similar to: Suggestion of Server Specifications for Asterisk

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"

Hello, I have a Google Voice phone number and want to connect it to my asterisk box to have calls 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="

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 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:

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

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

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

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

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

Hello, Can someone recommend me where is best place to find Asterisk Expert/Consultant for freelance work? If you are interested to work as a freelancer, you can email me directly. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150902/5a99cbfe/attachment.html>

Hi I came across couple of pointers on the Internet regarding solutions available for providing hosted PBX service. 1. Multiple PBXs: Using separate hardware to host each PBX. Pretty straightforward, but no hosting company wants to use it. 2. Multi-tenant PBX: Configuring multiple PBXs within the same instance of Asterisk. I.e. partitioning a single instance of Asterisk into multiple PBXs by way

Hello, I was trying to use a queue in linear order and to provide the exact order of members to dial by adjusting the uniqueid value. Obviously it doesn't work and it seems an old problem: https://issues.asterisk.org/jira/browse/ASTERISK-18480 Realtime configuration can't identify "orders" in the list of results, so the members for the queue are returned in random order.

Hello friends, when a call arrives in the queue, a CDR record is created, but there is no info about which agent has picked up the call. I can find that info only in queue_log. Is there a way to have that info in the CDR or maybe in a variable in the "h" context, when the call is ended? Leandro -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, have you noticed the message num (VM_MSGNUM) is off by one? For example, I receive the following message: "Just wanted to let you know you were just left a 0:03 long message (number 7)" but in attach there is the msg0006.wav Leandro -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, do you know if it is possible to define the SLA configuration in the database for realtime usage with asterisk? Leandro -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20141114/7c4f09a4/attachment.html>

Hello, I have a recording started in the dialplan with the MixMonitor application. I want to be able to stop it during a call and maybe restart it. I tried using the value defined in [featuremap] but it starts another MixMonitor application even if there already one instead of stopping it. Any idea on how I can stop the MixMonitor application while it is running? [featuremap] automixmon =>

When you set sendrpid=yes in sip.conf, a very nice feature is activated. When dialing an extension, the callerid of the dialed extension is returned back on the display of the calling phone. So if you call extension 100, you can see you are calling Ann (for example). I want to selectively disable the transmission of this information back to the caller. How can I do it? I tried setting

Hello, I am migrating my dialplan from asterisk 11 to asterisk 12 and it seems the ${CDR(start)} is not returning any data. Other functions, like ${CDR(duration)} or ${CDR(src)} or ${CDR(accountcode)} are returning correct values. Where is my mistake? Has this function being renamed? Leandro -------------- next part -------------- An HTML attachment was scrubbed... URL:

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