[asterisk-users] simultaneous use of chan_sip/chan_pjsip
hello, is it possible simultaneously use chan_sip and chan_pjsip? if yes, can you recommend settings i'm thinking about - chan_sip - for sip hardphones/softphones (sip udp 5060) - chan_pjsip - for webrtc -- --- Marek Cervenka === -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] One way audio - doesn't seem to be NAT issue
Hi D'arcy Have you checked your RTP port ranges (I'm sure you have), and also that the server IP for RTP as specified in the initial SIP is correct? Not sure how this will relate to your setup, but we had something similar here using Asterisk 1.8.11.0 on both sides of the connection, via a VOIP service provider in the middle. We had slightly different parameters, e. g. that we would have no RTP at all, but a call that did connect to total silence, dialed from either side. We subscribe to two trunk numbers provided by the VOIP service provider at each site in Asterisk. It turned out after carefully looking at the SIP flowing back and forth that the service provider was providing an RTP server IP that specified not the same IP as the SIP server (which is their standard practice) but a -different- RTP server IP. Due to the routing we have, neither system on either side of the SIP negotiated call could send packets to this new RTP server IP. We therefore added a route that specifically allowed that new RTP server IP to be reached by both machines on both sides of the VOIP service provider link. So can you carefully check that the SIP-negotiated RTP streams are going to IPs that are reachable in BOTH directions? Also check what RTP port ranges are being used - I have had this one-directional problem where the port range in /etc/asterisk/rtp.conf was too broad, and the firewall on my server was only allowing a smaller subset of RTP ports. E. g. /etc/asterisk/rtp.conf specified 1 - 5 as allowable RTP ports, but my firewalld firewall under Centos was only allowing 1 - 2 - so I'd regularly get that my SECOND call to test the server would have audio in one direction - because Asterisk was allocating an RTP port on one side of the SIP call that was outside the range my firewalld was allowing. It might require some careful tracing of SIP messages, maybe you can try this? Specifically try to determine what RTP port number is being negotiated when you have your zero-audio back from the remote party - what RTP port and RTP server IP is he using at that moment on his side? Is that port allowed through all the PPP / network segments between you? Is the IP / IPs between you used to transfer RTP reachable from his side? Message: 1 Date: Tue, 11 Aug 2015 15:10:44 -0400 From: D'Arcy J.M. Cain da...@vex.net To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] One way audio - doesn't seem to be NAT issue Message-ID: 20150811151044.79872ce9@imp Content-Type: text/plain; charset=US-ASCII Given that both of us can make and accept calls and the server is simply connecting two separate channels I can't see where the problem might lie. Can anyone suggest a possible setup issue? I have tried so many things but I am willing to try them again. Feel free to make any suggestion no matter how silly. I really need to fix this. Cheers. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] simultaneous use of chan_sip/chan_pjsip
On Thu, Aug 13, 2015 at 3:54 AM, Marek Červenka cerv...@fpf.slu.cz wrote: hello, is it possible simultaneously use chan_sip and chan_pjsip? if yes, can you recommend settings i'm thinking about - chan_sip - for sip hardphones/softphones (sip udp 5060) - chan_pjsip - for webrtc You can use both.. you will want to make sure your bind addresses and ports don't conflict. Why not use chan_pjsip for all SIP connectivity? -- Rusty Newton Digium, Inc. | Community Support Manager445 Jan Davis Drive NW - Huntsville, AL 35806 - USdirect: +1 256 428 6200 Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One way audio - doesn't seem to be NAT issue
On Thu, 13 Aug 2015 10:41:31 +0200 Stefan Viljoen viljo...@verishare.co.za wrote: Have you checked your RTP port ranges (I'm sure you have), and also Yes. The ATA is using a range well within the range open on the server. that the server IP for RTP as specified in the initial SIP is correct? Both the server and client are outside of NAT so I don't know what this might mean. They both have public IPs. Not sure how this will relate to your setup, but we had something similar here using Asterisk 1.8.11.0 on both sides of the connection, via a VOIP service provider in the middle. This is an Asterisk server talking to an ATA. We had slightly different parameters, e. g. that we would have no RTP at all, but a call that did connect to total silence, dialed from either side. Was NAT involved? Also check what RTP port ranges are being used - I have had this one-directional problem where the port range in /etc/asterisk/rtp.conf was too broad, and the firewall on my server was only allowing a smaller subset of RTP ports. rtpstart=1 rtpend=2 which is exactly what my packet filter allows through. It might require some careful tracing of SIP messages, maybe you can try this? Specifically try to determine what RTP port number is being negotiated when you have your zero-audio back from the remote party - what RTP port and RTP server IP is he using at that moment on his side? I will check that. Thanks for your suggestions. -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/ IM:da...@vex.net VoIP: sip:da...@vex.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] simultaneous use of chan_sip/chan_pjsip
Dne 13.8.2015 v 17:20 Rusty Newton napsal(a): On Thu, Aug 13, 2015 at 3:54 AM, Marek Červenka cerv...@fpf.slu.cz mailto:cerv...@fpf.slu.cz wrote: hello, is it possible simultaneously use chan_sip and chan_pjsip? if yes, can you recommend settings i'm thinking about - chan_sip - for sip hardphones/softphones (sip udp 5060) - chan_pjsip - for webrtc You can use both.. you will want to make sure your bind addresses and ports don't conflict. Why not use chan_pjsip for all SIP connectivity? because it's BIG change for production environment we have own web gui for config generation and we need move to chan_pjsip safely -- --- Marek Cervenka === -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is peer order in sip.conf important?
Hi All Noticed in sip.conf that the asterisk (v11) is sensitive to the order of peers. Here is my sip.conf [general] context = demo ; Default context for incoming calls bindport = 5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr = 0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup = yes ; Enable DNS SRV lookups on outbound calls context=incoming disallow=all allow=ulaw allow=alaw allow=g729 allow=g723 externip=72.220.28.226 localnet=192.168.0.0 nat=yes maxexpiry=15 minexpiry=14 register =16194077214:password@69.59.234.67:5060 [vonage-out] username=16194077214 type=friend secret=password port=5061 nat=yes host=69.59.234.67 fromuser=1619xxx fromdomain=69.59.234.67 dtmfmode=rfc2833 auth=md5 [69.59.234.67] username=1619xxx ;type=friend type=peer ;type=user secret=password port=5061 nat=yes insecure=port,invite host=69.59.234.67 fromuser=1619xxx fromdomain=69.59.234.67 ;dtmfmode=inband context=from-pstn canreinvite=no ;auth=md5 disallow=all allow=ulaw ;allow=alaw ;allow=g729 ;allow=g723 When I make the INBOUD call, vonage-out peer is selected based on the debug. In other words if my sip.conf is as follows [general] [vonage-out] [69.59.234.67] ... Then the peer Asterisk selects is vonage-out. I want vonage-out to be used for OUTBOUND as the name implies. However if I switch them, as follows: [general] ... [69.59.234.67] ... [vonage-out] . Then the peer 69.59.234.67 is selected which is what I want for an INBOUND. Any idea why? Your kind help is appreciated. Best regards murthy -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
