Re: [asterisk-users] Upgrade from version 1.6.24 to 1.8.12 - Retransmission timeout error
Le 02/06/2012 19:18, Administrator TOOTAI a écrit : Le 30/05/2012 15:02, Andres a écrit : Considering that you made progress on your initial problem by setting nat=force_rport (resulting in connected calls with no audio) and now you're mentioning the use of externaddr, I'd recommend a very careful reading of the NAT SUPPORT section of sip.conf.sample in the configs directory of the Asterisk source tree. In Asterisk 1.8, there is a new configuration option named media_address which may be of particular interest. It sounds like a NAT issue to me too. Why don't you do a quick test and put the Asterisk box on a public IP if you can. If it works, you will have narrowed down the issue to a NAT problem. You could have a nat router with a broken SIP ALG. Back to the story: even out of VM -which means on a public IP- the timeout problem till appears. And more odd, if a communication start, the call get hanged up because of this timeout :-( All peers and users are setted with nat=yes, phones connected to Asterisk have directmedia=nonat and peers gateways have directmedia=yes. Remember, we only face this problem with Dellmont services and asterisk 1.8/10. Previous asterisk versions are working well. Does someone else use Dellmont services (VoipBuster, SipDiscount, Intenetcalls, Voicetrading, ...) with asterisk 1.8 or 10? If yes and without problem, would it be possible to share configurations? Thanks for your help. For the archives. Problem was with Dellmont services: no audio or calls stopping after 120 seconds. They gave me another IP for setting outgoing calls and now everything is going smoothly with both versions. Thanks for help. -- Daniel -- _ -- 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] Upgrade from version 1.6.24 to 1.8.12 - Retransmission timeout error
Le 30/05/2012 15:02, Andres a écrit : Considering that you made progress on your initial problem by setting nat=force_rport (resulting in connected calls with no audio) and now you're mentioning the use of externaddr, I'd recommend a very careful reading of the NAT SUPPORT section of sip.conf.sample in the configs directory of the Asterisk source tree. In Asterisk 1.8, there is a new configuration option named media_address which may be of particular interest. It sounds like a NAT issue to me too. Why don't you do a quick test and put the Asterisk box on a public IP if you can. If it works, you will have narrowed down the issue to a NAT problem. You could have a nat router with a broken SIP ALG. Back to the story: even out of VM -which means on a public IP- the timeout problem till appears. And more odd, if a communication start, the call get hanged up because of this timeout :-( All peers and users are setted with nat=yes, phones connected to Asterisk have directmedia=nonat and peers gateways have directmedia=yes. Remember, we only face this problem with Dellmont services and asterisk 1.8/10. Previous asterisk versions are working well. Does someone else use Dellmont services (VoipBuster, SipDiscount, Intenetcalls, Voicetrading, ...) with asterisk 1.8 or 10? If yes and without problem, would it be possible to share configurations? Thanks for your help. -- Daniel -- _ -- 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] Upgrade from version 1.6.24 to 1.8.12 - Retransmission timeout error
Administrator TOOTAI wrote: 10.0.70.12 is the IP of Asterisk server (kvm virtual machine) which is replaced by externaddr parameter from sip.conf. If you have other ideas, welcome ;-) Considering that you made progress on your initial problem by setting nat=force_rport (resulting in connected calls with no audio) and now you're mentioning the use of externaddr, I'd recommend a very careful reading of the NAT SUPPORT section of sip.conf.sample in the configs directory of the Asterisk source tree. In Asterisk 1.8, there is a new configuration option named media_address which may be of particular interest. This is confusing because your first email said you had nat=no in your working 1.6.24 setup, but everything you're saying indicates a NAT problem to me. A diagram showing all network elements between your Asterisk server and the remote host would be helpful. To avoid further confusion, please include full and unaltered logs, SIP traces, and configurations in future posts. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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] Upgrade from version 1.6.24 to 1.8.12 - Retransmission timeout error
Considering that you made progress on your initial problem by setting nat=force_rport (resulting in connected calls with no audio) and now you're mentioning the use of externaddr, I'd recommend a very careful reading of the NAT SUPPORT section of sip.conf.sample in the configs directory of the Asterisk source tree. In Asterisk 1.8, there is a new configuration option named media_address which may be of particular interest. It sounds like a NAT issue to me too. Why don't you do a quick test and put the Asterisk box on a public IP if you can. If it works, you will have narrowed down the issue to a NAT problem. You could have a nat router with a broken SIP ALG. -- Technical Support http://www.cellroute.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] Upgrade from version 1.6.24 to 1.8.12 - Retransmission timeout error
Le 30/05/2012 14:44, Matthew J. Roth a écrit : Considering that you made progress on your initial problem by setting nat=force_rport (resulting in connected calls with no audio) and now you're mentioning the use of externaddr, I'd recommend a very careful reading of the NAT SUPPORT section of sip.conf.sample in the configs directory of the Asterisk source tree. I did read all those documentation, belive me. Also keep in mind that I *ONLY* face this problem with this provider, people using voipbuster or sipdiscount should have the same problem. Concerning externaddr, this test server -dedicated to asterisk- being running in VM since ages, I never would suspect a NAT issue! Especially if previous 1.4 and 1.6 version are running smoothly ... In Asterisk 1.8, there is a new configuration option named media_address which may be of particular interest. media_address seems not an option, can be set only in general not per peer. This is confusing because your first email said you had nat=no in your working 1.6.24 setup, but everything you're saying indicates a NAT problem to me. Again, 1.6 version is perfectly working with this setup and conf files, and before 1.4 was too. And those both asterisk versions with *this* provider. . A diagram showing all network elements between your Asterisk server and the remote host would be helpful. Phone registration: phone (Snom320 and GS GXV3175) - firewall1 (linux router) - Internet - firewall2 (linux router) - VM - phone account Call: phone account - Out of VM - firewall2 (linux router) - Internet - Peer IP - ??? To avoid further confusion, please include full and unaltered logs, SIP traces, and configurations in future posts. During the time you and Andres replied to my post ;-) I got the same idea then him; and guess what, it's working! So problem is Asterisk 1.8/10 in VM _only_ this provider(s) which are all Dellmont services. Can someone confirm the problem? Question is now, who is faulty? Should I open a bug? Thanks for your time and support. -- Daniel -- _ -- 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] Upgrade from version 1.6.24 to 1.8.12 - Retransmission timeout error
Hi Matthew Le 28/05/2012 19:28, Matthew J. Roth a écrit : Administrator TOOTAI wrote: we are upgrading our Asterisk production server from 1.6.24 to 1.8.12 version and face the following problem: one of our peer (voicetrading.com) doesn't accept our calls anymore, we receive a timeout error Packet timed out after 32000ms with no response. Switching back to 1.6 make things working again! In sip.conf we have nat=no, peer conf is: Asterisk 1.8.12 is not getting responses to the INVITES it sends. Comparing the INVITES, the only significant difference I see is that Asterisk 1.6.24 includes the rport field in the Via header and Asterisk 1.8.12 does not: 1.6.24 - Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK58aef527;rport 1.8.12 - Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK0c8907be Try setting nat=force_rport in sip.conf. Please reply back to the list with the results. We tested this setting this WE, effectively this problem disappear but another appears: call get connected but no audio. We installed Asterisk 10.3.1 - connection and no audio too, so same behaviour. There may be other differences between the versions that you haven't accounted for. Read the CHANGES and UPGRADE.txt files in the root of the Asterisk source tree for details. We did read those files, don't see which parameter we could have forget. media_address nor nat=comedia seems options for us. Hereunder a debug from call with force_rport: as you can see, the RTP audio is coming from another IP (77.77.777.77) We think asterisk doesn't accept this and don't know which parameter could solve this. --- SIP read from UDP:111.111.1.111:5060 --- SIP/2.0 200 Ok Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK04e390b0;rport From: TOOTAi sip:00333@222.222.22.22;tag=as1335adb1 To: sip:0336@111.111.1.111;tag=4e0313ac670313ac4f9920c31847cea Contact: sip:0336@111.111.1.111:5060 Call-ID: 72d5d3df06c07cc6037786ee59f574df@222.222.22.22:5060 CSeq: 102 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Type: application/sdp Content-Length: 159 v=0 o=CARRIER 1338276550 1338276550 IN IP4 77.77.777.77 s=SIP Call c=IN IP4 77.77.777.77 t=0 0 m=audio 41462 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 - --- (11 headers 8 lines) --- Found RTP audio format 0 Found audio description format PCMU for ID 0 Capabilities: us - (ulaw|alaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 77.77.777.77:41462 list_route: hop: sip:0336@111.111.1.111:5060 set_destination: Parsing sip:0336@111.111.1.111:5060 for address/port to send to set_destination: set destination to 111.111.1.111:5060 Transmitting (NAT) to 111.111.1.111:5060: ACK sip:0336@111.111.1.111:5060 SIP/2.0 Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK0d106caa;rport Max-Forwards: 70 From: TOOTAi sip:00333@222.222.22.22;tag=as1335adb1 To: sip:0336@111.111.1.111;tag=4e0313ac670313ac4f9920c31847cea Contact: sip:00333@222.222.22.22:5060 Call-ID: 72d5d3df06c07cc6037786ee59f574df@222.222.22.22:5060 CSeq: 102 ACK User-Agent: TOOTAiAudio Content-Length: 0 --- -- SIP/myPeerDef-0003 answered SIP/104-0002 Thanks for your support. -- Daniel -- _ -- 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] Upgrade from version 1.6.24 to 1.8.12 - Retransmission timeout error
Administrator TOOTAI wrote: We tested this setting this WE, effectively this problem disappear but another appears: call get connected but no audio. We installed Asterisk 10.3.1 - connection and no audio too, so same behaviour. We did read those files, don't see which parameter we could have forget. media_address nor nat=comedia seems options for us. Hereunder a debug from call with force_rport: as you can see, the RTP audio is coming from another IP (77.77.777.77) We think asterisk doesn't accept this and don't know which parameter could solve this. Daniel, Asterisk is fine with RTP coming from another IP. It used to work for you on 1.6.24. Here are the relevant bits from the 200 OK responses: 1.6.24 - o=CARRIER 1338135052 1338135052 IN IP4 77.72.168.74 c=IN IP4 77.72.168.74 1.8.12 - o=CARRIER 1338276550 1338276550 IN IP4 77.77.777.77 c=IN IP4 77.77.777.77 Is that really the response that you received? 77.77.777.77 is not a valid IP address (the 3rd octet is greater than 255), so if that's what you're getting than your configuration is fine and the remote end (or some proxy) is now the problem. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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] Upgrade from version 1.6.24 to 1.8.12 - Retransmission timeout error
Le 29/05/2012 14:50, Matthew J. Roth a écrit : Is that really the response that you received? 77.77.777.77 is not a valid IP address (the 3rd octet is greater than 255), so if that's what you're getting than your configuration is fine and the remote end (or some proxy) is now the problem. The IP address is valid, was 77.72.168.29 My bad with the caching stuff in the posted message, sorry. I quit don't understand what happends: I reinstalled a fresh 1.6.24 keeping the parameters from 1.8.13 and 10.3 version and it works! I again installed 10.3 and get again the [2012-05-29 18:06:47] WARNING[17982]: chan_sip.c:3663 retrans_pkt: Retransmission timeout reached on transmission 4fb581df7bc2f11f252f1ebe4718f264@10.0.70.12:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response 10.0.70.12 is the IP of Asterisk server (kvm virtual machine) which is replaced by externaddr parameter from sip.conf. I checked carefully a sip debug from the same call, same conf files, between 1.6.24 and 10.3.1: as with 1.6.24 I receive after the first INVITE a 183 Session progress, on the 10.3.1 I didn't receive it and Asterisk resend the INVITE. Despite this, the call is progressing, the phone on the other end is ringing but when answered, no audio, which seems normal. My guess is that there is misunderstanding between Asterisk and the other end with 1.8.13/10.3 Will try with older version of 1.8 to see if problem is already there ... If you have other ideas, welcome ;-) -- Daniel -- _ -- 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] Upgrade from version 1.6.24 to 1.8.12 - Retransmission timeout error
Administrator TOOTAI wrote: we are upgrading our Asterisk production server from 1.6.24 to 1.8.12 version and face the following problem: one of our peer (voicetrading.com) doesn't accept our calls anymore, we receive a timeout error Packet timed out after 32000ms with no response. Switching back to 1.6 make things working again! In sip.conf we have nat=no, peer conf is: Asterisk 1.8.12 is not getting responses to the INVITES it sends. Comparing the INVITES, the only significant difference I see is that Asterisk 1.6.24 includes the rport field in the Via header and Asterisk 1.8.12 does not: 1.6.24 - Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK58aef527;rport 1.8.12 - Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK0c8907be Try setting nat=force_rport in sip.conf. Please reply back to the list with the results. There may be other differences between the versions that you haven't accounted for. Read the CHANGES and UPGRADE.txt files in the root of the Asterisk source tree for details. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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
