search for: sip2sip

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] NOTICE[22492]: res_pjsip/pjsip_distributor.c:659 log_failed_request: Request 'IN...

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 mu...

...it all here, I've posted my logs and conf files on that thread, too. Problem is that while there are quite a few sip examples, I have chosen to take the path of pjsip. Seems I can manage to attach Blink, Zoiper, Microsip and my ITSP with multiple extensions without problem to my Asterisk, but sip2sip has beaten me! It's presumably something ridiculously simple, but there comes a point where you can't see the wood for the trees. If someone can help me resolve this, I'll post a complete guide on Github Gist to help others in the future. Thanks.

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 Sip2sip.info , Can i able to communicate with this scenario ? How i should configure extention.conf in local asterisk sever to communicate with soft-phone which registered at Sip2sip.info ??...

...addr = stun.ideasip.com stunrefresh = 30 stun show status Hostname Port Period Retries Status ExternAddr externport stun.ideasip.com 3478 30 3 OK 61.12.17.171 39710 sip.conf localnet=192.168.0.0/255.255.255.0 register=>jai9999:123456:jai9999 at sip2sip.info/jai9999 when above command runs , it is sending register method with my private ip address. REGISTER sip:sip2sip.info SIP/2.0 Via: SIP/2.0/UDP 192.168.0.187:5060;branch=z9hG4bK46d2b3e0 Max-Forwards: 70 From: <sip:jai9999 at sip2sip.info>;tag=as555629a9 To: <sip:jai9999 at sip2sip.in...

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 machine xen8.vandervlis.nl: ------- paul at server2:~$ host -t SRV _sip._udp.vandervlis.nl _sip._udp.vandervlis.nl has SR...

...ype=peer dtmfmode=rfc2833 context=sip insecure=very host=sip.domain.com nat=no[1000];IPKall type=peer dtmfmode=rfc2833 context=sip insecure=very host=voiper.ipkall.com nat=no I pasted the log here -> http://pastie.org/1163238 I have tried connecting both of the clients to another sip service(sip2sip.info) and did not have the same problems. Any suggestions would be great. Thanks, Tom -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100916/8037954e/attachment.htm

...ere 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 call is originated thru a pstn/telco line. First test is to send the fax to a client in their SBC device.it was a direct sip2sip fax call. And it succeeded. Then when we tried to fax to a pstn number fax hung up because of communication error. The only error i received was Unknown RTP codec 100 received from xxx.xxx.xxx.xx If i got it right, they say for normal calls they use 99 as dtmf payload. But for fax they use 100....

...that may have a big share, but I find that limit. I understand it being in the kernel is attractive because it's much faster, but how many things do we want to trust in running in the kernel? Does a faster VPN really mean much these days? -- ------- inum: 883510009027723 sip: jungleboogie at sip2sip.info

Probably doesn't help diagnose the problem.... but there were also audio problems experienced with this cvs version even on LAN / sip2sip / no transcoding > > ERROR[1171] UTILS.C:509 TVFIX: WARNING NEGATIVE TIMESTAMP -194931. ... > > I will be looking into this issue later today. __________________________________ Yahoo! Mail Stay connected, organized, and protected. Take the tour: http://tour.mail.yahoo.com/mail...

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

...find helpful data for 1.8. I am trying to figure out hardware configuration for following features implemented in Asterisk 1.8? (1)100 SIP clients. (2)ACD (Around 15 realtime queues) (3)Call recording for all SIP clients. (4)4 port PRI (E1). There would be around 100 concurrent calls. (DAHDI2SIP,SIP2SIP,SIP2DAHDI) (5)IVR (6)Around 50 Mysql queries per call (through ODBC). (Remote Database) (7)MOH I can provide further information if missing something. Thanking you in advance. --AM -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/piperma...

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

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

I've been following the list for months, and I have a working Asterisk setup, but it'd be *really* useful to me at this point if someone could summarize when Asterisk has echo problems and when it doesn't. For instance, I usually hear a far-end echo when talking on my 7940, but not when using a POTS phone plugged into a TDM400 FXO port. It doesn't seem to matter if the call

In a few weeks, no SIP call is going to terminate unless they are signed properly, as mandated by law. We are in the business of Stir-Shaken, signing calls, as an FCC-approved provider. A big differentiator between our service and the rest: we are the only ones who don't need to receive the calls in our servers to sign them. We do this over a MySQL call, easily connectable to Asterisk via