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Hi, I don't know how to call this functionality, but what I want to do is join an already established communication between PSTN---FXS_connected_phone using my SIP phone (I have an asterisk v11 with digium TDM400P at home) Is it possible? What I don't want is using the conference sound and menu.... It's just a normal call between to channels that I have to join for few minutes.

Actually I can start and receive SIP calls (PC client, iphone client) but?I have an issue with calling external number throught PSTN (certified-asterisk-1.8.11-cert2). I'm having this ?error when making a call: *CLI> ? == Using SIP RTP CoS mark 5 ? ? -- Executing [9000 at local:1] Dial("SIP/3000-00000006", "DAHDI/1/4384019357,10") in new stack [Jun 23 16:18:09]

...TDM410P To: Asterisk Users Mailing List - Non-Commercial Discussion ? ? ? ?<asterisk-users at lists.digium.com> Message-ID: ? ? ? ?<CAC8s5NRG34tQgQN+Ff0kuqr2MPcs8WovF7Yghwh_GbZ67u7p+Q at mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 On Sat, Jun 23, 2012 at 10:32 AM, neo haux <neo.haux at gmx.com> wrote: > Actually I can start and receive SIP calls (PC client, iphone client) > but?I have an issue with calling external number throught PSTN > (certified-asterisk-1.8.11-cert2). > > I'm having this ?error when making a call: > > *CLI> ? ==...

Hi, I am using asterisk 11.1.0. How to display the caller number (from asterisk -rvvv terminal) in the first step of the extension (before doing any action) ? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130417/8e6cbb1a/attachment.htm>

Hi, I want to recieve calls to my Skype account and forward them to a SIP/FXS line. I searched for chan_skype for asterisk (v11), but found it only available for asterisk 10 I know that Digium gives no support for this module, but I am sure that someone somewhere did write some tool to allow such connectivity. Do have any idea if I can use Skype with my asterisk v11 ? Thanks --------------

To Jonas: I have an asterisk box at home and I have this line in my rtp.conf file: rtpstart=10000 rtpend=10100 And My FW is setup to forward all incoming ports of range 10000-10100 to the asterisk PC. I've never had a problem since one year, but I have never received more than two simultaneous calls with SIP clients. Message: 5 Date: Fri, 13 Sep 2013 11:49:59 +0200 From: Jonas Kellens

Hi, I have trouble establishing a call between between two SIP phones. One sip phone is, with asterisk server, at home behind a firewall. The second sip phone is an iPhone with 3G wireless connection. When I call from the SIP device at home the SIP account on the Internet (iphone + 3G) I can hear voice on both devices. But when I call from the Internet to my home SIP I get the ring but when I

No tell me that's a jock ! I can't believe it: http://nerdvittles.com/?p=7940 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140409/4e1c87d7/attachment.html>