Hi, I'm having a major problem with random calls dropping. After spending weeks trying to figure it out, i've finally spotted the issue but don't know how to resolve it. I run a sip server that's hosted in a data centre. It has a public IP address with no nat involved. My provider also has a public ip with no nat involved. The sip phones are in a remote office behind a nat router. Every so often, all the rtp data coming from the remote location stops arriving at my sip server. So after about 30 seconds, the call gets terminated by my provider because i'm not sending any rtp packets to them. Any ideas why the rtp data should stop coming in, and how can I resolve it? Asterisk v1.4.30 6 x Linksys SPA921 Router at remote site is a Thomson TG585v7 Any assistance will be greatly appreciated. Many thanks Dan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100429/3c2d057f/attachment.htm
i'm having the same problem with one of my call centers located in Egypt.. although the ip-phones are located on a Dedicated Leased Line yet calls drop out of the blue.almost an identical setup as yours..provider located in France (data center) my server located in Sweden (data center) both on public network no NAT.. and the remote office is behind NAT.somehow i suspect Internet problems with your case.. as RTP packets should not stop arriving unless internet connection is timing out. i suppose your calls that are dropping are INBOUND coming from your provider and directed to your remote location.. and you don't have any problems with OUTBOUND calls from your remote location to your server ( I have setup a loop test that goes between 5 locations originating from my remote location and returns to the remote location through 5 hops including IPKALL servers and call goes well with no problem). and let me take a wild guess.. your provider is offering a premium number services.my advise check your internet connection on the remote location and keep a ping from that network to your server running all the time to check for time outs. -- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308 From: dan at keshercommunications.com To: asterisk-users at lists.digium.com Date: Thu, 29 Apr 2010 16:33:06 -0400 Subject: [asterisk-users] Calls Dropping Hi, I?m having a major problem with random calls dropping. After spending weeks trying to figure it out, i?ve finally spotted the issue but don?t know how to resolve it. I run a sip server that?s hosted in a data centre. It has a public IP address with no nat involved. My provider also has a public ip with no nat involved. The sip phones are in a remote office behind a nat router. Every so often, all the rtp data coming from the remote location stops arriving at my sip server. So after about 30 seconds, the call gets terminated by my provider because i?m not sending any rtp packets to them. Any ideas why the rtp data should stop coming in, and how can I resolve it? Asterisk v1.4.30 6 x Linksys SPA921 Router at remote site is a Thomson TG585v7 Any assistance will be greatly appreciated. Many thanks Dan _________________________________________________________________ Hotmail is redefining busy with tools for the New Busy. Get more from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_2 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100430/4cde9b4c/attachment.htm
Hi Bob, Thanks for that. Is there any way I can make the task run in the background and free up the console? Also so that I can disconnect my ssh session without losing the task. Thanks Dan Sent from my Windows Mobile? phone. -----Original Message----- From: Bob Smither <Smither at C-C-I.com> Sent: 02 May 2010 14:04 To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] Calls Dropping On Sun, 2010-05-02 at 08:34 -0400, Dan Journo wrote: <snip>> How can i log a continuous ping test to a file and include the date > and time of each ping?Try this: #!/bin/sh for (( ; ; )) do NOW=$(date +"%T %m/%d/%Y") PING=$(ping -qc 1 example.com) echo $NOW: $PING >> pinger.log done exit 0 You can then monitor the log file using: $ tail -f pinger.log You will need to use ^C to kill the script. Hope this helps. -- _____________________________________________________________________ -- 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
On Sun, 2010-05-02 at 09:52 -0400, Dan Journo wrote:> Hi Bob, > > Thanks for that. Is there any way I can make the task run in the > background and free up the console? Also so that I can disconnect my > ssh session without losing the task. > > Thanks > DanMatthieu NICAISE mentioned screen which should work. Another way would be to activate the script through cron: 1. create a script that does a few pings and e-mails the results. 2. activate the script with cron as often as needed. Once this is setup, you can quit your ssh access to the remote server. Contact me offlist if you need more information. Best regards, Bob Smither