search for: griepentrog

...ot include the SIP Headers I added. For chan_sip, I have no problem with this. Even the original Queue code I had includes the added SIP headers with it?s INVITE to the Agent. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Scott Griepentrog Sent: Thursday, August 27, 2015 4:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet? Local channels: http://www.asteriskdocs.org/en/3rd_Edition/aste...

...then the usual methods 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&gt...

...nels essentially be an internal bridge? How would I ?Register Local/number at agent in the queue on behalf of the agent (replace number with the agent's extension number)? From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Scott Griepentrog Sent: Thursday, August 27, 2015 1:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet? To add a header to the call leg that goes to the agent, try usi...

...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 > -- [image: Digium logo] Scott Griepentrog Digium, Inc ? Software Developer 445 Jan Davis Drive NW ? Huntsville, AL 35806 ? US direct/fax: +1 256 428 6239 ? mobile: +1 256 580 6090 Check us out at: http://digium.com ? http://asterisk.org -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium....

...CANCEL, INFO, INVITE, 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...

...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 > -- [image: Digium logo] Scott Griepentrog Digium, Inc ? Software Developer 445 Jan Davis Drive NW ? Huntsville, AL 35806 ? US direct/fax: +1 256 428 6239 ? mobile: +1 256 580 6090 Check us out at: http://digium.com ? http://asterisk.org -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium....

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 &gt...

...exten => 1234,1,Verbose(X-My-DNID:${MY_DNID}) same => n,Set(X-My-DNID=${MY_DNID}) same => n,Set(PJSIP_HEADER(add,X-My-DNID2)=${MY_DNID}) same => n,Dial(PJSIP/Agent1) From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Scott Griepentrog Sent: Thursday, August 27, 2015 4:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet? Are you using this method of setting headers on PJSIP? https:/...

...; 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 gmai...

...>Subject: Re: [asterisk-users] recording in mp3 </div><div> </div>i would go for recording into wav. then at regular intervals eg every night at 01:00 i would start a script to convert the wav to mp3 and then delete the wav files. it is really easy. On 30/6/2014 23:30, Scott Griepentrog wrote: ?You will not be able to able to save much space if any by using MP3 instead of ulaw or wav -- at least not without expending a lot of CPU time to encode the file at a very low bitrate which sounds pretty bad even with just speech. One of the better space savings options for recor...

...> > 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 numbe...

Howdy, I built an LXC container with an "image" of asterisk 11.18 precompiled and installed. It runs fine on the dev platform, which is a Dell R320 running Ubuntu 14.04LTS. I shutdown the container, tarred it up, and untarred on a Dell PE1850, also running Ubuntu 14.04LTS. The container itself is Ubuntu 14.04LTS. Both platforms as far as I know are amd64. The container boots

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

...tching 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> > wrote: > >>...

I have a call coming in. I need to add a SIP Header to the channel. Then, I need to send the call to the Queue so it is sent to the Agent. The SIP header I added, I need to have appear in the INVITE sent to the Agent. It works in chan_sip. I send the call to a macro which does... n,SIPAddHeader(My-Header-Name:${MY-HEADER-VALUE}) n,Queue(${ARG2}) In PJSIP , this doesn't seem to work. Is

...amic) [demo-bob](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...

...tmfmode=rfc2833 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...

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...

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 &lt...

Hi, we have quite a few Asterisk machines running and try to keep them on a current version of the Asterisk 11 branch. But since we upgraded to 11.14.0 a couple weeks ago, we have to restart the Asterisk process every week because the load gets too high and our monitoring complains. Those machines are doing only SIP-to-SIP call relay, the dialplan is quite complex, transcoding is done only on a