similar to: 302 Moved Temporarily and channel variable

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"

No, there is no Music On Hold starting and the bad thing is the call duration reported by asterisk was just few seconds while the call duration reported by the provider was few thousand seconds, the max allowed. So they will be able to terminate the call on the asterisk side and have it run on the provider side. Leandro 2016-09-15 19:18 GMT+02:00 Max Grobecker <max.grobecker at

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:

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

Hi, I want to check the status of a blind transfer (only sip endpoint) between various phones. Transfer is working perfectly, using ## from features.conf or using transfer key from phone, here SNOM320. My problem is that if party to transfer to is busy, the transfer fail and the call is ended. What I want to do is to return the call to the party who originate the transfer. I checked

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

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

Hi, I have the following problem that when asterisk receives SIP response 302 it cannot forward the call I get such debug: [Jan 4 10:43:27] WARNING[18671]: channel.c:3281 ast_request: No channel type registered for 'Local' [Jan 4 10:43:27] NOTICE[18671]: app_dial.c:505 wait_for_answer: Unable to create local channel for call forward to 'Local/poczta at routing-sip' (cause = 66)

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, 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

Hi, When a SIP hardphone is transfering a call while ringing (caller and callee don't speak to each other) using phone's Transfer key, it seems BLINDTRANSFER remains empty. Though I can see a 302 MOVED TEMPORARILY message coming in. Is there a work around or something obvious I'm missing (it's the first time I'm playing with Dialplan transfert features. context mylocal {

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

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>

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:

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: