Eric Kuhnke
2012-Nov-11 10:46 UTC
[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 at 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 at 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* -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20121111/0f24de8a/attachment.htm>
Miguel Oyarzo
2012-Nov-11 12:14 UTC
[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 at 69.71.222.196 <mailto:08a728706baea3b740aa806e41e9d13d at 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 at 69.71.222.196 <mailto:08a728706baea3b740aa806e41e9d13d at 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-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20121111/31e2ad07/attachment.htm>
Markus
2012-Nov-12 00:17 UTC
[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?
Samira Hosseini
2012-Nov-12 06:24 UTC
[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. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20121112/ae2a7477/attachment.htm>
qasimakhan at gmail.com
2012-Nov-12 07:50 UTC
[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 <samiramhosseini at 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20121112/a4bb7e68/attachment.htm>
kingman chui
2012-Nov-12 08:56 UTC
[asterisk-users] 回覆︰ Asterisk SIP authenticate using Radius / LDAP
HI, ? I have connect to Radius with asterisk 1.8.11 before. For CDR, I use the cdr_radius and the cdr can write to radius server. ? For auth with radius server, I use php-radius to write php script and use agi in dialplan to auth the account . ? It is work .. ? ? Regard/chui king man ???? "qasimakhan at gmail.com" <qasimakhan at gmail.com>>???? Samira Hosseini <samiramhosseini at yahoo.com>; Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> >????? 2012?11?12? (??) 3:50 PM >??? 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 <samiramhosseini at 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 > >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20121112/126f8d10/attachment.htm>
kingman chui
2012-Nov-12 10:35 UTC
[asterisk-users] 回覆︰ 回覆︰ Asterisk SIP authenticate using Radius / LDAP
I try before . I use asterisk 1.8.11 , I cannot compile the patch under asterisk 1.8.11 . So, it isn ot work in asterisk 1.8.11 an Use php-radius to write ph p script? to auth radius server .... Please advice other method to auth with radius server under asterisk 1.8.11 if you know ...... ? Thank Regard/chui king man ???? "sami at yahoo.com" <samiramhosseini at yahoo.com>>???? kingman chui <chuikingman at yahoo.com.hk> >????? 2012?11?12? (??) 6:22 PM >??? Re: ??? [asterisk-users] Asterisk SIP authenticate using Radius / LDAP > > >Hello, thanks for your help, >but do you think I will able to connect asterisk(that is installed radius client on it) to the radius server by the following link? >https://issues.asterisk.org/jira/browse/ASTERISK-5278?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel > > > >From: kingman chui <chuikingman at yahoo.com.hk> >>To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com>; Samira Hosseini <samiramhosseini at yahoo.com> >>Sent: Monday, 12 November 2012, 12:26:32 >>Subject: ??? [asterisk-users] Asterisk SIP authenticate using Radius / LDAP >> >> >>HI, >>? I have connect to Radius with asterisk 1.8.11 before. >>For CDR, I use the cdr_radius and the cdr can write to radius server. >>? >>For auth with radius server, I use php-radius to write php script and use agi in dialplan to auth the account . >>? >>It is work .. >>? >>? >>Regard/chui king man >> >> >>???? "qasimakhan at gmail.com" <qasimakhan at gmail.com> >>>???? Samira Hosseini <samiramhosseini at yahoo.com>; Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> >>>????? 2012?11?12? (??) 3:50 PM >>>??? 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 <samiramhosseini at 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 >>> >>> >> >> > >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20121112/471ad53e/attachment.htm>