search for: 3cx

Hi, I created a extension in Asterisk, the extension has been configured in Android softphone 3cx. When I tried to call from Andorid phone to some other IP extension which is registered in Asterisk, I am not able to hear the voice, when I check the asterisk log or wireshark there is only one way RTP traffic, from Android I am connecting to Asterisk via 2G GSM network. Any idea would be appreci...

Hi all, Yes, this is not the right list for such a question and I am using Asterisk myself its for a friend who isn't used to Linux. You can write me off list if you want. He is looking for a Windows based PBX with same functionality as Asterisk. Any tips? Many thanks!

Hi Folks, I'm trying to register softphone(3CX Phone) in AWS Cloud but I'm not able to register I got below screen. [image: Inline image 1] Register Screen for 3CX Phone [image: Inline image 1] Regards Akhilesh -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/aster...

Hello, everyone I'm relatively new to Asterisk (and VOIP in general), but I have a project that it will really help with. So, I setup a test system on an ancient 400MHz P3 we had lying around. It worked great. I had a test dialplan working, and had no trouble connecting to it with SIP using 3CX SoftPhone over our LAN (and over the Net through our NAT). So, we went ahead and bought a server to put up on our colo site with our web apps and database. We put together a 2 proc, dual-core AMD system with 8GB memory, plenty of disk space (>2TB) 3 GB NICs etc. and setup Asterisk 1.4.13 (from...

********************************************************************** Este email assim como os ficheiros que possa ter em anexo s?o confidenciais e para uso exclusivo da pessoa ou organiza??o para o qual foi enviado. Se recebeu este email por engano por favor notifique Redes@bnc.pt Esta nota confirma que esta mensagem foi verificada pelo MIMEsweeper n?o tendo sido encontrados virus.

Hi, Thank for your answer. 22.04.2019 23:47, Joshua C. Colp пишет: > On Mon, Apr 22, 2019, at 1:43 PM, Pavel wrote: >> Hi, >> >> Got problems with incoming SIP calls. >> >> Scenario: >> >> Server1: 3cx or any other server >> >> Server2: Asterisk 16.2.1 . PJPROJECT 2.8 >> >> Server2 registers on Server1 with SIP ID 1121. >> >> Registration is OK. >> >> Server2 outgoing calls are OK. >> >> INVITE, unauthorized, INVITE with password, OK, RI...

Hi, Got problems with incoming SIP calls. Scenario: Server1: 3cx or any other server Server2: Asterisk 16.2.1 . PJPROJECT 2.8 Server2 registers on Server1 with SIP ID 1121. Registration is OK. Server2 outgoing calls are OK. INVITE, unauthorized, INVITE with password, OK, RINGING,... Troubles with incoming calls / incoming INVITE's . I can not identify...

I have clarification in which how we can enable direct dial when we press numbers in 3cx phone on hook. Now its like we have to use dial button to dial. Previously I am able to dial directly after entering number. Now its not working. Can someone help me on it. Is this a setup that we have to do in freepbx or In 3cx Phones? Regards Darin Egocentrix -------------- next part ----------...

...derstand what a SIP invite is. Certainly it's explained as: "This article explains the main fields included in a SIP INVITE, which is sent to set-up a VoIP call. A SIP INVITE message contains typically between 4 and 6 header entries with contact information inside them." http://www.3cx.com/blog/voip-howto/sip-invite-header-fields/ The article enumerates the headers and explains them. But what sends the invite? Asterisk? A soft-phone? I found sample config's to send an invite with Asterisk but no other method was given. Can only Asterisk send an invite? Why? The artic...

Hi, I have written an click2dial application that rings an agent soft phone and connects the agent with a customer. very often I can see, that the agent softphones signal a busy back to server, although the phone is definitely hung up and the previous calls where handled normally. I testet 3cx Version 6 and X-Lite V1 up to V3... all show the same misbehaviour. I did a SIP Trace and can see, the phone replies with a busy on the invite... but I don?t know why. Has anybody experienced similar things and knows a reason or a workaround for this? I am working on a asterisk 11.7 using sip-re...

Hi all, I'm having trouble with voicemail on my *NOW 1.5/FreePBX box. I have enabled voicemail in the extensions area, and set the default password. However, every time I try to log in with a mailbox and password, I get the message "login incorrect". I've tried changing the voicemail password, and also disabling and re-enabling the voicemail feature. What else can I do to set up

...what reinvite does. We are running Asterisk 1.6.1.1 on CentOS 5.3. SIP is set to canreinvite=nonat. We have tried RTP with strictrtp set to both yes and no. We have also tried extending the Asterisk rtp port range to accommodate the differing default ranges of the soft phones (Twinkle on Linux, 3CX on Windows). Testing revealed no problems when the soft phones we used for testing were on the same physical and logical network. Once we moved the soft phones to OpenVPN connections (same logical network but different physical media), the call is setup, the receiver hears the caller for the brie...

Hi, I have an Asterisk box behind a NAT address and also a Xlite 4 soft phone behind a different NAT network. Asterisk -> Nat -> Internet -> Nat -> Softphone. I can register my softphone to the asterisk box ok via SIP but the RTP stream from the asterisk box is addressed to the private non-routeable address of the softphone when I turn on rtp debuging. How can I configure the rtp

We have a couple of cisco SPA phones and 3CX softphones behind a NAT firewall in a remote location. Firewall is connected to a bridged router which connects them to the public internet. Router 5.6.7.8 Firewall 5.6.7.9 (gateway 5.6.7.8) Cisco SPA phone 192.168.1.4 Softphone 192.168.1.5 When these phones try to register, this is...

Hello, can someone recommend a good and free Softphone for Windows which does not display advertisments inside the program? We have X-Lite but free version display advertisments. thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140608/c3a9e887/attachment.html>

Hi, I`m not actually asking for a comparaison between the two, I have one on hand and will make up my own mind. But I can't find much info on whether the Snom (370 to be exact) accepts FTP provisioning like the Polycom (but few others) do. Any Snom user can answer this one for me? Mike -------------- next part -------------- An HTML attachment was scrubbed... URL:

[please keep the list CC'd. Thanks!] On Wed, Aug 13, 2014 at 9:33 AM, Marcello Vezzelli <mv at 3cx.com> wrote: > I noticed that 2 of Sweex UPS have an identical serial number 20100813, > which looks like a date. > The other one has an empty serial number. > > The blew one had the serial number, so I'm left with two different UPSes > from the serial number point of view....

You might check your phones as well. We had this problem early on with a softphone and it was a setting in the phone that was set to hang up after 30 seconds of inactivity "in case of network disruption". For some reason it was detecting "network disruption" in every call even when the calls were proceeding normally. Unchecking this box solved the problem. It may not be

Digium is 100% lost in the map. If they would come up with a Paid version of Asterisk, one that would use the .NET framework in Windows, something simple to install, they could go public on the product. Linux has a very steep learning curve. A Windows application that would do exactly the same would be a home run. Note: I am a Linux expert user, but it took me years to get here. And still, moving

...bKd16b1eb7 CSeq: 110 REGISTER WWW-Authenticate: Digest realm="asterisk",nonce="1424762038/41d5874af9ea9408c257949c309c8aa0",opaque="7f15d8c2312c7b0d",algorithm=md5,qop="auth" Content-Length: 0 username and password are correct, this phone was working with 3CX just fine but won?t work with asterisk for some reason. ( any idea what may cause the problem? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150224/7be9231e/attachment.html>