similar to: Asterisk 13.6 + pjsip: sip2sip registers but incoming calls get "No matching endpoint found".

Using pjsip 2.7.2 on Asterisk 15.5 Really struggling to make sense of translating these old 1.8 SIP instructions into a neat pjsip_wizard conf suitable for 2018 http://wiki.sip2sip.info/projects/sip2sip/wiki/SipDevicesAsterisk#Version-18 In pjsip_wizard.conf, I have the following, which seems to get me registered, and it responds to an incoming call, but I always get this: [Jul 28 18:32:29]

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

Microsip (Windows) is free and small. 2.5Mb download, 10Mb RAM usage, does everything I need and configuring TLS is a doddle. http://www.microsip.org/ On 16 February 2017 at 13:04, Max Grobecker <max.grobecker at ml.grobecker.info> wrote: > Hello, > > I'm a big fan of PhonerLite. > It's more poplar in Germany, but also available in English language. > This client

Hi, I have to create scenario like following, I have 2 sip soft phone.I configured Asterisk server on local network, on Linux.With two soft-phone , local asterisk sever, i able to communicate.Now i have communicate with other network SIP client.For that i have opened account at @sip2sip.info, they provided me credentials.Then i registered one SIP phone to local Asterisk sever and another to

Hello, I cannot call my server over the internet with SIP anymore. Even when I do a maximum logging on my firewall, I don't see packets coming from outside. I've tried it from an ekiga.net account and an sip2sip.info account. What could be wrong? I would expect incoming traffic on port 5060 UDP... The account is "paul at vandervlis.nl". This should connect trought DNS to the

Hello, can someone recommend a good and free Softphone for Windows which does not display advertisments inside the program? We have X-Lite but free version display advertisments. thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140608/c3a9e887/attachment.html>

I am having a one way audio issue with xlite clients behind NAT. They can connect to the server and make calls but no audio is heard on the other end. my sip conf [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes canreinvite=no[tomfmason] type=friend secret=secret callerid="Thomas Johnson" <XXXX> host=dynamic nat=yes canreinvite=no disallow=all allow=gsm

Hi first post, so hope I'm not violating protocol. Been using Asterisk as home phone and hobby use for nearly 10 years. I love this project. Anyway, would someone mind verifying my pjsip.conf ? This seems to work well for 14.3.1 but I get no rtp into my natted Linphone when I upgrade to 14.4.1. Other than that the phone registers properly on 14.4.1. I can provide a pjsip log as well,

Hi, (please excuse me for lack of proper jargon usage and the vagueness of description...) i use Asterisk 11.12.1, (well... as included in FreePBX), . . . The softphones are mostly on machines without proper sound hardware (no mics, no speakers/headsets); This is partly because the workforce is quite conservative in what they want to use :) meaning handsets are important; As the handsets have

Hello using asterisk 1.8.32.3 I am not able to make a call with video support. I do not know what I am missing to make this video call. Codec h264 should be supported. sip*CLI> core show codecs Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. INT BINARY HEX TYPE NAME

Well, what immediately stands out is: "FILE * open failed!" Have you triple checked that the full filepath is correct and that the user that Asterisk is running as has full permissions to access your valid certificate file? I have it working with microsip and a free TLS cert from LetsEncrypt. When I get to the PC with that on, I can write up what settings I've got if that helps?

Hello in sip.conf I have ; videosupport=yes Kind regards. J. On 20-04-17 13:09, Marcelo Terres wrote: > I suppose that you enable the video support on sip.conf, right? > > Regards, > Marcelo H. Terres <mhterres at gmail.com> > IM: mhterres at jabber.mundoopensource.com.br > https://www.mundoopensource.com.br > https://twitter.com/mhterres >

Hello, I have a user that prefers Soft SIP phone install on his laptop, for security reasons I have enable TLS on our Asterisk server to support TLS authentication, It works well with hard phones. Has anybody in this forum use SIP Soft phones with TLS authentication enabled? Any suggestions? Thanks, Motty -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Am 16.02.2017 um 14:19 schrieb Annus Fictus: > And Microsip using PJSIP SIP stack :) Sorry (also, for off-topic), based on my latest experience with PJSIP, I'm not sure if this really is a sign of good quality. Maybe it's a problem with the implementation in Asterisk (I haven't tried PJSIP in other software), but after just five minutes of testing I found several bugs

Hello, We have a sip trunk between our voip operator and our asterisk 1.6.2.9 We have no problem during voice communications. But we can not send any t38 fax via this gateway. We tried to trace the error made some tests.. There are 2 main tests we tried to do. As i learned their voip path is like .. we connect to session border controller..then it routes the call to a cisco media gateway if the

Hi, I?m trying to get TLS to work with asterisk and client phones, and all I?m getting from asterisk is [Aug 23 11:46:42] WARNING[1170]: tcptls.c:673 handle_tcptls_connection: FILE * open failed! == Problem setting up ssl connection: error:00000000:lib(0):func(0):reason(0) [Aug 23 11:46:44] WARNING[1171]: tcptls.c:673 handle_tcptls_connection: FILE * open failed! when clients try to

Dear all I have asterisk 1.4.11 on CentOS. I have SIP IP phone arround 100 but i got Noice on voice call so what would be the resone and how to fine tune my voice quality on asterisk ?? what codec would be best for my asterisk --------------------------------- Catch up on fall's hot new shows on Yahoo! TV. Watch previews, get listings, and more! -------------- next

I need to access number received after a I dial a SIP or H323 call? suppose I get one of these: *404 Not found **486 Busy here **408 Request Timeout **480 Temporarily unavailable **480 Temporarily unavailable **403 Forbidden (+) ** 410 Gone **301 Moved Permanently **410 Gone ** 404 Not Found (=) **502 Bad Gateway **484 Address incomplete* How do I get the 404, 486, etc. F.A. -------------- next

(this time with subject....) Hello, I?m trying to get Asterisk to accept incoming calls from budgetphone.nl. When I dial my budgetphone nr on a PSTN KPN line it immediately gives a busy tone. I tried X-lite, which worked perfect, so my modem (with nat) probably is not the problem. I did a sip debug and got the following output. Because I?m new to Asterisk I can?t get the error why this is not

Hello everybody, I saw Guus already had contact with Jason over email. What do you guys tinc of wireguards, are there advantages? Jason seems to have a good grip of what he is talking about. https://fosdem.org/2017/schedule/event/wireguard/attachments/slides/1675/export/events/attachments/wireguard/slides/1675/wireguard_slides.pdf https://fosdem.org/2017/schedule/event/wireguard/ Kind