Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In
- Original Message - From: Guenther Boelter gboel...@gmail.com To: asterisk-users@lists.digium.com Sent: Tuesday, May 5, 2015 1:05:44 AM Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In Looking into it further, in my case it does not appear to be a NATing issue, since running OpenVPN from pfSense means there's no NATing occurring between the clients or between the clients and the asterisk server. Although I was unable to reproduce the problems, I did notice some packet loss and jitter in sip show channelstats, here is a sample: Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter 192.168.32.26446613544@1 00:03:03 94 004238 (97.83%) 0. 00 000244 ( 0.00%) 0. 192.168.32.385b2ebdc92fd 00:03:03 59 01 ( 1.67%) 0. 00 91 ( 0.00%) 0.0028 I was unable to find documentation each of these columns, but the high percentage of loss for received packets for 192.168.32.26 seems suspicious. Do these statistics indicate a problem? Thanks, Andrew Hi Andrew, is this a linux machine? If so, check your NIC with ifconfig for hardware errors. Guenther Guenther, Yes, this machine is running CentOS 6.4 (see my original post for more details). This asterisk server has 2x gigabit NICs set up in a bond with bond mode 1. Both ifconfig and ethtool do not report any hardware errors, although they do show a few checksum errors: eth0 Link encap:Ethernet HWaddr 00:11:22:33:44:55 UP BROADCAST RUNNING SLAVE MULTICAST MTU:1500 Metric:1 RX packets:467927100 errors:0 dropped:0 overruns:1 frame:0 TX packets:304724661 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:1000 RX bytes:131747094082 (122.6 GiB) TX bytes:93869585242 (87.4 GiB) Memory:fb92-fb94 eth1 Link encap:Ethernet HWaddr AA:BB:CC:DD:EE:FF UP BROADCAST RUNNING SLAVE MULTICAST MTU:1500 Metric:1 RX packets:41250363 errors:0 dropped:0 overruns:0 frame:0 TX packets:3467 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:1000 RX bytes:5190889937 (4.8 GiB) TX bytes:1594075 (1.5 MiB) Memory:fb90-fb92 From ethtool -S eth0: tx_smbus: 164709 rx_smbus: 119082408 dropped_smbus: 104036 rx_queue_0_packets: 97532982 rx_queue_0_bytes: 16800645524 rx_queue_0_drops: 1 rx_queue_0_csum_err: 0 rx_queue_0_alloc_failed: 0 rx_queue_7_packets: 53850556 rx_queue_7_bytes: 12797600155 rx_queue_7_drops: 0 rx_queue_7_csum_err: 41 rx_queue_7_alloc_failed: 0 Thanks, Andrew -- _ -- 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] Authenticated SUBSCRIBE and NOTIFY's R-URI
Hello, I've got a deployment with the SBC in between the clients and Asterisk (11.17.1 version). When the UAC tries to subscribe for dialog event package, the NOTIFY request sent by Asterisk fails. The SBC uses a different Contact (user part) for the 1st and the 2nd SUBSCRIBE (with Auth.). The issue is that Asterisk sends the NOTIFY with R-URI of the first SUBSCRIBE's Contact, not the 2nd one and SBC does not recognise this request, as it would expect the NOTIFY with R-URI containing the 2nd SUSBSCRIBE's Contact. I would say Asterisk should use the 2nd Contact? the log and trace: SBC -- PBX (Asterisk) -- 1st SUBSCRIBE: SUBSCRIBE sip:1...@testing.net SIP/2.0 ... CSeq: 1 SUBSCRIBE Call-ID: 3d28dadd-87e5e749-1b7c8e1d@192.168.1.133 Event: dialog Expires: 3600 Contact: sip:33F18ADD-554124D20004E0F0-6CB68700@10.0.0.32;transport=udp ... [2015-05-04 16:56:50] DEBUG[1948]: chan_sip.c:16341 build_route: build_route: Contact hop: sip:33F18ADD-554124D20004E0F0-6CB68700@10.0.0.32 ;transport=udp -- 401 unauthorized -- 2nd SUBSCRIBE (authenticated): SUBSCRIBE sip:1...@testing.net SIP/2.0 ... CSeq: 2 SUBSCRIBE Call-ID: 3d28dadd-87e5e749-1b7c8e1d@192.168.1.133 Event: dialog Authorization: Digest username=100, realm=asterisk, nonce=69f0a340, uri=sip:1...@testing.net, response=580f1a83fb04d58e2bc5cb9c4c531771, algorithm=MD5 Expires: 3600 Contact: sip:1D3BB238-554124D200064934-6C865700@10.0.0.32;transport=udp [2015-05-04 16:56:50] DEBUG[1948]: chan_sip.c:16259 build_route: build_route: Retaining previous route: sip:1D3BB238-554124D200064934-6C865700@10.0.0.32;transport=udp ... [2015-05-04 16:56:50] DEBUG[1948]: chan_sip.c:11811 reqprep: Strict routing enforced for session ... ... set_destination: Parsing sip:33F18ADD-554124D20004E0F0-6CB68700@10.0.0.32;transport=udp for address/port to send to ... -- 200 OK -- NOTIFY: NOTIFY sip:33F18ADD-554124D20004E0F0-6CB68700@10.0.0.32;transport=udp SIP/2.0 ... Contact: sip:1001@10.0.0.46:5060 Call-ID: 3d28dadd-87e5e749-1b7c8e1d@192.168.1.133 CSeq: 102 NOTIFY -- 491 Call leg/Transaction does not exists Why Asterisk does the build_route: Retaining previous route:... and doesn't update it according to the 2nd SUBSCRIBE? Thanks in advance for any hint, -Vlada -- _ -- 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] Recommendations for IMAP Voicemail
Hello, I'm currently studying what is needed to implement IMAP Voicemail with Asterisk 11 and up. More precisely, I would like to let users check voicemail with their smartphone from outside (ie not connected to LAN). My first questions are: 1. What happens if Asterisk cannot reach its configured IMAP store ? Are voicemails locally stored in a persistent directory surviving reboots or are they lost for ever ? Are voicemails saved back to the IMAP store whenever the IMAP store is back online ? 2. From personal experience, would you rate an IMAP migration as an easy or as a difficult task ? By IMAP migration, I mean changing from one IMAP software to another, on the same or on an other box. Regards -- _ -- 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] OpenVPN Clients Intermittently Cannot Call In
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 05/05/2015 10:59 AM, Andrew Martin wrote: - Original Message - From: Administrator TOOTAI ad...@tootai.net To: asterisk-users@lists.digium.com Sent: Friday, May 1, 2015 6:42:38 AM Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In Le 01/05/2015 00:05, Andrew Martin a écrit : - Original Message - From: Administrator TOOTAI ad...@tootai.net To: asterisk-users@lists.digium.com Sent: Thursday, April 30, 2015 4:43:33 PM Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In I am running Asterisk 11.12.0 on CentOS 6.4. The asterisk server and internal phones are located on the 10.10.32.0/21 LAN subnet. I have many internal SIP phones, which appear to be working correctly. I have a few external phones (Yealink SIP-T32G or other Yealink model) on 192.168.32.0/24 which have an OpenVPN client configured on them that connects back to the LAN network through a pfSense gateway with OpenVPN configured on it. I faced problems with pfsense -no VPN involved- and finally installed siproxd on it. Also set the firewall mode to conservative. Daniel, Thanks for the information. Do you have an example or documentation on the siproxd configuration that you used? No, just follow the basis of the parameters given by the package. If I remember, SIP use the proxy siproxd and RTP is direct. Looking into it further, in my case it does not appear to be a NATing issue, since running OpenVPN from pfSense means there's no NATing occurring between the clients or between the clients and the asterisk server. Although I was unable to reproduce the problems, I did notice some packet loss and jitter in sip show channelstats, here is a sample: Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter 192.168.32.26446613544@1 00:03:03 94 004238 (97.83%) 0. 00 000244 ( 0.00%) 0. 192.168.32.385b2ebdc92fd 00:03:03 59 01 ( 1.67%) 0. 00 91 ( 0.00%) 0.0028 I was unable to find documentation each of these columns, but the high percentage of loss for received packets for 192.168.32.26 seems suspicious. Do these statistics indicate a problem? Thanks, Andrew Hi Andrew, is this a linux machine? If so, check your NIC with ifconfig for hardware errors. Guenther - -- DavaoSOFT, the home of ERPel ERPel, das deutsche Warenwirtschaftssystem fuer LINUX http://www.davaosoft.com -BEGIN PGP SIGNATURE- Version: GnuPG v2 iQIcBAEBAgAGBQJVSF22AAoJENexF5oIz3BC7SUQAL7guaLv8rKHLfJah58/qhT7 qWiyYcjiFiLOUC1J6tgZ+BpT1tXGs5A5NAx+0yC3QWoDHEb/dAg+tzy9YqWqfrtz sePuqAHYPivqtqve1WBM3cB8BGwAL402bQpI8ythpIqJx6RJEFJ8uCQ6eCG/qLjV WKTknHe0r18bV9TTUVmwSHUoU2T/dfz/Wueb/hwjs+ZxrmwiF+jPNeTEr3hUhfFq P1jWi59OMQt01cQbPBmNUogfgiSrN/t7fwitqmbDXK3DoGqviynud1pueigBfONs bboocgqEvx5LZM3Z653VrhjXf38cqPpTwemQ/VVJjRrqWbEHdm5/bT/n2UvT1w3U Nv1Hi/dVPL2/PSuYqW46PqVaqgGYSUAUMRbrrh9ogH2aQAcAw29p4Nl+wK9pHni4 Ix8OFaa4HyefA6a45a+butVGj7tSgZ0k/NYBdsXj9CFBLnBViyB84twINNzDDb9q 6ca1Bhdf8uE6iM4AcyUzcdnoa4L1CA4tBbEwJ2F0lAK4+TWzmGK43Fxy4wctZLim XikVlBeLtGO55cQcI3UZ/IEkYRw7EkXvznNegq4LpXgrPf3pO2n6hNvEZS+uHnC5 q1mY07kCznAI9lU2iCWb9x/YbRpvum4iMy+2Y2ZiuTZd7xI9kZylkKSB/3syIlcv i1nd/nCKQ49nct0agKL3 =xjex -END PGP SIGNATURE- -- _ -- 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
