Re: [asterisk-users] One way audio - doesn't seem to be NAT issue - SOLVED!
On Sat, 15 Aug 2015 12:42:38 -0300 Joshua Colp jc...@digium.com wrote: I am not sure why this hasn't bit anyone else. Perhaps most Asterisk systems are in one of two classes, connecting to all NAT phones or connecting to all public phones, and I am in a minority situation where I am talking to a mix of setups. Most people run without direct media unless they know the network topology will allow it 100%. Perhaps but the default is to run it. Perhaps the default should be no to prevent these problems. On the other hand, the documentation seemed to suggest that the default should have worked anyway. One leg was public, the other behind a NAT. It should recognize the latter and not try to put then in direct contact. It's almost like it saw the public one and didn't bother checking the other. Or, it checked both with an OR instead of an AND as I said. That seems more likely since it didn't matter who started the call. I don't really care at this point. If 1% of the calls go through the server when they didn't really need to it's no big deal. Cheers. -- 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] One way audio - doesn't seem to be NAT issue
Not 100% ure, but maybe play with the canreinvite or directmedia settings. On Wed, Aug 12, 2015 at 3:10 AM, D'Arcy J.M. Cain da...@vex.net wrote: I have been banging my head against the wall for weeks now on this one. I have a switch running NetBSD and Asterisk 11.19.0 although I have had this problem on older versions as well. I, and my users, can call out, we can receive calls, quality is excellent but I cannot talk with one user. The different elements are as follows: The switch as described above which is in a server room on the Internet backbone with a public IP address. My home system which is behind a bridged modem through a Linksys WRT54GS with priority given to my ATA. The ATA is a Cisco SPA112. I also have an actual SIP phone. The problem happens with both. Obviously I am using NAT but both devices work just fine if I am going to the PSTN. My user who is also going through a bridged modem to a Linksys SPA-2102 which is doing the PPPOE so it has a public IP address and no NAT involved although it serves NAT for the connected computer. So here is the problem. While both of us have no problems externally, when we call each other we get one way audio and it is always from me to him no matter who initiates the call. A further test, I can call from the SIP phone to the ATA connected phone and vice versa just fine. That involves two devices behind the same NAT but since they still need to use the server as an intermediary I can't see how that would matter. 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. -- 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 -- Michael Dupree jr. p: +1-248-935-4147 f: +1-866-671-6867 Skype: MichaelDupreeJr PGP Pub Key: http://www.michaeldupree.net/?page_id=53 This is a private message. This e-mail message, and any attachments thereto, is for the sole use of the intended recipient(s) and may contain legally privileged and/or confidential information. Any unauthorized review, use, disclosure or distribution is strictly prohibited. If you are not the intended recipient, please contact the sender by reply email and permanently delete all copies of the original message. --- -- _ -- 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 - SOLVED!
On Sat, 15 Aug 2015 16:30:39 +0800 Michael Dupree mich...@easybitllc.com wrote: Not 100% ure, but maybe play with the canreinvite or directmedia settings. Yes! That was it. Just for future searches here is what I did. I added directmedia = no in sip.conf. This fixed the issue. I believe that Asterisk was getting confused when one leg was inside NAT and the other was outside. Perhaps there was an OR where there should be an AND. It makes sense because the other user was the one outside NAT and he could hear me and I could not hear him no matter who initiated the call. He could make outside calls because both he and my provider were on public IPs. I am not sure why this hasn't bit anyone else. Perhaps most Asterisk systems are in one of two classes, connecting to all NAT phones or connecting to all public phones, and I am in a minority situation where I am talking to a mix of setups. Thanks for that. I was going nuts trying to figure this out. -- 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] chan_sip.c: Retransmission timeout reached on transmission
Hi, You must have two thing for start: 1. Set your FW to allow sip port (by default 5060) to your asterisk IP address. 2. Set your asterisk configuration with the external public IP and your local subnet address (so asterisk will put his public address for outside the networks calls) Google for asterisk NAT configuration parameters. נשלח מטלפון נייד בתאריך 14 באוג' 2015 22:12, Daniel - Asterisk earohua...@gmail.com כתב: Hello Sam, Do you have any recommendation to overcome these NAT issues? On 8/14/15, Sam Basan sba...@bluebe.net wrote: Hi, It's looks like you are having NAT problem. Packets from the provider fail reaching your box. נשלח מטלפון נייד בתאריך 14 באוג' 2015 15:56, Daniel - Asterisk earohua...@gmail.com כתב: Hello friends: I am facing cutoffs randomly when negotiating calls. The PBX dials the destination, the provider (softswitch) receives the request *[1]* and sudenly the PBX hangs up the call* [2]* while the provider is still dialing it, as a consequence the remote peer receives a ghost call. Along the atempt I could see six times a messages regarding NAT isuues *[3]* I hope anyone can give me an idea to solve this issue. Softswitch is using an implementation of RFC 3264 and the PBX being used is Elastix 2.3 with Asterisk 1.8.11.0 Thanks in advance Elder D. Arohuanca Lima - Peru *[1]* [Aug 12 19:21:05] VERBOSE[17115] app_dial.c:-- Called SIP/SIP-PROVIDER/965034648 *[2]* [Aug 12 19:21:14] WARNING[3477] chan_sip.c: Retransmission timeout reached on transmission 0e51f669152c660b3c97de1876d9e971@*PROVIDER-IP* for seqno 103 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 8832ms with no response [Aug 12 19:21:14] WARNING[3477] chan_sip.c: Hanging up call 0e51f669152c660b3c97de1876d9e971@*PROVIDER-IP* - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions ). [Aug 12 19:21:14] VERBOSE[17115] app_dial.c: == Everyone is busy/congested at this time (1:0/0/1) [Aug 12 19:21:14] VERBOSE[17115] pbx.c: -- Executing [s@macro-dialout-trunk:20] NoOp(SIP/143-01d8, Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 111) in new stack [Aug 12 19:21:14] VERBOSE[17115] pbx.c: -- Executing [s@macro-dialout-trunk:21] Goto(SIP/143-01d8, s-CHANUNAVAIL,1) in new stack *[3]* Retransmitting #3 (no NAT) to PROVIDER-IP:5060: INVITE sip:dialed_number@PROVIDER-IP SIP/2.0 Via: SIP/2.0/UDP PBX-PUBLIC_IP:5060;branch=z9hG4bK06c2c701 Max-Forwards: 70 From: PBX-DID sip:outbound-trunk@PROVIDER-IP;tag=as27ef83ae To: sip:dialed_number@PROVIDER-IP Contact: sip:outbound-trunk@PBX-PUBLIC_IP:5060 Call-ID: 6b9ad82d4673fdab722f9e53411a767d@PROVIDER-IP CSeq: 103 INVITE User-Agent: FPBX-2.8.1(1.8.11.0) Proxy-Authorization: Digest username=outbound-trunk, realm=SoftSwitch, algorithm=MD5, uri=sip:dialed_number@PROVIDER-IP, nonce=d1b5806808a0888112190722408572932332, response=40c94f3c04e87e3382c7652d1f012dc9 Date: Thu, 13 Aug 2015 00:56:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Remote-Party-ID: PBX-DID sip:PBX-DID@PROVIDER-IP ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 260 v=0 o=root 502733417 502733418 IN IP4 PBX-PUBLIC_IP s=Asterisk PBX 1.8.11.0 c=IN IP4 PBX-PUBLIC_IP t=0 0 m=audio 13042 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv -- _ -- 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 -- _ -- 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 -- _ -- 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 - SOLVED!
On Sat, Aug 15, 2015, at 12:08 PM, D'Arcy J.M. Cain wrote: On Sat, 15 Aug 2015 16:30:39 +0800 Michael Dupree mich...@easybitllc.com wrote: Not 100% ure, but maybe play with the canreinvite or directmedia settings. Yes! That was it. Just for future searches here is what I did. I added directmedia = no in sip.conf. This fixed the issue. I believe that Asterisk was getting confused when one leg was inside NAT and the other was outside. Perhaps there was an OR where there should be an AND. It makes sense because the other user was the one outside NAT and he could hear me and I could not hear him no matter who initiated the call. He could make outside calls because both he and my provider were on public IPs. I am not sure why this hasn't bit anyone else. Perhaps most Asterisk systems are in one of two classes, connecting to all NAT phones or connecting to all public phones, and I am in a minority situation where I am talking to a mix of setups. Most people run without direct media unless they know the network topology will allow it 100%. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com www.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
