similar to: OpenVPN Clients Intermittently Cannot Call In

----- Original Message ----- > From: "Administrator TOOTAI" <admin at tootai.net> > To: asterisk-users at lists.digium.com > Sent: Thursday, April 30, 2015 4:43:33 PM > Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In > > > I am running Asterisk 11.12.0 on CentOS 6.4. The asterisk server and > > internal phones are located on

----- Original Message ----- > From: "Administrator TOOTAI" <admin at tootai.net> > To: asterisk-users at lists.digium.com > Sent: Friday, May 1, 2015 6:42:38 AM > Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In > > Le 01/05/2015 00:05, Andrew Martin a ?crit : > > ----- Original Message ----- > >> From:

Le 01/05/2015 00:05, Andrew Martin a ?crit : > ----- Original Message ----- >> From: "Administrator TOOTAI" <admin at tootai.net> >> To: asterisk-users at lists.digium.com >> Sent: Thursday, April 30, 2015 4:43:33 PM >> Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In >> >>> I am running Asterisk 11.12.0 on CentOS

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On 05/05/2015 10:59 AM, Andrew Martin wrote: > > > ----- Original Message ----- >> From: "Administrator TOOTAI" <admin at tootai.net> To: >> asterisk-users at lists.digium.com Sent: Friday, May 1, 2015 6:42:38 >> AM Subject: Re: [asterisk-users] OpenVPN Clients Intermittently >> Cannot Call In

Le 30/04/2015 19:18, Andrew Martin a ?crit : > Hello, Hello > > I am running Asterisk 11.12.0 on CentOS 6.4. The asterisk server and internal phones are located on the 10.10.32.0/21 LAN subnet. I have many internal SIP phones, which appear to be working correctly. I have a few external phones (Yealink SIP-T32G or other Yealink model) on 192.168.32.0/24 which have an OpenVPN client

Hello, using Asterisk 1.8.12.2 case : I call with my cellphone to our public telephone number Our receptionist answers the incoming call and does an attended transfer to my colleague ( A ) Colleague answers and the receptionist tells him that I am on the other side. Receptionist transfers the call and I am connected to my colleague ( B ) My question is about the CallerID that the

Hello, On a Asterisk 11.12.0, I'm studying BLF behaviour with Yealink phones. My ultimate goal is to present Operator the name and number of every incoming call so that he/she can if it's worth to pickup a ringing incoming call. I've discovered notifycid option in sip.conf. When a call comes in, I can see that Asterisk is sending two successive NOTIFY messages while the target is

Andrew Martin wrote: > ----- Original Message ----- >> From: "Joshua Colp"<jcolp at digium.com> >> To: "Asterisk Users Mailing List - Non-Commercial Discussion"<asterisk-users at lists.digium.com> >> Sent: Monday, May 11, 2015 12:32:06 PM >> Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped calls after 32

Hi Everyone, I am using pfSense to do firewall and NAT on an Asterisk server. I have ports 5060 TCP/UDP and 10k-20k UDP forwarded to the Asterisk server local IP 192.168.5.5. However, when a user from outside using Linksys WRP400 ata connects to the Asterisk server and registers I see them as 192.168.1.1 in the "sip show peers" command. In face, all many different of the Linksys WRP400

----- Original Message ----- > From: "Andrew Martin" <amartin at xes-inc.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> > Sent: Monday, May 11, 2015 1:35:07 PM > Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped calls after 32 seconds > > > That should

Ok, I have a 7960 that's plugged into my laptop. my home network is wireless so I don't have a switch anywhere to plug the phone into directly. I'm running siproxd on my OS X laptop and I can make outbound calls from the 7960 fine (I guess I don't have the phone configured to register inbound calls via SIP), but the phone isn't registering to the asterisk box via siproxd

Hello, I am running Asterisk 11 on CentOS 6.4 with SIP clients (Yealink phones). All the SIP clients are on a LAN, so no NAT is involved. I have been experiencing an intermittent problem where a call will be successfully answered, but then dropped by Asterisk 32 seconds after it is answered (with a "Retransmission timeout reached on transmission" error). Here is an example of this

I've just started deploying these (well the T28P model) after years of Snom issues and they look pretty good (although the documentation is execrable; if you thought the Snom stuff was obtuse Yealink have got them knocked into a cocked hat!). Anyway, for provisioning I use HTTP with a DHCP entry like:- # # Yealink Phones # group { #

Hello all, After numerous issues with Snom phones (360/370/870) potentially looking to migrate too Yealink as their product range looks very promising indeed. Are any of you using them with Asterisk ? How do they perform ? Do you use mass deployment at all ? Would be very interested to hear from you. -- Thanks, Phil -------------- next part -------------- An HTML attachment was

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi List, I have come across the above handset a few times in the UK, They are quite cheap over here (~?80) Not the best handset in the world but works well enough. I have been asked to setup a central config server for a large collection of these handsets. I know they can do Auto provisioning via FTP/HTTP/TFTP I have got an example of the generic

Hello, I have Snom, Cisco, Grandstream & YeaLink phones. Is there a way to push a centralized phone book to these phones ?? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101202/ea25cd7e/attachment.htm

I have two problems. The first is that when I dial my number from a phone on the Internet or any phone outside my LAN, Asterisk does not respond in any way, which means somehow my system is not picking up the fact that there's an incoming call to it. The second problem is that I thought I'd try an internal phone to see if I could get the hello-world stuff working at the least. I

I have a setup with Yealink phones & Asterisk Server (all latest patches). I am using BLF to display the states of other phones. While this works MOST of the time (busy, being called) it does NOT work when a phone is NOT regisstered at all, the yealink phones display a green dot EVEN if a phone is turned off (try explain this to users, they are shaking their heads!!!) I can see on the

Hi, Sorry for an OT post but striking out a bit at the moment trying to get a response from Yealink R&D. Has anybody successfully managed to get a Yealink phone to work across Open-VPN when using tlsauth ? We really do hope that it is possible due to the benefits tlsauth offers against DoS. -- Thanks, Phil -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, We have a user who is putting large call volumes through Asterisk, and wants to BLF monitor up to 90 extensions. We are struggling to find a handset that can keep up with Asterisk :) 1) Is there a handset that will do this? 2) Is there a different (standard) way to send BLF and allow directed pickups? 2a) Or even a handset specific way? Asterisk handles the BLF volume fine, even on quite