöran On Thu, Aug 15, 2019 at 4:07 PM Joshua C. Colp <jcolp at digium.com> wrote: > On Thu, Aug 15, 2019, at 8:23 AM, Jöran Vinzens wrote: > > Hi All, > > > > We are using asterisk 16.5 and having an issue with the first re-invite > > after the call has been established. > > We can see the call gets up and you see in the logs the bridge type has > > changed and after that a re-invite is trig...

...nd we have not figured it out. There is no "/ari/channel/progress" command we can use and there is no "early_media=true" in pjsip.conf which would enable the desired behaviour. We would love to get a hint in the right direction and we very much appreciate any help. -- Jöran Vinzens - vinzens at sipgate.de sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.co.uk -------------- next part -------------- An HTML attachment was...

...our bad config looks something like this: > [default] > brokenlinehere > exten = _X.1,Noop(foo) > same = n,SomethingElse() after reloading it prints out "Line 2 contains no '='" and the reload is successful, but it shouldn't. many Thanks! BR Jöran -- Jöran Vinzens sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: <htt...

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 Of Jöran Vinzens Sent: Friday, August 7, 2020 12:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] With ARI, is it possible to create (originate) a call and pass both the caller id name and number? Hi Dan, as far as PPI and...

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

Hi Gang To increase security against phished passwords and similar attacks, we consider offering customers to define IP ranges (or GeoIP locations) from which their dynamic registrations are being accepted. I can already look at the source IP in the dial plan, so no issue with validate an INVITE against a source IP. But I would also like to prevent registrations from outside of this

...solved in this release: Bugs fixed in this release: ----------------------------------- * ASTERISK-28811 - Crash occurs when fax session switches from T.38 to audio (Reported by Alexey Vasilyev) * ASTERISK-28839 - Sporadic crashes with Segmentation fault (Reported by Joeran Vinzens) * ASTERISK-28824 - BuildSystem: Search for Python/C API when possibly needed only. (Reported by Alexander Traud) * ASTERISK-27717 - [patch] BuildSystem: In NetBSD, the Python Programming Language is python-2.7. (Reported by Alexander Traud) * ASTERISK-28859 - pjsi...

...solved in this release: Bugs fixed in this release: ----------------------------------- * ASTERISK-28811 - Crash occurs when fax session switches from T.38 to audio (Reported by Alexey Vasilyev) * ASTERISK-28839 - Sporadic crashes with Segmentation fault (Reported by Joeran Vinzens) * ASTERISK-28824 - BuildSystem: Search for Python/C API when possibly needed only. (Reported by Alexander Traud) * ASTERISK-27717 - [patch] BuildSystem: In NetBSD, the Python Programming Language is python-2.7. (Reported by Alexander Traud) * ASTERISK-28859 - pjsi...

Jon, We are only using FastAgi. On the second system (running Asterisk 16) there are no agi's running (just some bash scripts on call hangup). I did add some hackey code (netstat -nua | grep -v 'udp 0 0' | grep -v udp6 | grep -v ' 0 0.0.0.0' | grep udp) to my bash script to check out the packet queue (with the help of

...L <dialog-info> version number (Reported by Bryan Nelson) * ASTERISK-28605 - chan_dahdi: Deadlock in Hangup Scenarios with concurrent command pri show span X (Reported by Dirk Wendland) * ASTERISK-28633 - stasis bridge topic leak (Reported by Joeran Vinzens) * ASTERISK-28492 - pjsip reload not reloading wizard endpoint/pickup_group endpoint/call_group (Reported by Jean-Denis Girard) * ASTERISK-27243 - contrib: valgrind.supp doesn't suppress what it's supposed to due to invalid syntax (Reported by Richard...

...L <dialog-info> version number (Reported by Bryan Nelson) * ASTERISK-28605 - chan_dahdi: Deadlock in Hangup Scenarios with concurrent command pri show span X (Reported by Dirk Wendland) * ASTERISK-28633 - stasis bridge topic leak (Reported by Joeran Vinzens) * ASTERISK-28492 - pjsip reload not reloading wizard endpoint/pickup_group endpoint/call_group (Reported by Jean-Denis Girard) * ASTERISK-27243 - contrib: valgrind.supp doesn't suppress what it's supposed to due to invalid syntax (Reported by Richard...

...L <dialog-info> version number (Reported by Bryan Nelson) * ASTERISK-28605 - chan_dahdi: Deadlock in Hangup Scenarios with concurrent command pri show span X (Reported by Dirk Wendland) * ASTERISK-28633 - stasis bridge topic leak (Reported by Joeran Vinzens) * ASTERISK-28492 - pjsip reload not reloading wizard endpoint/pickup_group endpoint/call_group (Reported by Jean-Denis Girard) * ASTERISK-28562 - SIP WSS message not processed until next frame arrives (Reported by Robert Sutton) * ASTERISK-27243 - contrib: valgrin...

...L <dialog-info> version number (Reported by Bryan Nelson) * ASTERISK-28605 - chan_dahdi: Deadlock in Hangup Scenarios with concurrent command pri show span X (Reported by Dirk Wendland) * ASTERISK-28633 - stasis bridge topic leak (Reported by Joeran Vinzens) * ASTERISK-28492 - pjsip reload not reloading wizard endpoint/pickup_group endpoint/call_group (Reported by Jean-Denis Girard) * ASTERISK-28562 - SIP WSS message not processed until next frame arrives (Reported by Robert Sutton) * ASTERISK-27243 - contrib: valgrin...

...L <dialog-info> version number (Reported by Bryan Nelson) * ASTERISK-28605 - chan_dahdi: Deadlock in Hangup Scenarios with concurrent command pri show span X (Reported by Dirk Wendland) * ASTERISK-28633 - stasis bridge topic leak (Reported by Joeran Vinzens) * ASTERISK-28492 - pjsip reload not reloading wizard endpoint/pickup_group endpoint/call_group (Reported by Jean-Denis Girard) * ASTERISK-28562 - SIP WSS message not processed until next frame arrives (Reported by Robert Sutton) * ASTERISK-27243 - contrib: valgrin...