[asterisk-users] WebRTC - Transport Issues.
Hey all. I have webrtc up and running with asterisk 11. All is going well with TLS now working. At least I hope it is using TLS and wss. Based on what I am seeing I have UDP, WSS listed in the Allowed transports, but every time I connect the Primary transport shows WS.. Why is this? Am I actually running ws in wss mode? Prim.Transp. : WS Allowed.Trsp : UDP,WSS Def. Username: 6167761066.2011 SIP Options : (none) Codecs : (ulaw) Codec Order : (ulaw:20) Auto-Framing : No Status : OK (71 ms) Useragent: SIP.js/0.7.7 Reg. Contact : sip:fed97qgu@192.0.2.35;transport=wss Any Insights would be appreciated. Thanks Bryant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] double NAT - one way audio
Hi Andre, Some routers just simply won't support this double-nat scenario you describe. Othera will... And without any special forwarding. Is it possible to put the first router into "bridge" mode, and use the second router as the actual NAT router? This may be the quickest solution to your problems. Good luck! Thanks, Glenn (mobile) On Mar 11, 2017 8:50 AM, "Andre Gronwald" wrote: Hi all, I have a setup which is not working right now: Provider - DSL-Router (192.168.2.1) - Bintec-Router (10.17.46.66) - Asterisk (10.17.46.99) My issue: Everything works, but RTP is only going from my Asterisk towards the provider. Asterisk is configured to use SIP-ports 55060 and RTP-ports 51000-51999. Those ports are forwarded on DSL-router to the bintec router and from the bintec router to asterisk. what I see is the Invite from provider goes to 192.168.2.1 and rtp port 7070. my asterisk responds with audio to be sent to ip address 80.142.12.12 port 51242. Afterwards RTP goes from 10.17.46.99:51242 to 192.168.2.1:7070. but the RTP back is not coming in. I would expect the RTP traffic to be sent to 80.142.12.12 (intetnal 192.168.2.1) port 51242 - then i would have successfully two way audio. But why is port 7070 used? The DSL-router is a speedport w724v type A. regards, andre -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] double NAT - one way audio
Hi all, I have a setup which is not working right now: Provider - DSL-Router (192.168.2.1) - Bintec-Router (10.17.46.66) - Asterisk (10.17.46.99) My issue: Everything works, but RTP is only going from my Asterisk towards the provider. Asterisk is configured to use SIP-ports 55060 and RTP-ports 51000-51999. Those ports are forwarded on DSL-router to the bintec router and from the bintec router to asterisk. what I see is the Invite from provider goes to 192.168.2.1 and rtp port 7070. my asterisk responds with audio to be sent to ip address 80.142.12.12 port 51242. Afterwards RTP goes from 10.17.46.99:51242 to 192.168.2.1:7070. but the RTP back is not coming in. I would expect the RTP traffic to be sent to 80.142.12.12 (intetnal 192.168.2.1) port 51242 - then i would have successfully two way audio. But why is port 7070 used? The DSL-router is a speedport w724v type A. regards, andre -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk/FFA version upgrade recommendation
Hi all, I'm needing to upgrade Asterisk from 10.x to whatever the recommended version is that will allow me to continue to use Fax For Asterisk. I don't have many upgrade windows, I'd like to get the most bang for my buck, but I can't afford to be a beta tester on this server. The FFA site says that it's supported by Asterisk version 12 and lower, but version 12 doesn't seem to be supported. Perhaps my information is outdated? Anyway, I can't go with the spandsp route because my system listens for AMA events that spandsp doesn't seem to produce and I can't emulate easily. Any recommendations would be very welcome. -- Mike Diehl -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tcpbind and source IP address
Hey guys, any thoughts on that? Probably a bug or is it a default behavior? On Thu, Mar 9, 2017, 2:05 PM Kseniya Blashchuk wrote: > Hi all! > I am running asterisk 13.1.0 on Ubuntu server 16.04. There are two IP > addresses from the same subnet set on one interface, and bindaddr is set to > the second on them in sip.conf and in iax.conf. > Incoming connections work as expected. However, for outgoing connections > it seems that asterisk tells the kernel to use the specific "bind" address > only in case of UDP usage (both SIP and IAX work like that). In case of > outgoing TCP connections (SIP TCP and TLS) the first IP address from the > interface is used. > In my understanding, normally 'bind' should not only tell on which address > to listen, but also which source address to request for outgoing > connections, but it works only for UDP connections for some reason. > Can anybody explain if it's a normal behavior? > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Optimizing forwarded and redirected calls by avoiding signaling and media data redirection
Hi! Apparently this is possible; my asterisk server is doing this when my SIP phone redirects the call with a SIP REFER message. The phone is excluded from the call after it transfers the call. I'll contact my ITSP if their trunk can also do this. Regards, Sree On 03/09/2017 11:03 PM, Sree Harsha Totakura wrote: > Hi! > > I'm having a setup where my asterisk PBX connects to PSTN via a single > SIP trunk. Now, when I transfer or redirect incoming calls from the SIP > trunk to another number which is routed through the SIP trunk, my > asterisk stays on the way; it just dials out the new destination number > the call is transferred/redirected to and connects the newly dialed > channel to the existing incoming channel. > > Since these two channels are in the same SIP trunk, would it be possible > to tell the trunk SIP server to not involve my asterisk anymore, both > for signaling and media data? Or is this inherently not possible via SIP? > > Regards, > Sree > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
