[asterisk-users] port 5060 is blocked by ISP
dear i have configured properly asterisk. At the one end i am using x-lite soft ph and another end twinkle. call is going properly from both end but after picking the phone not able to listen other one. when i checked the port 5060 on the asterisk server it is always showing closed while i have flushed all the rules from iptables (iptables -F) PORT STATE SERVICE VERSION 5060/tcp closed sip telnet localhost 5060 (could not connect) regards alok -- _ -- 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] port 5060 is blocked by ISP
Port 5060 when used with the sip protocol is used witj UDP protocol. Telnet is using TCP. I am typing from my mobile phone... Il giorno 01/lug/2012 09:35, alok srivastava alok...@gmail.com ha scritto: dear i have configured properly asterisk. At the one end i am using x-lite soft ph and another end twinkle. call is going properly from both end but after picking the phone not able to listen other one. when i checked the port 5060 on the asterisk server it is always showing closed while i have flushed all the rules from iptables (iptables -F) PORT STATE SERVICE VERSION 5060/tcp closed sip telnet localhost 5060 (could not connect) regards alok -- _ -- 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] port 5060 is blocked by ISP
No voice means you have to look at the rtp ports. You can find more via google firewall rtp ports asterisk B. Op 1-7-2012 9:34, alok srivastava schreef: dear i have configured properly asterisk. At the one end i am using x-lite soft ph and another end twinkle. call is going properly from both end but after picking the phone not able to listen other one. when i checked the port 5060 on the asterisk server it is always showing closed while i have flushed all the rules from iptables (iptables -F) PORT STATE SERVICE VERSION 5060/tcp closed sip telnet localhost 5060 (could not connect) regards alok -- _ -- 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
[asterisk-users] T.30 Fax session error: Received bad response to DCS or training
Hello, On http://pastebin.com/AP5GBWUR you may see an excerpt from asterisk full log that includes a failing fax sending session. As you can see in line 328, the transmission fails with error Received bad response to DCS or training. It seems that something goes wrong along the lines 315 to 320, but I can't figure it out. Perhaps somebody with enough experience with the T.30 protocol can understand what is happening. What does this DCN 'ff 13 fa' response mean? I would highly appreciate any tip on how to get more info about this error or workaround it. I' m using asterisk 1.8.11.1, as does the other end. -- _ -- 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] T.30 Fax session error: Received bad response to DCS or training
On 07/01/2012 01:46 AM, Petros Moisiadis wrote: Hello, On http://pastebin.com/AP5GBWUR you may see an excerpt from asterisk full log that includes a failing fax sending session. As you can see in line 328, the transmission fails with error Received bad response to DCS or training. It seems that something goes wrong along the lines 315 to 320, but I can't figure it out. Perhaps somebody with enough experience with the T.30 protocol can understand what is happening. What does this DCN 'ff 13 fa' response mean? I would highly appreciate any tip on how to get more info about this error or workaround it. I' m using asterisk 1.8.11.1, as does the other end. DCN is disconnect. So DCN in response to DCS means that the receiver didn't like something about DCS, TCF or possibly TSI so much that it decided to abort the fax by disconnecting after announcing it. Thanks, Lee. -- _ -- 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] Binary packages for Ubuntu Precise
On 12-06-30 03:43 PM, Chris Gentle wrote: On Wed, Jun 13, 2012 at 5:12 AM, Administrator TOOTAI ad...@tootai.netwrote: someone knows when asterisk binary packages will be available on asterisk.org for Ubuntu precise (aka 12.04)? I did a fresh install of Ubuntu Server 12.04 LTS (precise) on my system today and was a bit surprised to find there are still no binary packages for this release. Anybody know when we can expect some? I guess I'll go compile from source ... I no longer maintain the packages at asterisk.org, I'm sure if you are interested in helping package them development might find that helpful. A side from that, we are currently rolling our own Ubuntu Precise packages based on 1.8.7.1, backporting patches we require. Everything is on github[1] if you want to do the same. [1] https://github.com/kickstandproject/asterisk/wiki/Installing-Packages -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] port 5060 is blocked by ISP
On Sun, 2012-07-01 at 13:04 +0530, alok srivastava wrote: dear i have configured properly asterisk. At the one end i am using x-lite soft ph and another end twinkle. call is going properly from both end but after picking the phone not able to listen other one. when i checked the port 5060 on the asterisk server it is always showing closed while i have flushed all the rules from iptables (iptables -F) PORT STATE SERVICE VERSION 5060/tcp closed sip telnet localhost 5060 (could not connect) regards alok Hi Alok, telnet is a very crude tool to test with. Try hping or nmap instead. Hans -- _ -- 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] after upgrade from 1.4.26.2 to 1.8.11.0 my provider gets rport instead of port
if you check out your sip.conf. On Jun 29, 2012, at 5:54 PM, gincantalupo wrote: Hi all, after upgrading my Asterisk 1.4.26.2 to 1.8.11.0 I cannot register to my VoIP provider because it says I'm trying to connect to port 55150 (that's what the call center guy told me)...but I'm not. In my sip I've set port=5060, not 55150. The strange thing is that the rport inside SIP packets (sip set debug) coming back from my provider is set to 55150.seen on both Asterisk 1.4 and 1.8 Does anybody have any idea? Thank you. Giorgio -- _ -- 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