similar to: Gateway setup

Hi ALL, Am new to Elastix and wanted to try build new modules in the Elastix , so i want to know how the PHP is running ?? as i see no Apache server inside ?? so wanted to know how its running ? which server and architecture? *--* *Thanks & Regards* *Upendra* -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, i am trying to install asterisk newer version on the Elastix machine, but while installing the chan_sip,c module is not building while make. when i see in make menuselect options it showing "XXX" -- extended , please let me know how to enable it and make build chan_sip module. -- Upendra -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, i wanted to know that if i have a message indicator SIP phone , then MWI will work in ELASTIX ?? Let me know the Details of MWI and how test it. *--* *Thanks & Regards* *upendra* -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140804/125551b7/attachment.html>

Hi, i am using elastix 2.3 and created some dahdi extensions,now i dialing between the extensions i.e like 2000 to 2001 , but there is delay of 3 to 4 second before it ring the destination. so cany anyone know how fix it so that after dialing the digits the destination should ring . without any delay after dialing. regards Upendra. -------------- next part -------------- An HTML attachment was

Hi, I am trying to write dial plan for sip to auto answer (auto attend) the incoming call to the sip phone. - If i call from sip1 to sip2 then sip2 should automatically answer the call and play some sound file. I am trying to do this but as new to the asterisk dial plan configuration , so not able Todo this. help me if anyone already done this setup. Regards Upendra. -------------- next part

Hi, In my office have setuped the Elastix machine and i have a static IP (external IP given by ISP), now the issue is that whenerve call from outside sip extensions which is register to the sip server , am not able hear audio from both side. both callee and caller cant hear audio. please help me on this -- Regards Upendra -------------- next part -------------- An HTML attachment was

On 8/9/2016 11:42 AM, Birta Levente wrote: > So, again: > Centos 7 > 2 NICs > enp2s0-192.168.1.12 > enp3s0-192.168.1.13 > default gateway on enp2s0 is 192.168.1.1, defined in > /etc/sysconfig/network > > Which other way (preferred with "ip route") can I add this, but: > #route add default gw 192.168.1.1 dev enp3s0 those are both the same network, and the

On 09/08/2016 23:11, Gordon Messmer wrote: > On 08/09/2016 12:03 PM, John R Pierce wrote: >> those are both the same network, and the default gateway is a global >> thing. packets forwarded to 192.168.1.1 could use either 192.168.1.12 >> or .13, as they are all the same. in reality, they will use the first >> match they find. > > > Generally, but not

On 08/09/2016 12:03 PM, John R Pierce wrote: > those are both the same network, and the default gateway is a global > thing. packets forwarded to 192.168.1.1 could use either 192.168.1.12 > or .13, as they are all the same. in reality, they will use the first > match they find. Generally, but not necessarily. What Birta is trying to accomplish is called

How is this insecure? Most large business and wholesale providers use only IP authentication, relying on a session border controller to do the authentication work resulting in great scalability on the softswitch (since it does not have to act as a proxy as well). If they know your IP, and you know their IP, the only risk is that your IP address can somehow be hijacked. IP authentication is

Hi, every one I'd like to introduce some new feature of our products. mg3000-r fxo gateway provides more feature to work with asterisk. 1.play asterisk ivr with no interuption. when the mg3000-r received call from co line, it wouldn't conect instantly.instead, it start call to asterisk ivr first,when the ivr ready, it connect the co line. this feature make user feel friendly.

Hello, In the last 3 days I setup my SOHO in 2 ways (1) attempt using a retail wifi/router by Netgear. The wifi is not part of this question. WAN (TW Cable modem) | | Netgear (192.168.1.1) ? ? ? \ ? ? ???\ ???_____\______________ ? |? ? eth0? ? ? ? ? ? | ? |? ? ? \? ? ? ? ? ???| ? |? ? ???\--br0--eth2 | ? |? ? ? ? ???|? ? ? ? | ? | C5.5? ???eth1? ? ? | ? |? ? ? ???/? ? ? ? ? | ? |_______

Hello list, I recently bought a Linksys SPA400 as a PSTN gateway. The gateway is connected to an * server and i have 10 users using this setup. I do have some problems in establishing a call to an outside location (call that goes through the SPA400). The first attempt doesn't get through. I suspect the spa400 being the source of the problem. The Linksys SPA400 has a lot of params on the

Hi, I'm a new user of Aterisk, and I have to configure a VoIP Gateway. I have an Alcatel PBX with an E1 card, connected, for the moment, to a local carrier. I would like work with a french VoIP provider, but, for this, I need to use a VoIP Gateway for connect my E1. Thus, I want to build my own voip gateway. It very simple, I want to connect my PBX to the gateway (E1 link) for both call

Hello I'm using a Linksys 3102 as VoIP gateway to connect a POTS line to a PBX. I also have an IP phone in a remote network across the Net. The PBX + gateway, and the phone are both behind a NAT router. I was wondering: 1. When a customer calls us through the POTS line and I pick up the call with the remote IP phone, do RTP packets go directly from the VoIP gateway to the IP phone, or

On 09/08/2016 15:47, Jonathan Billings wrote: > On Tue, Aug 09, 2016 at 10:58:40AM +0300, Levente Birta wrote: >> What I don't understand why the route command allow to add a second default >> gateway with different interface, but the ip route command doesn't? > You can only have one default gateway. > > It sounds to me like you want to use both interfaces, which

I'm setting up a demo for two asterisk machines. One will be a central Asterisk server which will handle everything already in VoIP (office-like functions plus agents functionality). The second Asterisk box will be used strictly as a VoIP gateway to the first server. The gateway server will have 4 T1s connected to it and what I was thinking on doing was the following: in

Hello ALL! 1. eviroment: 1) PBX:Asterisk@HOME(Asterisk 1.2.7.1) 2)Media GateWay:Grandstream HT 488(register to pbx using SIP account,such as 9001),connect to PSTN(phone no is 82820088) using its FXO. 3)Queue:set up a queue named [myqueue],dial 2020 can call queue 2.my question: when customers need help ,they call 82820088, and then call 2020 to enter the queue . From now on ,the

From a 'sip debug': Sip read: SIP/2.0 100 Trying From: "Evert"<sip:[username]@[my ext. IP]>;tag=as6e18534e To: <sip:[dialled number]@[SIP server of VoIP provider]> Call-ID: 6cbf41c25281f08b2e7bbc5043061975@[my ext. IP] CSeq: 102 INVITE Via:SIP/2.0/UDP [my ext. IP]:5060;branch=z9hG4bK4fd1045b Content-Length:0 7 headers, 0 lines Sip read: SIP/2.0 403 Forbidden

Package: xen-utils-common Version: 4.0.0-1 Severity: normal Tags: patch do_ifup() in network-bridge exits badly, if the interface doesn't have a default gateway. Since it's wrapped in xen's locking script it causes it to be retied 100 times and sleep for 95 seconds before it continues. In my setup this amounts to: 16 vlans without a default gateway * 95 secs / bridge = 25 minutes