similar to: Binding SIP on multiple ports [SOLVED])

Hello Joshua, What will be example of endpoint configuration that not require authentication from specific ip ? volga629 On Sun, 3 Dec, 2017 at 11:01 AM, Joshua Colp <jcolp at digium.com> wrote: > On Sun, Dec 3, 2017, at 10:55 AM, volga629 at networklab.ca wrote: >> If understand correctly type=identify is more for sip trunk >> configuration ? >> >>

If understand correctly type=identify is more for sip trunk configuration ? ;[mytrunk] ;type=identify ;endpoint=mytrunk ;match=198.51.100.1 ;match=198.51.100.2 In chan_sip it was just reply 200 OK on keepalive packet without need define trunks. volga629 On Sun, 3 Dec, 2017 at 10:45 AM, Joshua Colp <jcolp at digium.com> wrote: > On Sun, Dec 3, 2017, at 10:42 AM, volga629 at

On Fri, 2019-03-01 at 14:15 -0500, Joshua C. Colp wrote: > you can try line functionality on the outbound registration which > may or may not work[2] (requires the upstream to adhere to the RFC, > which not all do). My provider seems to implement this. However even with the line=... in the: SIP to address: sip:5555551212@<my_IP_address>:5060;line=dpnlyiu res_pjsip is still

Sorry, should have included that. Asterisk 16.2.1 Mark. On Thu, 4 Apr 2019 at 14:56, Joshua C. Colp <jcolp at digium.com> wrote: > On Thu, Apr 4, 2019, at 10:53 AM, Mark Farmer wrote: > > As I understand it, delays like this are almost always caused by slow > > or failing DNS lookups. Running a packet capture on all interfaces > > filtering on port 53 shows no DNS

I'm using chan_sip, I experimented with adding a 'Route' header in the originate command and used the Dial command like 'SIP/peer/exten', but problem is that Request-URI isn't populated correctly. I'm using Asterisk 13. Thanks, Nitesh On Wed, Apr 13, 2016 at 10:09 PM, Joshua Colp <jcolp at digium.com> wrote: > Nitesh Bansal wrote: > >> Hello,

Hello, With asterisk 12 improvements, is it now possible to bind an asterisk SIP stack to several ports ? For instance, to both emit or listen on ports 5060 and 5062. If positive, any hint on how to get more detail about this ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Ludovic Gasc wrote: > 2015-05-21 17:59 GMT+02:00 Jean-Denis Girard <jd.girard at sysnux.pf > <mailto:jd.girard at sysnux.pf>>: > > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Le 21/05/2015 00:16, Joshua Colp a ?crit : > > If CCSS is needed then the only option is to use chan_sip. The > > chan_pjsip module does not implement

Ok that did it after I did the steps to completely remove everything and do a new install. Thanks! > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- > bounces at lists.digium.com] On Behalf Of Joshua Colp > Sent: Wednesday, September 23, 2015 10:12 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject:

On Sun, Dec 21, 2014 at 4:54 AM, Recursive <lists at binarus.de> wrote: > Dear list, > > I am currently trying to send faxes via T.38 using PJSIP (newest version 2.3) with Asterisk 13.0.2. After having configured PJSIP, I have seen several things the cause of which I would like to know. > > 1) Ports and IP addresses which PJSIP bind to > > I have configured one transport

Hi! Unfortunately pjsip is broken in Ubuntu Asterisk installed from repo. Yes I also thought to try with pjsip, just to know if it's also affected. I'll try to make a test next days. On Sun, Apr 16, 2017, 8:18 PM Ludovic Gasc <gmludo at gmail.com> wrote: > Hi Kseniya, > > You might test with chan_pjsip: We have less production experience with > chan_pjsip than

The Asterisk Development Team would like to announce the release of Asterisk 17.0.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 17.0.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Certified Asterisk 16.8-cert1. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/certified-asterisk The release of Certified Asterisk 16.8-cert1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following

The Asterisk Development Team would like to announce the release of Certified Asterisk 16.8-cert1. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/certified-asterisk The release of Certified Asterisk 16.8-cert1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following

On Sat, Apr 6, 2019, at 10:04 AM, sean darcy wrote: > On 4/5/19 10:36 AM, sean darcy wrote: > > I'm trying to set up pjsip to work with an obi202 and google voice. But > > I can't configure the endpoint. > > > > pjsip: > > > > [obi202-auth](!) > > type = auth > > auth_type = userpass > > password = <mypass> > > >

Dear list, I am currently trying to send faxes via T.38 using PJSIP (newest version 2.3) with Asterisk 13.0.2. After having configured PJSIP, I have seen several things the cause of which I would like to know. 1) Ports and IP addresses which PJSIP bind to I have configured one transport like that: [tr_wZCMk5MvC2ATNzAr] type = transport protocol = udp bind = 192.168.20.48 Nevertheless, PJSIP

Thanks for your (fast) reply ! Le ven. 18 janv. 2019 à 16:32, Joshua C. Colp <jcolp at digium.com> a écrit : > On Fri, Jan 18, 2019, at 11:22 AM, Olivier wrote: > > Hello, > > > > I've just read in [1] about SIP MESSAGE addition to both chan_pjsip and > > ConfBridge. > > It seems very interesting addition as it brings the capability to mix > >

Andrew Martin wrote: > ----- Original Message ----- >> From: "Joshua Colp"<jcolp at digium.com> >> To: "Asterisk Users Mailing List - Non-Commercial Discussion"<asterisk-users at lists.digium.com> >> Sent: Monday, May 11, 2015 12:32:06 PM >> Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped calls after 32

The Asterisk Development Team would like to announce the release of Asterisk 15.0.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 15.0.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The Asterisk Development Team has announced the release of Asterisk 13.5.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.5.0 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:

The Asterisk Development Team would like to announce the release of Asterisk 18.0.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 18.0.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release: