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...AEI&clientId=3")} executes and get answer from the server [{"RequestedCount":0,"MissedCount":7,"Total":7}] i dont want it to be executed Thanks list for your help -- Best regards Antony tel. +380669197533 tel2. +380636564340 Paypal http://paypal.me/Satskiy <http://paypal.me/Satskiy?ppid=PPC000654&cnac=PL&rsta=en_PL(en_DK)&cust=NN8XJS9XEP22C&unptid=21db79ac-ef8d-11e5-9553-9c8e992ea258&t=&cal=4d776c21ca7d2&calc=4d776c21ca7d2&calf=4d776c21ca7d2&unp_tpcid=ppme-social-business-profile-created&page=main:email&...

...ll question i use call-limit=1 on peers but call limit is not working if user is not registered on PBX and making calls so the main question is -- how to Disallow CALLS without registering on PBX -- Best regards Antony tel. +380669197533 tel2. +380636564340 Paypal http://paypal.me/Satskiy <http://paypal.me/Satskiy?ppid=PPC000654&cnac=PL&rsta=en_PL(en_DK)&cust=NN8XJS9XEP22C&unptid=21db79ac-ef8d-11e5-9553-9c8e992ea258&t=&cal=4d776c21ca7d2&calc=4d776c21ca7d2&calf=4d776c21ca7d2&unp_tpcid=ppme-social-business-profile-created&page=main:email&...

> On 11/02/2017, at 3:40 am, Frank Vanoni <mailinglist at linuxista.com> wrote: > > On Thu, 2017-02-09 at 14:58 +0200, ????? ?????? wrote: > > >> so the main question is -- how to Disallow CALLS without registering >> on PBX > > sip.conf configuration > In the [general] section, define: > > > [general] > ... > allowguest=no >

Hi. I was studying about how to use databases in Asterisk, accessing it from the dial plan. In my project, my dial plan will have to store simple data (ex: IP number, port number, device name, etc) in a persistent way, so that it will be possible to retrieve such information in future moments, still via dial plan. For this case, I would like to know? 1. What is the best choice for storing and

Hello, fail2ban does not ban offending IP. NOTICE[29784] chan_sip.c: Registration from '"user3"<sip:1005 at asterisk-ip:5060>' failed for 'offending-IP:53417' - Wrong password NOTICE[29784] chan_sip.c: Registration from '"user3"<sip:1005 at asterisk-ip:5060>' failed for ?offending-IP:53911' - Wrong password systemctl status

...ccount2 host=sip.myitsp.com If sip.myitsp.com directs a call to asterisk with a request line of: INVITE line1 at mybindaddr SIP/2.0 then it is matched to the line2 peer whereas it would probably be better matched to the line1 peer -- Best regards Antony ??? (066) 919-75-33 ??? (063) 656-43-40 satskiy.a at gmail.com <mail%3Asatskiy.a at gmail.com> -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150702/df108adc/attachment-0001.html>

...[4294](unistim-phones,group3) device=0016caf460f5 line=> 4294 callerid="Victoriya Mukan" <4294> [4211](unistim-phones,group4) device=000ae475faed line=> 4211 callerid="Gomenyuk tatyana" <4211> -- Best regards Antony ??? (066) 919-75-33 ??? (063) 656-43-40 satskiy.a at gmail.com <mail%3Asatskiy.a at gmail.com> -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150706/b50436d4/attachment.html>

...NS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE* *Supported: replaces, timer* *WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="63fdf36b"* *Content-Length: 0* *<------------>* -- Best regards Antony ??? (066) 919-75-33 ??? (063) 656-43-40 satskiy.a at gmail.com <mail%3Asatskiy.a at gmail.com> -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150306/2181cab3/attachment.html>

...Inc. | Director of Technology > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at: http://digium.com & http://asterisk.org > > > > ------------------------------ > > Message: 2 > Date: Tue, 28 Apr 2015 17:19:46 +0300 > From: ????? ?????? <satskiy.a at gmail.com> > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Subject: [asterisk-users] hi list need your help > Message-ID: > <CAFgS45v=t-qkfkTypJhj5YijWOh+D5pQY2JXF3w8YN9iR+B5mg at mail.gmail.com> > Conten...

...-28847 - ARI channels cuts the endpoint string over 80 characters (Reported by sungtae kim) * ASTERISK-28835 - IPv6 addresses in SDP incorrectly formatted (Reported by Daniel Heckl) * ASTERISK-28372 - Asterisk REPLY Wrong Contact header port (TCP) (Reported by Anton Satskiy) * ASTERISK-24428 - Document that Asterisk will use the default SIP ports (5060 for TCP, 5061 for TLS) if the extern option variants aren't used (Reported by sstream) * ASTERISK-28838 - AST_MODULE_INFO requires, MODULEINFO does not mention (Reported by Alexander...

...-28847 - ARI channels cuts the endpoint string over 80 characters (Reported by sungtae kim) * ASTERISK-28835 - IPv6 addresses in SDP incorrectly formatted (Reported by Daniel Heckl) * ASTERISK-28372 - Asterisk REPLY Wrong Contact header port (TCP) (Reported by Anton Satskiy) * ASTERISK-24428 - Document that Asterisk will use the default SIP ports (5060 for TCP, 5061 for TLS) if the extern option variants aren't used (Reported by sstream) * ASTERISK-28838 - AST_MODULE_INFO requires, MODULEINFO does not mention (Reported by Alexander...

...C8:04:1F:DC:FE:B7:56:27:26:FF:18:CD:BB:71:99:B8:97:F9:81:2B:08:74:72:67:3B:A9:88:5F:00:34 a=sendrecv thats why i got Failed to set remote offer sdp: Called with SDP without ice-ufrag and ice-pwd Waiting for your advice ---thanks -- Best regards Antony ??? (066) 919-75-33 ??? (063) 656-43-40 satskiy.a at gmail.com <mail%3Asatskiy.a at gmail.com> -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150428/4f152d16/attachment.html>

...ev) * ASTERISK-28839 - Sporadic crashes with Segmentation fault (Reported by Joeran Vinzens) * ASTERISK-28835 - IPv6 addresses in SDP incorrectly formatted (Reported by Daniel Heckl) * ASTERISK-28372 - Asterisk REPLY Wrong Contact header port (TCP) (Reported by Anton Satskiy) * ASTERISK-24428 - Document that Asterisk will use the default SIP ports (5060 for TCP, 5061 for TLS) if the extern option variants aren't used (Reported by sstream) * ASTERISK-28838 - AST_MODULE_INFO requires, MODULEINFO does not mention (Reported by Alexander...

...ev) * ASTERISK-28839 - Sporadic crashes with Segmentation fault (Reported by Joeran Vinzens) * ASTERISK-28835 - IPv6 addresses in SDP incorrectly formatted (Reported by Daniel Heckl) * ASTERISK-28372 - Asterisk REPLY Wrong Contact header port (TCP) (Reported by Anton Satskiy) * ASTERISK-24428 - Document that Asterisk will use the default SIP ports (5060 for TCP, 5061 for TLS) if the extern option variants aren't used (Reported by sstream) * ASTERISK-28838 - AST_MODULE_INFO requires, MODULEINFO does not mention (Reported by Alexander...

...ev) * ASTERISK-28839 - Sporadic crashes with Segmentation fault (Reported by Joeran Vinzens) * ASTERISK-28835 - IPv6 addresses in SDP incorrectly formatted (Reported by Daniel Heckl) * ASTERISK-28372 - Asterisk REPLY Wrong Contact header port (TCP) (Reported by Anton Satskiy) * ASTERISK-24428 - Document that Asterisk will use the default SIP ports (5060 for TCP, 5061 for TLS) if the extern option variants aren't used (Reported by sstream) * ASTERISK-28838 - AST_MODULE_INFO requires, MODULEINFO does not mention (Reported by Alexander...

The Asterisk Development Team has announced the release of Asterisk 11.17.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.17.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: New

...ev) * ASTERISK-28839 - Sporadic crashes with Segmentation fault (Reported by Joeran Vinzens) * ASTERISK-28835 - IPv6 addresses in SDP incorrectly formatted (Reported by Daniel Heckl) * ASTERISK-28372 - Asterisk REPLY Wrong Contact header port (TCP) (Reported by Anton Satskiy) * ASTERISK-24428 - Document that Asterisk will use the default SIP ports (5060 for TCP, 5061 for TLS) if the extern option variants aren't used (Reported by sstream) * ASTERISK-28838 - AST_MODULE_INFO requires, MODULEINFO does not mention (Reported by Alexander...

Hi Luca Y need to check your wifes codec priority list -seems to be GSM on the first place. Luca Bertoncello <lucabert at lucabert.de> wrote: >Hi list! > >I noticed that when the phone of my wife calls the gsm codec will be used, >but if someone calls the phone, alaw will be used: > >00493511111111 calls 00493512222222: >OpenWrt*CLI> sip show channels >Peer