[asterisk-users] Asterisk 11 / FreePBX: Incoming calls timeout after 13 seconds, outgoing works perfectly
Hi all, I'm trying to troubleshoot an issue with my SIP service. All outgoing calls work normally. The following is a SIP debug log from Asterisk. The test setup is as follows: One Yealink SIP-T22P phone (10.0.0.107), extension 10, configured to talk to my local FreePBX/Asterisk 11.0 server which is at 10.0.0.17. The Yealink phone doesn't seem to have any problem placing outgoing calls through the Asterisk server, which is registered to Diamondcard. I can reach both the Asterisk server itself (for example to use voicemail) or call any number on the PSTN. Likewise I have the server configured to pass incoming DID calls for myDIDnumber to extension 10. Calls from the PSTN to myDIDnumber ring the phone, including CID passing, and will connect a full duplex audio call session. The problem is that the phone won't stay connected longer than 13 to 17 seconds. When the phone is manually configured to use my account and password on the diamondcard servers directly, both incoming and outgoing calls work normally, with RTP/UDP port 5060 traffic passing through my NAT without trouble. I have made no special modifications to the NAT. 13 seconds after picking up an incoming call, the phone disconnects at the same time as the log shows this: [2012-11-11 01:51:40] WARNING[3201] chan_sip.c: Retransmission timeout reached on transmission 08a728706baea3b740aa806e41e9d13d@69.71.222.196 for seqno 103 (Critical Response) -- Seehttps://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 17853ms with no response [2012-11-11 01:51:40] WARNING[3201] chan_sip.c: Hanging up call08a728706baea3b740aa806e41e9d13d@69.71.222.196 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). *The full log and configuration is at:* *http://pastebin.com/1Mgn72vN* -- _ -- 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] Asterisk 11 / FreePBX: Incoming calls timeout after 13 seconds, outgoing works perfectly
It seems a firewall or signaling problem. The calling part is not sending a ACK response to your host because it never get an OK 200 from your host. In other words, the called part is trying to send to the calling part a) TRYING 100, then b) RING 180 and finally c) OK 200 but the calling part seems not being receiving no signals from you host. As a result, your host has sent 4 times SIP/2.0 200 OK (retransmissions) to the calling part but it never got an ACK from the other end to establish the communication. Then, the link is destroyed. regards, -- == Miguel Oyarzo Senior [ Network | Systems Design ] Engineer http://www.linkedin.com/in/mikeaustralia Linux User: # 483188 - counter.li.org Melbourne, Australia On 11/11/2012 9:46 PM, Eric Kuhnke wrote: Hi all, I'm trying to troubleshoot an issue with my SIP service. All outgoing calls work normally. The following is a SIP debug log from Asterisk. The test setup is as follows: One Yealink SIP-T22P phone (10.0.0.107), extension 10, configured to talk to my local FreePBX/Asterisk 11.0 server which is at 10.0.0.17. The Yealink phone doesn't seem to have any problem placing outgoing calls through the Asterisk server, which is registered to Diamondcard. I can reach both the Asterisk server itself (for example to use voicemail) or call any number on the PSTN. Likewise I have the server configured to pass incoming DID calls for myDIDnumber to extension 10. Calls from the PSTN to myDIDnumber ring the phone, including CID passing, and will connect a full duplex audio call session. The problem is that the phone won't stay connected longer than 13 to 17 seconds. When the phone is manually configured to use my account and password on the diamondcard servers directly, both incoming and outgoing calls work normally, with RTP/UDP port 5060 traffic passing through my NAT without trouble. I have made no special modifications to the NAT. 13 seconds after picking up an incoming call, the phone disconnects at the same time as the log shows this: [2012-11-11 01:51:40] WARNING[3201] chan_sip.c: Retransmission timeout reached on transmission08a728706baea3b740aa806e41e9d13d@69.71.222.196 mailto:08a728706baea3b740aa806e41e9d13d@69.71.222.196 for seqno 103 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 17853ms with no response [2012-11-11 01:51:40] WARNING[3201] chan_sip.c: Hanging up call 08a728706baea3b740aa806e41e9d13d@69.71.222.196 mailto:08a728706baea3b740aa806e41e9d13d@69.71.222.196 - no reply to our critical packet (seehttps://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). *The full log and configuration is at:* *http://pastebin.com/1Mgn72vN* -- _ -- 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] Asterisk crashing when trying to start a Jabber session with ejabberd
On Sat, 10 Nov 2012 18:08:58 + Phil Reynolds phil-aster...@tinsleyviaduct.com wrote: I have tried today to change my Jabber server from OpenFire to ejabberd. Unfortunately I have not been able to get Asterisk logged in to Jabber since. If SASL is not enabled, then ejabberd rejects the connection as it does not support non-SASL authentication. However, if SASL is enabled, Asterisk crashes when it logs in. This is what appears in CLI when the crash occurs: JABBER: asterisk OUTGOING: ?xml version='1.0'?stream:stream xmlns:stream='http://etherx.jabber.org/streams' xmlns='jabber:client' to='firthpark.tinsleyviaduct.com' version='1.0' JABBER: asterisk INCOMING: ?xml version='1.0'?stream:stream xmlns='jabber:client' xmlns:stream='http://etherx.jabber.org/streams' id='2662035953' from='firthpark.tinsleyviaduct.com' version='1.0' xml:lang='en'stream:featuresstarttls xmlns='urn:ietf:params:xml:ns:xmpp-tls'/mechanisms xmlns='urn:ietf:params:xml:ns:xmpp-sasl'mechanismSCRAM-SHA-1/mechanismmechanismDIGEST-MD5/mechanismmechanismPLAIN/mechanism/mechanismsc xmlns='http://jabber.org/protocol/caps' hash='sha-1' node='http://www.process-one.net/en/ejabberd/' ver='TQ2JFyRoSa70h2G1bpgjzuXb2sU='/register xmlns='http://jabber.org/features/iq-register'//stream:features JABBER: asterisk OUTGOING: starttls xmlns='urn:ietf:params:xml:ns:xmpp-tls'/ JABBER: asterisk INCOMING: proceed xmlns='urn:ietf:params:xml:ns:xmpp-tls'/ JABBER: asterisk OUTGOING: ?xml version='1.0'?stream:stream xmlns:stream='http://etherx.jabber.org/streams' xmlns='jabber:client' to='firthpark.tinsleyviaduct.com' version='1.0' JABBER: asterisk INCOMING: ?xml version='1.0'?stream:stream xmlns='jabber:client' xmlns:stream='http://etherx.jabber.org/streams' id='1069044991' from='firthpark.tinsleyviaduct.com' version='1.0' xml:lang='en' JABBER: asterisk INCOMING: stream:featuresmechanisms xmlns='urn:ietf:params:xml:ns:xmpp-sasl'mechanismSCRAM-SHA-1/mechanismmechanismDIGEST-MD5/mechanismmechanismPLAIN/mechanism/mechanismsc xmlns='http://jabber.org/protocol/caps' hash='sha-1' node='http://www.process-one.net/en/ejabberd/' ver='TQ2JFyRoSa70h2G1bpgjzuXb2sU='/register xmlns='http://jabber.org/features/iq-register'//stream:features JABBER: asterisk OUTGOING: auth xmlns='urn:ietf:params:xml:ns:xmpp-sasl' mechanism='PLAIN'AGFzdGVyaXNrAGNsaWQvaW0=/auth JABBER: asterisk INCOMING: success xmlns='urn:ietf:params:xml:ns:xmpp-sasl'/ JABBER: asterisk OUTGOING: ?xml version='1.0'?stream:stream xmlns:stream='http://etherx.jabber.org/streams' xmlns='jabber:client' to='firthpark.tinsleyviaduct.com' version='1.0' JABBER: asterisk INCOMING: ?xml version='1.0'?stream:stream xmlns='jabber:client' xmlns:stream='http://etherx.jabber.org/streams' id='3920780923' from='firthpark.tinsleyviaduct.com' version='1.0' xml:lang='en' JABBER: asterisk INCOMING: stream:featuresbind xmlns='urn:ietf:params:xml:ns:xmpp-bind'/session xmlns='urn:ietf:params:xml:ns:xmpp-session'/c xmlns='http://jabber.org/protocol/caps' hash='sha-1' node='http://www.process-one.net/en/ejabberd/' ver='TQ2JFyRoSa70h2G1bpgjzuXb2sU='/register xmlns='http://jabber.org/features/iq-register'//stream:features JABBER: asterisk OUTGOING: iq type='set' id='a'bind xmlns='urn:ietf:params:xml:ns:xmpp-bind'resourceasterisk/resource/bind/iq JABBER: asterisk OUTGOING: iq type='set' id='auth'session xmlns='urn:ietf:params:xml:ns:xmpp-session'//iq JABBER: asterisk INCOMING: iq id='a' type='result'bind xmlns='urn:ietf:params:xml:ns:xmpp-bind'jidaster...@firthpark.tinsleyviaduct.com/asterisk/jid/bind/iq JABBER: asterisk OUTGOING: presence from='aster...@firthpark.tinsleyviaduct.com/asterisk'statusasterisk running/statuspriority0/priorityc node='http://www.asterisk.org/xmpp/client/caps' ver='asterisk-xmpp' ext='voice-v1' xmlns='http://jabber.org/protocol/caps'//presence JABBER: asterisk OUTGOING: iq type='get' id='roster'query xmlns='jabber:iq:roster'//iq JABBER: asterisk INCOMING: iq type='result' id='auth'session xmlns='urn:ietf:params:xml:ns:xmpp-session'//iq JABBER: asterisk INCOMING: iq from='firthpark.tinsleyviaduct.com' to='aster...@firthpark.tinsleyviaduct.com/asterisk' id='1072575207' type='get'query xmlns='http://jabber.org/protocol/disco#info' node='http://www.asterisk.org/xmpp/client/caps#asterisk-xmpp'//iq JABBER: asterisk OUTGOING: iq from='aster...@firthpark.tinsleyviaduct.com/asterisk' to='firthpark.tinsleyviaduct.com' type='result' id='1072575207'query xmlns='http://jabber.org/protocol/disco#info'identity category='client' type='pc' name='asterisk'/feature var='http://www.google.com/xmpp/protocol/voice/v1'/feature var='http://jabber.org/protocol/disco#info'//query/iq -- Phil Reynolds mail: phil-aster...@tinsleyviaduct.com Web: http://phil.tinsleyviaduct.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live
Re: [asterisk-users] Asterisk 11 / FreePBX: Incoming calls timeout after 13 seconds, outgoing works perfectly
Am 11.11.2012 11:46, schrieb Eric Kuhnke: I'm trying to troubleshoot an issue with my SIP service. All outgoing calls work normally. The following is a SIP debug log from Asterisk. The test setup is as follows: Miguel already explained what's going on. Have a look at the SIP packets to figure out more. On the Asterisk box: tcpdump -nnqt -s 0 -A -i eth0 port 5060 Also, check your router/firewall logs, respectively activate them, to find out why the packets are not going through. Maybe also try qualify=yes in your Diamondcard SIP stanza to help keep NAT open? Is this a D-Link router? -- _ -- 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] Asterisk 11 / FreePBX: Incoming calls timeout after 13 seconds, outgoing works perfectly
On Mon, Nov 12, 2012 at 11:17 AM, Markus unive...@truemetal.org wrote: Am 11.11.2012 11:46, schrieb Eric Kuhnke: I'm trying to troubleshoot an issue with my SIP service. All outgoing calls work normally. The following is a SIP debug log from Asterisk. The test setup is as follows: Miguel already explained what's going on. Have a look at the SIP packets to figure out more. On the Asterisk box: tcpdump -nnqt -s 0 -A -i eth0 port 5060 Also, check your router/firewall logs, respectively activate them, to find out why the packets are not going through. Maybe also try qualify=yes yes, correct. In addition, He might being notifying to the calling part incorrectly about the called part is behind a nat :) http://pastebin.com/1Mgn72vN (line 71: nat=no) -- ==** Miguel Oyarzo Senior Systems Design Engineer Linux User: # 483188 - counter.li.org http://au.linkedin.com/in/mikeaustralia Melbourne, Australia ==** -- _ -- 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] Asterisk SIP authenticate using Radius / LDAP
Hi all, based on the following link, I am going to authenticate SIP asterisk users via Radius client that is installed on my Asterisk then the radius client connect to asterisk using the radius and ldap: https://who.rocq.inria.fr/Philippe.Sultan/Asterisk/asterisk_sip_external_authentication.html#AEN237 So I want to know for implementing the mentioned authentication method I need to use the patched asterisk as follow : https://issues.asterisk.org/jira/browse/ASTERISK-5278?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel Thanks.-- _ -- 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] Asterisk SIP authenticate using Radius / LDAP
You can use Radius Agi developed by PortaOne from following link. http://www.voip-info.org/wiki/view/PortaOne+Radius+auth Regards, Qasim On Mon, Nov 12, 2012 at 11:24 AM, Samira Hosseini samiramhosse...@yahoo.com wrote: Hi all, based on the following link, I am going to authenticate SIP asterisk users via Radius client that is installed on my Asterisk then the radius client connect to asterisk using the radius and ldap: https://who.rocq.inria.fr/Philippe.Sultan/Asterisk/asterisk_sip_external_authentication.html#AEN237 So I want to know for implementing the mentioned authentication method I need to use the patched asterisk as follow : https://issues.asterisk.org/jira/browse/ASTERISK-5278?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel Thanks. -- _ -- 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] Need help for Aculab Prosody X PCI card installation and configuration with Asterisk
Hi All, I need to install and configuration of Aculab prosody X PCI card with Asterisk-1.8.9.1 on Centos-5.7 system. I will try for that but not success. so, please suggest me way to achieve it. Thanks in Advance. -- Best Regards, Rajni Vanza -- _ -- 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] Web based Click to Call Application
its too early for webrtc. im also waiting for further development on this. On Sun, Nov 11, 2012 at 3:51 AM, Joshua Colp jc...@digium.com wrote: Adolphe Cher-Aime wrote: Hi Marcus, You're right,WebRTC is the way to go. The only drawback is the fact that only astersik 11 support it natively. It's also not yet finished. Specifications are still being discussed, finalized, completed. Implementations have certainly come a long way but are still in a great state of flux. It's early for WebRTC. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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-**usershttp://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
