similar to: Binding SIP on multiple ports

Great ! I'm gonna it try ASAP ! Is there another way (ie not using different ports) to get several trunks to a given ITSP ? Let me explain this a bit further. My setup is: ITSP <---- SIP----> Asterisk <----> Phones For various reasons, I want my Asterisk box to have several trunks/SIP account with my ITSP. First method, is to configure a specific port for each trunk: ITSP will

Hello, For lab testing, I'm trying to build two differents PJSIP trunks between two Asterisk 13.8.0enabled boxes. I thought I could set up both trunks like this: Box A/port 5060 <------ Trunk1 -----> Box B/port 5060 Box A/port 5062 <------ Trunk2 -----> Box B/port 5062 and declare trunks like this: [foobar1] type=endpoint transport=simpletrans context=from-customer

Hello, On a newly built Asterisk 13.13.1 system, I can't make HEP work with chan_sip (though I could make it work with PJSIP on another instance). Looking either at [1] or hep.conf, I can't see anything meaning HEP requires PJSIP. Before diging deeper, may I simply ask if HEP requires PJSIP or not ? What about a line mentioning the answer in above documents (to keep other from wondering

-----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 CCSS in any way. Is CCSS support planned for PJSIP? chan_sip is in "extended" state in asterisk-13, so chan_pjsip should be preferred for new installations, ri ght? Thanks, - --

I'm trying to configure sip2sip, which says: http://wiki.sip2sip.info/projects/sip2sip/wiki/SipDevicesAsterisk "Asterisk, is currently unable to handle more that one result for a DNS SRV lookup, and the Asterisk configuration needed for getting it work with the SIP2SIP service is not trivial" It then gives a complex multi-section workaround in SIP. I remember reading there'd be

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, thank you for the assistance! ??, 13 ???. 2017 ?. ? 16:38, Joshua Colp <jcolp at digium.com>: > On Mon, Mar 13, 2017, at 10:32 AM, Kseniya Blashchuk wrote: > > Tested with latest Asterisk 14.3.0 on Ubuntu 16 kernel 4.4.0-66-generic > > and > > Centos 7 kernel 3.10.0-514.10.2.el7.x86_64. Absolutely the same behavior. > > Joshua, maybe you can advice what can

Hello, I think I met a case similar to the one solved by [1] . Quoting this case : * res_pjsip: Handle deferred SDP hold/unhold properly. Some SIP devices indicate hold/unhold using deferred SDP reinvites. In other words, they provide no SDP in the reinvite. A typical transaction that starts hold might look something like this: * Device sends reinvite with no SDP * Asterisk

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 voice, video and text in conferencing. On an other hand, there are some softphones (Zoiper, Bria, ...) that tout voice, video and chat capability. Though Enhanced Messaging solution described in [1] seems more attractive to me in the

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 traffic leaving the server. I have ensured that there is a DNS record for the server & that it can resolve it. I've also added records to my hosts file and checked using 'genet ahosts hostname' but still the issue

We have a customer with a system rejecting calls from Asterisk. It's indicating the ptime is 60, but the system admin is saying they only support 20. They are running asterisk 16.2.1 and using chan_sip Is there a way to specify this with chan_sip? Also, for my own curiosity, is there a way to specify this with PJSIP? (Trying to migrate customers to PJSIP, but we are holding until asterisk

Hello, I'd like to know if anyone of you is finding my same problems using any recent asterisk version, after 13.7 / 13.8 with chan_sip. If I use any recent asterisk version, after just few seconds asterisk completely locks up, stopping processing SIP/UDP packets. Nothing is written in the asterisk log, but if I run "netstat -nap | grep 5060" I see the UDP buffer filled up. If I

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi list, It looks like Call Completion Supplementary Services is not available for PJSIP channels, am I right? Is there another solution? Thanks, - -- Jean-Denis Girard SysNux Syst?mes Linux en Polyn?sie fran?aise http://www.sysnux.pf/ T?l: +689 40.50.10.40 / GSM: +689 87.79.75.27 -----BEGIN PGP SIGNATURE-----

On Fri, 11 May 2018, Joshua Colp wrote: >> In the above example, even though the INVITE/SDP says they prefer gsm >> over ulaw and the OK/SDP says I prefer ulaw over gsm, they can choose >> to use gsm or ulaw? > > Yes. > >> Can it be asymmetrical? They send gsm and I send ulaw? > > Technically, yes. In practice it's a bit iffy - specifically because

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,

Hi all, we are trying to move our servers from chan_sip to chan_pjsip. At this time no problems with phones, they all register fine and can place calls. But for a trunk we face problem and can't place calls despite the fact that registration is OK. What we get is: [2018-04-16 16:08:33] WARNING[18665]: res_pjsip_outbound_authenticator_digest.c:178

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

Hello! I'm trying to send a fax via T.38 to a destination, which should be T.38 capable. My provider supports T.38, too. Unfortunately, it doesn't work. This means: Call is started and SDP is negotiated w/ alaw. Callee sends reinvite - for alaw again (and not for T.38)!! After about 30s, callee hangs up because of missing data (this is true, because I don't send alaw coded fax data.

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: