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I am banging my head over a simple asterisk trick I was seeing on one asterisk server. An extension dials an international premium number, the called number answers, then the extension hangups, but the call continue to run on the international number side, generating an high profit for the premium number company and a big loss for the asterisk owner. I think some sort of "transfer"

This is a great news, thank you. I have open the issue, https://issues.asterisk.org/jira/browse/ASTERISK-26177 and added the relevant files, let me know if you need more info. Leandro 2016-07-06 21:46 GMT+02:00 Joshua Colp <jcolp at digium.com>: > Leandro Dardini wrote: > >> Hello, >> I'd like to know if anyone of you is finding my same problems using any >> recent asterisk version, after 13.7 / 13.8 with chan_sip. >> >> If I use any recent asterisk version, after just few seconds asterisk >> completely locks up,...

What the minimum Server Specifications do I need to run 200 concurrent channels at the time with .WAV recording (MixMonitor)? It will be connected via VOIP sip account. Codec will be ulaw. Which UK dedicated server provider do you recommend and how much bandwidth do I need? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I'd like to know if anyone of you is finding my same problems using any recent asterisk version, after 13.7 / 13.8 with chan_sip. If I use any recent asterisk version, after just few seconds asterisk completely locks up, stopping processing SIP/UDP packets. Nothing is written in the asterisk log, but if I run "netstat -nap | grep 5060" I see the UDP buffer filled up. If I

Hello, I am moving from realtime chan_sip to pjsip and one of the problem I am facing is the lack of "sipregs". With chan_sip, when an extension registers, the server where it has registered to is stored in sipregs. Is there something similar in pjsip? How can I find on which server the pjsip extension has registered to? Leandro -------------- next part -------------- An HTML

...Maybe the client just put the call on hold. > So the call technically has not ended AND the client does not need to send > or handle any RTP data. > Is there any mention of "music on hold" for this channel? > > Greetings > Max > > > ----- Nachricht von Leandro Dardini <ldardini at gmail.com> --------- > Datum: Thu, 15 Sep 2016 18:06:14 +0200 > Von: Leandro Dardini <ldardini at gmail.com> > Antwort an: Asterisk Users Mailing List - Non-Commercial Discussion < > asterisk-users at lists.digium.com> > Betreff: [asteris...

I just discover an "hidden" problem with AEL macro I want to have your feedback. If you use a macro to dial out, like &dialout(${EXTEN}), the leg extension will became ~~~~s~~~~ and if it happens you transfer the call, that will be the callerid appearing on the other phone display. I am just rewriting all the dialplan getting rid of the macro and using gosub, even if asterisk is

Hello, I have an asterisk box with a peer configured with nat=force_rport,comedia, but asterisk keeps sending the audio to the private IP address and ignoring the client peer nat settings. If I check the "sip show peer extension", I see both symmetric RTP and Force Rport are set to yes, but asterisk seems ignoring them. Force rport : Yes Symmetric RTP: Yes Asterisk is behind a

Hello, almost any useful CLI command has an analogue on Asterisk Manager Interface, but I cannot find a way to get the list of subscriptions using AMI. Which is the command, if any? The CLI command is "sip show subscriptions" Leandro -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'd like to dial two extensions (or external number) and ask for confirmation to accept the call. Dialing an extension, asking for confirmation and then dialing a second extension if the call has not been accepted is easy by using the dial option U(...), but if I dial two extensions at once, when the first answers, the other stops ringing. Any idea to make the first continue to ring until

I have some problem finding a smart way to add inbound trunks ip authentication. I don't want to set allowguests=yes Some of my providers just list some IP and I add them like: [provider](!) context=fromoutside type=friend insecure=port,invite disallow=all allow=g729 allow=ulaw allow=alaw canreinvite=no [magrathea1](provider) host=87.238.72.129 [magrathea2](provider) host=87.238.72.130

...handled to my SIP account. When I call the number I receive the correct INCOMING request on Jabber portion of asterisk, but the call is not connected to the gtalk part. JABBER: asterisk INCOMING: <iq from="+ 17174695631 at voice.google.com/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4" to=" ldardini at gmail.com/asterisk438D86E0" id="jingle:10.176.108.16-15899749:1:457BCF36" type="set"><ses:session type="initiate" id="SIP784359174 at 10.177.37.1" initiator="+ 17174695631 at voice.google.com/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4" xmlns:ses=...

Hi, On 64 bit centos 5.6 I have virtualbox 4 and 64 bit elastix latest. Since yesterday cpu utilization has been constantly peaking 65-75%. Hardly 1-2 concurrent calls. No other activity on server. Top shows asterisk on top. Its quad xeon server with 4 gb ram. Any suggestion where should I start and how should I go about with my investigation. Thank you and have a great weekend. Sans

Hi list someone is trying to hack my server . Is there any way by whcih I can stop hacking of my server except iptables ? I want to stop on the basis of sip.conf account only. bcoz I can't apply iptables rules on server it's remote server of server provider and we used it for making voip call for customers. for the time been i have close all sip accounts. but can't stop for more then

Hello, I was wondering if there is a tool that can create the realtime database structure for latest Asterisk version or a web resource/file containing the sql scripts. Hope I haven't missed obvious things, I had no luck searching on the web, in the wiki I found few pages with bits of sql or table structures, like:

You can use "Command" command, and "sip show subscriptions" as a parameter -- Alex Epshteyn email: alex at thirdlane.com web: www.thirdlane.com phone +1 415.261.6601 ----- Original Message ----- > From: "Leandro Dardini" <ldardini at gmail.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> > Sent: Thursday, January 15, 2015 3:00:30 PM > Subject: [asterisk-users] Showing sip subscriptions in Manager > > > > Hel...

Hello, while most of the physical phones have keys to handle attended and blind transfer, most soft phones have no support for it. Asterisk offers a "featuremap" to assign a key to blindxfer and atxfer and they work fine if the call is still in the same starting context, but if the call has moved in another context, then the new call will be started from such context with unpredictable

Followme is perfect to handle FMFM and it is now also realtime, but it seems impossible to assign some value to a variable, from within the followme to store info for example about the tenant the followme is running under, like instead happen for example in the queue with the setinterfacevar field. I just need to pass a variable from the channel placing the call to the followme to the channel

...yrev) * ASTERISK-29001 - chan_pjsip does not process or forward 181 responses (Reported by Torrey Searle) * ASTERISK-27273 - app_voicemail: When a voicemail is marked as "Urgent", it is not sent by email/processed by the mailcmd command (Reported by Leandro Dardini) * ASTERISK-29033 - res_pjsip_session: Aggressively terminates session on failed re-INVITE (Reported by Joshua C. Colp) * ASTERISK-28974 - res_rtp_asterisk: T.140 messages have appended RTP string to each message block. (Reported by Thomas Johnson) Improvements made...

...rey Searle) * ASTERISK-29034 - Lastpause of realtime members is reseting (Reported by Evandro C��sar Arruda) * ASTERISK-27273 - app_voicemail: When a voicemail is marked as "Urgent", it is not sent by email/processed by the mailcmd command (Reported by Leandro Dardini) * ASTERISK-29033 - res_pjsip_session: Aggressively terminates session on failed re-INVITE (Reported by Joshua C. Colp) * ASTERISK-28974 - res_rtp_asterisk: T.140 messages have appended RTP string to each message block. (Reported by Thomas Johnson) For a full list o...