similar to: I'm not able to register Softphone(X-lite) in asterisk(Which is installed in EC2 Cloud).

Hi Guenther, What did you recommend to me, I did accordingly but there is no log showing in asterisk CLI. I'm getting same problem. Regards Akhilesh On Mon, Apr 20, 2015 at 6:05 PM, Guenther Boelter <gboelter at gmail.com> wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > On 04/20/2015 12:31 PM, akhilesh chand wrote: > > Hi Folks, > > > >

Hi folks, I'm not able to install asterisk whenever I hit make command I get below error: make[1]: *** No rule to make target `../main/modules.link', needed by `asterisk'. Stop. make: *** [main] Error 2 Regards Akhilesh -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello all, I created a set of Docker images running Asterisk and exposing AMI / ARI ports that i found to be quite useful for ARI / AMI development and regression. As they are based on Docker with whaleware, adding new configuration files to roll your own dialplan / queues / voicemail etc is pretty easy. And you can run quite a lot on the same box to simulate clusters. There is no SIP / RTP

Hi, Sorry if off topic, but is anyone here on this list using it? I am currently searching for a good router for my home network wich supports SIP. Many thanks!

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Guenther Boelter <gboelter at gmail.com> schrieb: > Don't use Port 5061, your SIP-port should be always even like 5060, > 5062, 5064 or 5066. Could you please explain why? I see in /etc/services, that 5060 is the port for SIP and 5061 for SIP-TLS, but I don't find anything for the other ports... Thanks Luca Bertoncello (lucabert

jg <webaccounts173 at jgoettgens.de> schrieb: > It doesn't really depend on your sip.conf and Asterisk. Your gateway/router > will be the major problem. My summer project will be to look at session Are you sure? Right now I'm using an italian SIP-Provider (Messagenet), configured in my sip.conf and I can receive calls without any problem... So, I don't think, I have to

Hi list! Another day, another problem... I'm checking with Nagios my Asterisk at home, and since yesterday I noticed that, after my IP changes (Deutsche Telekom drop the DSL-line every 24 hours, so that I have a new IP every day), the peer of an VoIP-provider I use is UNREACHABLE. Yesterday I though it was a problem on the line, but today is the same, so I think it is something other...

Dear all, Is anyone has experience making Asterisk server with virtual server OPEN-VZ (in proxmox 3.4 box) ? My boss want to build a production server with it, and it will have +/- 300 sip user (concurrent call maybe < 150 call) Is it good to go, or not ? I really hope someone who have experience with it willing to share with me... Thanks in advance... Best Regards, Ikka - Jakarta,

----- 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:

Hi list! Finally I got my Asterisk works with my two phones... It was a problem on my Firewall (for the phone of my wife) and on my Dialplan (for forwarding calls). Now all works as expected, at least in the simulation I did with AsteriskNOW. Hopefully it will work later, when Deutsche Telekom changes my ISDN to VoIP... Well, now I have some time to spend with "fooling"... My phone

Hi Thomas, I followed your recommended command in asterisk CLI which is mentioned in above chain mail but I'm not able capture any log related to softphone. Chain INPUT (policy ACCEPT) target prot opt source destination ACCEPT all -- 0.0.0.0/0 0.0.0.0/0 state RELATED,ESTABLISHED ACCEPT icmp -- 0.0.0.0/0 0.0.0.0/0 ACCEPT all

Hi Thomas, Yes I'm able to access asterisk server but there is no logs capture into log file related to softphone.If you want more information regarding configuration means sip.conf and extension.conf I will share. Regards Akhilesh -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Guenther, Thanks for ur reply I have concern from long time I'm not able to login through softphone with AWS Cloud.Please let me know is there any document or guide line for the same. Regards Akhilesh On Fri, Apr 24, 2015 at 1:26 PM, Guenther Boelter <gboelter at gmail.com> wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > On 04/24/2015 03:34 PM, akhilesh

Hi Thomas, Could you tell how can I change the protocol of corresponding port means 5060 is configured with tcp protocol I want to configured with udp. When I execute nmap -p5060 xx.xx.xx.xx I got below output [root at ip-172-31-32-117 cel]# nmap -p5060 xx.xx.xx.xx Starting Nmap 5.51 ( http://nmap.org ) at 2015-04-21 11:19 UTC Nmap scan report for

Hi Guenther, When I executed nmap -p5060 <xx.xx.xx.xx> I got below output. [root at ip-172-31-32-117 cel]# nmap -p5060 xx.xx.xx.xx Starting Nmap 5.51 ( http://nmap.org ) at 2015-04-21 11:19 UTC Nmap scan report for ec2-xx-xx-xx-xx.us-west-2.compute.amazonaws.com (xx.xx.xx.xx) Host is up (0.00080s latency). PORT STATE SERVICE 5060/tcp filtered sip Nmap done: 1 IP address (1 host

Hi James, Please let me know how could I implement for sip.I will appreciate your help. Regards Akhilesh On Mon, Apr 27, 2015 at 7:01 PM, James Cass <jcass78 at gmail.com> wrote: > Akhilesh, > I have implemented several ec2 instances with both sip and iax2 and have > no problems with xlite or hard phones. Have you already opened the ports in > the vpc security group on the

Dear All, I'm installing the asterisk-1.6.2.24 in Centos 5.3, whenever i'm running following command ./configure I got below error: configure: *** XML documentation will not be available because the 'libxml2' development package is missing. configure: *** Please run the 'configure' script with the '--disable-xmldoc' parameter option configure: *** or install the

HI folks, I have set a barging facility with our production box.Client able to barge a agent but client raise a requirement, they want talk to barge agent but that communication is not listen by customer. It is possible with asterisk or not. thanks in advance. Regards Akhilesh -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear All, I want to disable internal call facility.Means agent(4002) does not make call to agent(4003) or other extensions. Regards Akhilesh -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131016/e1c5734f/attachment.html>

Hi Helvio, Could you tell me what is process to setup an environment for IAX. Regards Akhilesh On Fri, Apr 24, 2015 at 4:25 PM, Helvio Junior <helvio.listas at gmail.com> wrote: > Hi Akhilesh, > > SIP protocol use port 5060 (default) and many other ports to stablish > calls. You need to check if there is AWS firewall rule that allow your > communication from your client