search for: davaosoft

...d run asterisk -rvvvvvv. > > Then let your Phone try to register. Asterisk should show you what's > getting wrong. > > If you can't see anything while trying to register, shutdown your > firewall and try it again ... > > Regards > Guenther > > > - -- > DavaoSOFT, the home of ERPel > ERPel, das deutsche Warenwirtschaftssystem fuer LINUX > http://www.davaosoft.com > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v2 > > iQIcBAEBAgAGBQJVNPKpAAoJENexF5oIz3BCP84QAOOwrZRJn27PuGqVYT643pqZ > oUSIscdJfrDBcWImIRRWNZc1nbKN8hYpXfVgAvdWYaOzcdqeMS95a...

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 Folks, I'm trying to register softphone(X-lite) but I'm not able to register softphone whenever I'm trying to register softphone I got below error [image: Inline image 1] Is there any document/guide line where I will get process to register softphone in asterisk(Which is installed in EC2 Cloud). Regards Akhilesh -------------- next part -------------- An HTML attachment was

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:

...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 > > Does `../main/modules.link' exist after running ./configure? - -- DavaoSOFT, the home of ERPel ERPel, das deutsche Warenwirtschaftssystem fuer LINUX http://www.davaosoft.com -----BEGIN PGP SIGNATURE----- Version: GnuPG v2 iQIcBAEBAgAGBQJVK4bAAAoJENexF5oIz3BC4gsQAMDXbhvpvDlv7pTKcspO0VGF iJkLqRHUsEsrLtNg8WWOT1AvwO5aZE/lrVgY/xA0WYjVdfDwXZK3vT2cgyt1JvDf dyuw9DzHKjeQNoa+Hqxfdn...

...it to complicated ... Connect to your Asterisk via ssh and run asterisk -rvvvvvv. Then let your Phone try to register. Asterisk should show you what's getting wrong. If you can't see anything while trying to register, shutdown your firewall and try it again ... Regards Guenther - -- DavaoSOFT, the home of ERPel ERPel, das deutsche Warenwirtschaftssystem fuer LINUX http://www.davaosoft.com -----BEGIN PGP SIGNATURE----- Version: GnuPG v2 iQIcBAEBAgAGBQJVNPKpAAoJENexF5oIz3BCP84QAOOwrZRJn27PuGqVYT643pqZ oUSIscdJfrDBcWImIRRWNZc1nbKN8hYpXfVgAvdWYaOzcdqeMS95aLiEN6sX+zni A2aAOMajt5bajam5MYRR6y...

...api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every > Thurs: http://www.asterisk.org/hello > > asterisk-users mailing list To UNSUBSCRIBE or update options > visit: http://lists.digium.com/mailman/listinfo/asterisk-users > > > > - -- DavaoSOFT, the home of ERPel ERPel, das deutsche Warenwirtschaftssystem fuer LINUX http://www.davaosoft.com -----BEGIN PGP SIGNATURE----- Version: GnuPG v2 iQIcBAEBAgAGBQJVNh1pAAoJENexF5oIz3BCeTMQAO50VIcKKQndKmiJEevo+Uzu 4kxTFPg6vLMfV6yOIIC86oxlzqJW6qkuLpt2xb7jr3ppg5VMH88Ozhqpu3E0hZNk VYxXTafWFgHgJLSblxG1Zq...

...o SIP / RTP configured at the moment. > > See > https://github.com/l3nz/whaleware/blob/master/examples/asterisk-load-t est/README.md > > Maybe somebody else might find them useful. There is Asterisk 1.8, > 11, 12 and 13. Thanks Great, will try it out tonight ... Thanks - -- DavaoSOFT, the home of ERPel ERPel, das deutsche Warenwirtschaftssystem fuer LINUX http://www.davaosoft.com -----BEGIN PGP SIGNATURE----- Version: GnuPG v2 iQIcBAEBAgAGBQJVOJVpAAoJENexF5oIz3BC08oQAJMOx64PLaVQVtJYDyZH78Sl V134Rgv/Fq/1Udm4dZlt4Dooo3cEYPCd+WBC70VU+4hNegSgv6xZxGSMXyymPT/R SrjL/4jMJ/9S8uanIRdiZb...

...Hi list! > > 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... Don't worry, Asterisk works very well with Deutsche Telekom and there new ip-based connections ... - -- DavaoSOFT, the home of ERPel ERPel, das deutsche Warenwirtschaftssystem fuer LINUX http://www.davaosoft.com -----BEGIN PGP SIGNATURE----- Version: GnuPG v2 iQEcBAEBCAAGBQJVasmmAAoJEG6MZewV4LQAPf4H/iN9V52VMz1OMb6QI4WHC125 G1erQWVdUssq65/+/2tnQ7kQZwt56HhdlbieBZcLK0BpO0xnSVFZ2AGd0nzbWn9h 4NR8E6PRYfODXh/vs5INoN...

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!

...sip_reg_timeout: > -- Registration for '5406013068 at messagenet' timed out, trying again > (Attempt #1218) > > Could someone help me to understand the problem? Don't use Port 5061, your SIP-port should be always even like 5060, 5062, 5064 or 5066. Regards Guenther - -- DavaoSOFT, the home of ERPel ERPel, das deutsche Warenwirtschaftssystem fuer LINUX http://www.davaosoft.com -----BEGIN PGP SIGNATURE----- Version: GnuPG v2 iQEcBAEBCAAGBQJVfPbeAAoJEG6MZewV4LQA1i8H/jX0eimgTfWYzdz0yu91eete MYKIzFByiNlCsBn6boLfcPw/4q4cTOGTGPEfpEh4sqv8B5qh6rR8BnBID2KG53Vr gMieTlEgMLoKEkOJvnnir3...

-----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

...ert.de) > Hi Luca, that's what Deutsche Telekom has told me. I'm using it since a few month in Germany and it works well. It depends from which ports are supported by your provider. Sipgate for example is telling there customers to use 5060, 5160, 5260 etc. Regards Guenther - -- DavaoSOFT, the home of ERPel ERPel, das deutsche Warenwirtschaftssystem fuer LINUX http://www.davaosoft.com -----BEGIN PGP SIGNATURE----- Version: GnuPG v2 iQEcBAEBCAAGBQJVfSTGAAoJEG6MZewV4LQAYNUH/0D+bpzoQPLWqLytBoWljiV5 RLgybyifGvaGBr7z50rzC8ZoAdvl0oqglIRyviojTJKCR+1Mr9JJpCGJwjXPP3SB v3wntGA2V1axCpjeGVkgOT...

...e columns, but the > high percentage of loss for received packets for 192.168.32.26 > seems suspicious. Do these statistics indicate a problem? > > Thanks, > > Andrew Hi Andrew, is this a linux machine? If so, check your NIC with ifconfig for hardware errors. Guenther - -- DavaoSOFT, the home of ERPel ERPel, das deutsche Warenwirtschaftssystem fuer LINUX http://www.davaosoft.com -----BEGIN PGP SIGNATURE----- Version: GnuPG v2 iQIcBAEBAgAGBQJVSF22AAoJENexF5oIz3BC7SUQAL7guaLv8rKHLfJah58/qhT7 qWiyYcjiFiLOUC1J6tgZ+BpT1tXGs5A5NAx+0yC3QWoDHEb/dAg+tzy9YqWqfrtz sePuqAHYPivqtqve1WBM3c...

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

...for a live introductory webinar every > > Thurs: http://www.asterisk.org/hello > > > > asterisk-users mailing list To UNSUBSCRIBE or update options > > visit: http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > - -- > DavaoSOFT, the home of ERPel > ERPel, das deutsche Warenwirtschaftssystem fuer LINUX > http://www.davaosoft.com > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v2 > > iQIcBAEBAgAGBQJVNh1pAAoJENexF5oIz3BCeTMQAO50VIcKKQndKmiJEevo+Uzu > 4kxTFPg6vLMfV6yOIIC86oxlzqJW6qkuLpt2xb7jr3ppg5VMH88Oz...