search for: sgriepentrog

...NVITE, OPTIONS, PRACK, REFER, REGISTER, UPDATE v=0 o=- 0 0 IN IP4 217.0.23.68 s=- c=IN IP4 217.0.4.134 t=0 0 m=audio 36480 RTP/AVP 9 8 102 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:102 telephone-event/8000 a=maxptime:20 a=ptime:20 > Am 02.04.2015 um 22:00 schrieb Scott Griepentrog <sgriepentrog at digium.com>: > > Actually, the IP address is still used to identify the incoming invite. With the insecure=port option set, Asterisk will presume the invite to still match the trunk account even if the NAT router has mangled (changed) the port number. My suspicion is that when the ne...

...matching port number ;insecure=invite ; Do not require authentication of incoming INVITEs ;insecure=port,invite ; (both) Do I understand correctly that in this mode the IP address is not checked and no authentication is required? > Am 02.04.2015 um 20:11 schrieb Scott Griepentrog <sgriepentrog at digium.com>: > > ?I'd be curious if setting > > insecure=invite,port > > makes any difference either (without alllowguest on). > ? > > On Thu, Apr 2, 2015 at 9:03 AM, Daniel Heckl <daniel.heckl at gmail.com <mailto:daniel.heckl at gmail.com>> wro...

...ethods of moving channels in to bridges with ARI could be used.? On Sat, May 23, 2015 at 1:33 AM, Nick Awesome <jleed at me.com> wrote: > recreate Echo, if that is possible. trying to recode all dialplan to > stasis application > > On 22 May 2015, at 19:29, Scott Griepentrog <sgriepentrog at digium.com> > wrote: > > Nick- > > Are you wanting to recreate the dialplan Echo() application in stasis? > > Why not just send the call to Echo() instead of Stasis()? > > On Fri, May 22, 2015 at 11:25 AM, Matthew Jordan <mjordan at digium.com> > wrote: &gt...

...directmedia=no sendrpid=pai trustrpid=no insecure=port,invite disallow=all allow=g722 allow=alaw allow=gsm deny=0.0.0.0/0 permit=217.0.0.0/13 [DTAG-IP_IN18_016](telekom) host=217.0.18.16 [DTAG-IP_IN18_036](telekom) host=217.0.18.36 etc. > Am 02.04.2015 um 23:21 schrieb Scott Griepentrog <sgriepentrog at digium.com>: > > That sounds like asterisk was working 100% correctly. If you receive an INVITE from an unknown IP address, then it should fail. Unless you want to allow anonymous, which is genearlly a very bad idea. > > If you are registering to IP X, but the provider may be...

The Cisco/Linksys SPA devices are also able to be provisioned automatically. On Wed, Apr 15, 2015 at 3:20 PM, Bryant Zimmerman <BryantZ at zktech.com> wrote: > Alejandro > > All of the Grandstream devices can be remote provisioned if you know what > you are doing. > > Bryant > > ------------------------------ > *From*: "Alejandro" <cdgraff at

On Thu, 28 May 2015 11:15:45 -0500 Scott Griepentrog <sgriepentrog at digium.com> wrote: > The string "5a2600300339934f704528bb14ed05e9 at MyAsterisk:5060" is the unique > identifier for the call in SIP known as the Call-ID. If you have a packet > capture of the port 5060 SIP traffic, that identifier will be in each SIP > message related...

Nick- Are you wanting to recreate the dialplan Echo() application in stasis? Why not just send the call to Echo() instead of Stasis()? On Fri, May 22, 2015 at 11:25 AM, Matthew Jordan <mjordan at digium.com> wrote: > On Fri, May 22, 2015 at 4:41 AM, Nick Awesome <jleed at me.com> wrote: > > Can anyone tell me how can I create echo test using ARI stasis > application?

...t; o=- 0 0 IN IP4 217.0.23.68 > s=- > c=IN IP4 217.0.4.134 > t=0 0 > m=audio 36480 RTP/AVP 9 8 102 > a=rtpmap:9 G722/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:102 telephone-event/8000 > a=maxptime:20 > a=ptime:20 > > Am 02.04.2015 um 22:00 schrieb Scott Griepentrog <sgriepentrog at digium.com > >: > > Actually, the IP address is still used to identify the incoming invite. > With the insecure=port option set, Asterisk will presume the invite to > still match the trunk account even if the NAT router has mangled (changed) > the port number. My suspicion...

...](endpoint_internal) auth=demo-bob aors=demo-bob mailboxes=box_b rewrite_contact=yes [demo-bob](auth_userpass) password=demo-bob ; put a strong, unique password here instead username=demo-bob [demo-bob](aor_dynamic) Thank you for your help! On Thu, Jan 8, 2015 at 11:48 AM, Scott Griepentrog <sgriepentrog at digium.com> wrote: > It would appear that you have the Asterisk server on a public IP address, > your two endpoints are behind a NAT, and you have rewrite_contact enabled > in pjsip.conf. > > In which case, what you are seeing is correct. In order to be able to > send a ca...

Hi Scott, thanks for the answer, can share some link or documentation about how setup this in SPA3102? I try to get something about this using google, but found comments but nothing useful. Alejandro 2015-04-15 19:28 GMT-03:00 Scott Griepentrog <sgriepentrog at digium.com>: > The Cisco/Linksys SPA devices are also able to be provisioned > automatically. > > On Wed, Apr 15, 2015 at 3:20 PM, Bryant Zimmerman <BryantZ at zktech.com> > wrote: > >> Alejandro >> >> All of the Grandstream devices can be remote pro...

recreate Echo, if that is possible. trying to recode all dialplan to stasis application > On 22 May 2015, at 19:29, Scott Griepentrog <sgriepentrog at digium.com> wrote: > > Nick- > > Are you wanting to recreate the dialplan Echo() application in stasis? > > Why not just send the call to Echo() instead of Stasis()? > > On Fri, May 22, 2015 at 11:25 AM, Matthew Jordan <mjordan at digium.com <mailto:mjordan a...

...Reported By Jan Juergens Posted On December 16, 2013 Last Updated On December 16, 2013 Advisory Contact Scott Griepentrog <sgriepentrog AT digium DOT com> CVE Name Pending Description A 16 bit SMS message that contains an odd message length value will cause the message decoding loop to run forever. The message buffer...

...Reported By Jan Juergens Posted On December 16, 2013 Last Updated On December 16, 2013 Advisory Contact Scott Griepentrog <sgriepentrog AT digium DOT com> CVE Name Pending Description A 16 bit SMS message that contains an odd message length value will cause the message decoding loop to run forever. The message buffer...

Hi All I have a few lines like this at asterisk/messages. [May 25 15:22:42] WARNING[27725] chan_sip.c: Hanging up call 5a2600300339934f704528bb14ed05e9 at MyAsterisk:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). Since we have hundreds of clients with hundreds of simultaneous calls, how is it possible to know to which customer/IP

...er > ;insecure=invite ; Do not require authentication of incoming INVITEs > ;insecure=port,invite ; (both) > > Do I understand correctly that in this mode the IP address is not checked > and no authentication is required? > > Am 02.04.2015 um 20:11 schrieb Scott Griepentrog <sgriepentrog at digium.com > >: > > ?I'd be curious if setting > > insecure=invite,port > > makes any difference either (without alllowguest on). > ? > > On Thu, Apr 2, 2015 at 9:03 AM, Daniel Heckl <daniel.heckl at gmail.com> > wrote: > >> Ok, I have teste...

On Wed, Jul 29, 2015 at 10:16 AM, John Novack <jnovack at stromberg-carlson.org > wrote: > > > Murthy Gandikota wrote: > > > > ------------------------------ > To: asterisk-users at lists.digium.com > From: webaccounts173 at jgoettgens.de > Date: Wed, 29 Jul 2015 16:11:31 +0200 > Subject: Re: [asterisk-users] Windows Asterisk Help > > > >

..., but the problem is, that after an ip change AND a new registration the ip address of the peer is not updated automatically. INVITES are answered with 401. Only after a sip reload the peer works again. That can't be normal... Daniel > Am 31.03.2015 um 22:45 schrieb Scott Griepentrog <sgriepentrog at digium.com>: > > You have two options for dealing with an IP change during the registration period: > > 1) set the registration time to shorter period of time to minimize the downtime > > 2) detect that the IP address has changed via whatever method available, and then is...

I would recommend capturing traffic outside your Asterisk server with Wireshark, then running the Telephony/Rtp/Analysize Streams option to determine if you have packet loss at that point in the network. On Mon, Jan 19, 2015 at 1:00 PM, Todd R. <tjrlist at live.com> wrote: > Thanks but no Adtran here. > > I do think these stats are indicating an issue, I just don't know how to

You have two options for dealing with an IP change during the registration period: 1) set the registration time to shorter period of time to minimize the downtime 2) detect that the IP address has changed via whatever method available, and then issue a "sip reload" CLI command to asterisk, which will cause it to resend registrations immediately. On Tue, Mar 31, 2015 at 1:36 PM, Daniel

On Tue, Jan 20, 2015 at 12:43 AM, Scott Griepentrog <sgriepentrog at digium.com > wrote: > I would recommend capturing traffic outside your Asterisk server with > Wireshark, then running the Telephony/Rtp/Analysize Streams option to > determine if you have packet loss at that point in the network. > > On Mon, Jan 19, 2015 at 1:00 PM, Todd R. &l...