Re: [asterisk-users] Calls Dropping
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 Dan Matthieu 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 -- _ -- 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] Calls Dropping
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. How can i log a continuous ping test to a file and include the date and time of each ping? I've found this bash code but it only logs once the tests have all finished. If I set it to continuous and then kill the task when I want to view the pings, it doesn't record the data. #!/bin/sh NOW=$(date +%T %m/%d/%Y) PING=$(ping -qc 5 example.com | grep '5 packets') echo $NOW: $PING /home/matt/ping.log exit 0 Thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls Dropping
On Sun, 2010-05-02 at 08:34 -0400, Dan Journo wrote: snip How can i log a continuous ping test to a file and include the date and time of each ping? Try this: #!/bin/sh for (( ; ; )) do NOW=$(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
Re: [asterisk-users] Calls Dropping
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 smit...@c-c-i.com Sent: 02 May 2010 14:04 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@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 -- _ -- 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] Calls Dropping
Hi, try using screen : http://www.rackaid.com/resources/linux-screen-tutorial-and-how-to/ I think it's the best way of doing this. Regards, Matthieu NICAISE Responsable technique GSM : 06 72 19 09 55 techni...@thinkrosystem.com Thinkro System http://www.thinkrosystem.com/ Le 2 mai 10 à 15:52, Dan Journo a écrit : 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 smit...@c-c-i.com Sent: 02 May 2010 14:04 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@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 -- _ -- 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] Calls Dropping
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: d...@keshercommunications.com To: asterisk-users@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-- _ -- 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] Calls Dropping
The info you need is here http://www.voip-info.org/wiki/view/Asterisk+config+logger.conf Ish Dan Journo wrote: Hello, We have a problem that calls seem to be dropping for no reason. Is there any way to write a debug log to disk so that I can check it as soon as a call is lost? It happens randomly once or twice a day to different callers. Many thanks Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls Dropping
On 11 Dec 2009, at 11:19, Dan Journo wrote: Is there any way to write a debug log to disk so that I can check it as soon as a call is lost? It happens randomly once or twice a day to different callers. /var/log/asterisk/full? Most 'standard' setups produce it. Failing that google will reveal how to do this. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls Dropping
Thanks. I didnt stop that. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik Sent: 11 December 2009 11:47 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calls Dropping The info you need is here http://www.voip-info.org/wiki/view/Asterisk+config+logger.conf Ish Dan Journo wrote: Hello, We have a problem that calls seem to be dropping for no reason. Is there any way to write a debug log to disk so that I can check it as soon as a call is lost? It happens randomly once or twice a day to different callers. Many thanks Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls dropping
Hi,
I am using Originate in testing and also using call files in testing. I also
needed to capture DIALSTATUS and update my CDRs accordingly. My original
attempts at using Originate (or call files) did not report {DIALSTATUS} if
the call could not be connected (e.g. bad phone number like 555-555-)
because, it seems, the call never entered the dialplan context defined in
Originate. So, my latest code uses Local/mycontext/n
So it appears, intermittently, the channel disappears (fails?). This is the
output of an instance of failure. I have also listed the code from my
context and the AMI call. A more readable version is attached. I am 1.6.1.1
- Thanks for looking...
-- Executing [dialnum...@dialthem_private:2]
Dial(Local/dialnum...@dialthem_private-c64a;2,
SIP/1211...@flowroute,30,ghM(dialthem_private^callJohnSmith^SIP/121
1...@flowroute)) in new stack
== Using SIP RTP CoS mark 5
-- Called 1211...@flowroute
-- SIP/flowroute-0821ffc8 is making progress passing it to
Local/dialnum...@dialthem_private-c64a;2
== Spawn extension (dialthem_private, dialnumber, 2) exited non-zero on
'Local/dialnum...@dialthem_private-c64a;2'
[Jun 30 19:52:01] ERROR[26664]: pbx.c:8637 device_state_cb: Received invalid
event that had no device IE
[Jun 30 19:52:01] ERROR[26664]: app_queue.c:810 device_state_cb: Received
invalid event that had no device IE
[dialthem_private]
exten = dialnumber,1,UserEvent(BeforeDial,ActionID:${INSP_ActionID}
${UNIQUEID} ${CHANNEL} ${INSP_DialInfo} ${INSP_$
exten =
dialnumber,n,Dial(${INSP_DialInfo},${INSP_RingTimeout},ghM(INSP_private^${IN
SP_ActionID}^${INSP_DialInfo}))
exten = dialnumber,n,UserEvent(AfterDial,ActionID:${INSP_ActionID}
${UNIQUEID} ${CHANNEL} ${INSP_DialInfo} ${DIALST$
exten = dialnumber,n,Hangup()
[macro-INSP_private]
exten = s,1,UserEvent(SIPDial,ActionID:${ARG1} ${UNIQUEID} ${CHANNEL}
${ARG2})
http://192.168.1.2:8088/asterisk/rawman?action=Originatechannel=Local%2Fdia
lnumber%40dialthem_private%2Fnexten=scontext=detectpriority=1CallerID=19
9async=1actionID=callingJohnSmithaccount=myaccountvaluevariable=
INSP_ActionID%3DcallJohnvariable=INSP_DialInfo%3DSIP%2F121%40flowro
utevariable=INSP_RingTimeout%3D30variable=phonenumber%3D21variabl
e=file%3DFA3469AC-BCDE-E6EB-B3AA936266704744variable=alertID%3DFA3469AC-BCD
E-E6EB-B3AA936266704744variable=subscriberID%3D1234512345
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Backeberg
Sent: Friday, June 26, 2009 12:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Calls dropping
On Thu, Jun 25, 2009 at 9:55 PM, John Regaljre...@gmail.com wrote:
When using this method, it appears that the call file creates the first
part
of the call, then creates a second call with the Dial() app. Once the call
executed by the Dial() app is answered, the two calls are joined together.
What I am experiencing is that sometime the first part of the call drops
and
therefore is never joined to the second part of the call. I see errors
like
I don't quite understand what you're trying to do, but it sounds like
call two parties and join them together. Perhaps you'd prefer to use
Originate() via AMI rather than the dialplan and extension approach
you're using now?
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-- Executing [dialnum...@dialthem_private:2]
Dial(Local/dialnum...@dialthem_private-c64a;2,
SIP/1211...@flowroute,30,ghM(dialthem_private^callJohnSmith^SIP/1211...@flowroute))
in new stack
== Using SIP RTP CoS mark 5
-- Called 1211...@flowroute
-- SIP/flowroute-0821ffc8 is making progress passing it to
Local/dialnum...@dialthem_private-c64a;2
== Spawn extension (dialthem_private, dialnumber, 2) exited non-zero on
'Local/dialnum...@dialthem_private-c64a;2'
[Jun 30 19:52:01] ERROR[26664]: pbx.c:8637 device_state_cb: Received invalid
event that had no device IE
[Jun 30 19:52:01] ERROR[26664]: app_queue.c:810 device_state_cb: Received
invalid event that had no device IE
[dialthem_private]
exten = dialnumber,1,UserEvent(BeforeDial,ActionID:${INSP_ActionID}
${UNIQUEID} ${CHANNEL} ${INSP_DialInfo} ${INSP_$
exten =
dialnumber,n,Dial(${INSP_DialInfo},${INSP_RingTimeout},ghM(INSP_private^${INSP_ActionID}^${INSP_DialInfo}))
exten = dialnumber,n,UserEvent(AfterDial,ActionID:${INSP_ActionID}
${UNIQUEID} ${CHANNEL} ${INSP_DialInfo} ${DIALST$
exten = dialnumber,n,Hangup()
[macro-INSP_private]
exten = s,1,UserEvent(SIPDial,ActionID:${ARG1} ${UNIQUEID} ${CHANNEL}
${ARG2})
http://192.168.1.2:8088/asterisk/rawman?action=Originatechannel=Local%2Fdialnumber%40dialthem_private%2Fnexten=scontext=detectpriority
Re: [asterisk-users] Calls dropping
On Thu, Jun 25, 2009 at 9:55 PM, John Regaljre...@gmail.com wrote: When using this method, it appears that the call file creates the first part of the call, then creates a second call with the Dial() app. Once the call executed by the Dial() app is answered, the two calls are joined together. What I am experiencing is that sometime the first part of the call drops and therefore is never joined to the second part of the call. I see errors like I don't quite understand what you're trying to do, but it sounds like call two parties and join them together. Perhaps you'd prefer to use Originate() via AMI rather than the dialplan and extension approach you're using now? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls dropping...
Carlos Chavez wrote: I have a customer that recently started having a problem with their Call Center SIP extensions. The problem is that after some time the caller will hear a triple tone (beep, beep, beep), a 5 second pause, another triple tone and then the call will be dropped. This usually happens between the 8 an 10 minute mark. Until Tuesday we were running Asterisk 1.4.11 but I decided to upgrade to 1.4.12.1 just in case this was a bug with earlier versions. The problem only started recently, about a week ago. We have not made any significant changes to the configuration, mostly just dialplan changes so we do not know what exactly is causing this. Any ideas? Just curious, is this a calling card or prepaid kind of call center? Your configs and logs might help out alot. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calls dropping off
I did have busydetect turned on, but not callprogress. I've turned off busydetect and I'll see how it goes. Many thanks. On Tue, Feb 10, 2004 at 02:30:32PM -0600, Eric Wieling wrote: That sounds like a classic issue of busydetect=yes and callprogress=yes in zapata.conf. Don't do that. Set them to no On Tue, 2004-02-10 at 14:16, Tomica Crnek wrote: Might be, but even if you are not using voip, calls drop. I have a 2 E1 links and bridged calls between them drop from time to time. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ejay Hire Sent: Tuesday, February 10, 2004 5:46 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Calls dropping off I have this problem intermittently, and doing an asterisk -r showed too many retries. hunting around with ethereal found a bad hub. -e -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomica Crnek Sent: Tuesday, February 10, 2004 9:23 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Calls dropping off Last 2 days I have noticed that more and more often calls are just being dropped. I can't find any logs or anything indicating that something is wrong. If I do a trace and wait for a call to drop I can only see hangup and nothing else. Sometimes calls do last for minutes without problem and sometimes they are dropped after about 30 seconds. Until yesterday it worked fine. I am using TE410P with 2 E1 connected trunks with h.323, sip and skinny phones on voip side. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Foy Sent: Monday, February 09, 2004 3:35 PM To: Michael Nigrelli Cc: Asterisk-Users Subject: Re: [Asterisk-Users] Calls dropping off On Mon, Feb 09, 2004 at 09:26:43AM -0500, Michael Nigrelli wrote: Steve, Did you ever figure out why this happens. I have had asterisk up and running for a few weeks and all of a sudden this started happening. Exactly the same here, it was running fine for about a month or so. Then one day, a call disappeared, and gradually got more more frequent. Nothing appears in logs or console. What phones are you using? -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Go to http://www.digium.com/index.php?menu=documentation and look at the Unofficial Links section. This section has links to a wide variety of 3rd party Asterisk related pages. My page is the Asterisk Resource Pages. BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Calls dropping off
Last 2 days I have noticed that more and more often calls are just being dropped. I can't find any logs or anything indicating that something is wrong. If I do a trace and wait for a call to drop I can only see hangup and nothing else. Sometimes calls do last for minutes without problem and sometimes they are dropped after about 30 seconds. Until yesterday it worked fine. I am using TE410P with 2 E1 connected trunks with h.323, sip and skinny phones on voip side. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Foy Sent: Monday, February 09, 2004 3:35 PM To: Michael Nigrelli Cc: Asterisk-Users Subject: Re: [Asterisk-Users] Calls dropping off On Mon, Feb 09, 2004 at 09:26:43AM -0500, Michael Nigrelli wrote: Steve, Did you ever figure out why this happens. I have had asterisk up and running for a few weeks and all of a sudden this started happening. Exactly the same here, it was running fine for about a month or so. Then one day, a call disappeared, and gradually got more more frequent. Nothing appears in logs or console. What phones are you using? -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Calls dropping off
I have this problem intermittently, and doing an asterisk -r showed too many retries. hunting around with ethereal found a bad hub. -e -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomica Crnek Sent: Tuesday, February 10, 2004 9:23 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Calls dropping off Last 2 days I have noticed that more and more often calls are just being dropped. I can't find any logs or anything indicating that something is wrong. If I do a trace and wait for a call to drop I can only see hangup and nothing else. Sometimes calls do last for minutes without problem and sometimes they are dropped after about 30 seconds. Until yesterday it worked fine. I am using TE410P with 2 E1 connected trunks with h.323, sip and skinny phones on voip side. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Foy Sent: Monday, February 09, 2004 3:35 PM To: Michael Nigrelli Cc: Asterisk-Users Subject: Re: [Asterisk-Users] Calls dropping off On Mon, Feb 09, 2004 at 09:26:43AM -0500, Michael Nigrelli wrote: Steve, Did you ever figure out why this happens. I have had asterisk up and running for a few weeks and all of a sudden this started happening. Exactly the same here, it was running fine for about a month or so. Then one day, a call disappeared, and gradually got more more frequent. Nothing appears in logs or console. What phones are you using? -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Calls dropping off
Might be, but even if you are not using voip, calls drop. I have a 2 E1 links and bridged calls between them drop from time to time. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ejay Hire Sent: Tuesday, February 10, 2004 5:46 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Calls dropping off I have this problem intermittently, and doing an asterisk -r showed too many retries. hunting around with ethereal found a bad hub. -e -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomica Crnek Sent: Tuesday, February 10, 2004 9:23 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Calls dropping off Last 2 days I have noticed that more and more often calls are just being dropped. I can't find any logs or anything indicating that something is wrong. If I do a trace and wait for a call to drop I can only see hangup and nothing else. Sometimes calls do last for minutes without problem and sometimes they are dropped after about 30 seconds. Until yesterday it worked fine. I am using TE410P with 2 E1 connected trunks with h.323, sip and skinny phones on voip side. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Foy Sent: Monday, February 09, 2004 3:35 PM To: Michael Nigrelli Cc: Asterisk-Users Subject: Re: [Asterisk-Users] Calls dropping off On Mon, Feb 09, 2004 at 09:26:43AM -0500, Michael Nigrelli wrote: Steve, Did you ever figure out why this happens. I have had asterisk up and running for a few weeks and all of a sudden this started happening. Exactly the same here, it was running fine for about a month or so. Then one day, a call disappeared, and gradually got more more frequent. Nothing appears in logs or console. What phones are you using? -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Calls dropping off
That sounds like a classic issue of busydetect=yes and callprogress=yes in zapata.conf. Don't do that. Set them to no On Tue, 2004-02-10 at 14:16, Tomica Crnek wrote: Might be, but even if you are not using voip, calls drop. I have a 2 E1 links and bridged calls between them drop from time to time. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ejay Hire Sent: Tuesday, February 10, 2004 5:46 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Calls dropping off I have this problem intermittently, and doing an asterisk -r showed too many retries. hunting around with ethereal found a bad hub. -e -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomica Crnek Sent: Tuesday, February 10, 2004 9:23 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Calls dropping off Last 2 days I have noticed that more and more often calls are just being dropped. I can't find any logs or anything indicating that something is wrong. If I do a trace and wait for a call to drop I can only see hangup and nothing else. Sometimes calls do last for minutes without problem and sometimes they are dropped after about 30 seconds. Until yesterday it worked fine. I am using TE410P with 2 E1 connected trunks with h.323, sip and skinny phones on voip side. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Foy Sent: Monday, February 09, 2004 3:35 PM To: Michael Nigrelli Cc: Asterisk-Users Subject: Re: [Asterisk-Users] Calls dropping off On Mon, Feb 09, 2004 at 09:26:43AM -0500, Michael Nigrelli wrote: Steve, Did you ever figure out why this happens. I have had asterisk up and running for a few weeks and all of a sudden this started happening. Exactly the same here, it was running fine for about a month or so. Then one day, a call disappeared, and gradually got more more frequent. Nothing appears in logs or console. What phones are you using? -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Go to http://www.digium.com/index.php?menu=documentation and look at the Unofficial Links section. This section has links to a wide variety of 3rd party Asterisk related pages. My page is the Asterisk Resource Pages. BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calls dropping off
On Mon, Feb 09, 2004 at 09:26:43AM -0500, Michael Nigrelli wrote: Steve, Did you ever figure out why this happens. I have had asterisk up and running for a few weeks and all of a sudden this started happening. Exactly the same here, it was running fine for about a month or so. Then one day, a call disappeared, and gradually got more more frequent. Nothing appears in logs or console. What phones are you using? -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calls dropping off
On Fri, Feb 06, 2004 at 08:18:21PM -0500, Andres wrote: Feb 5 14:34:18 WARNING[1133742896]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 3 (Response) So did it drop a few seconds into the call...like 5 - 15 seconds? If so then you are having a problem with call setup. I would guess it is the ACK that is not receiving a STATUS 200 OK so Asterisk cuts off the call. No, they drop at random points in the calls. Sometimes after 30 seconds, sometimes up to 5 minutes :( Steve -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calls dropping off
I would have thought that if that was the problem, we couldn't makle or receive calls at all, or that we at least couldnt use all 3 Zap cards at the same time, but we can. The problem only happens every so often, but recently it's getting more and more frequent... management are starting to get pissed :/ No more ideas? I've tried everything else people have mentioned. Cheers, Steve On Mon, Feb 02, 2004 at 01:03:01PM -0500, Bill Hamel wrote: Hi, Have you checked for IRQ conflicts ? -b Quoting Steve Foy [EMAIL PROTECTED]: Hi, On Mon, Feb 02, 2004 at 06:04:40PM +0100, Philipp von Klitzing wrote: Steve, this really is a FAQ. You need add to EACH (!) sip user something like disallow=all allow=ulaw allow=alaw allow=gsm I do have that in my sip.conf. I am using ulaw. Calls from the SIP phones through Asterisk and out one of my X100P cards are working 95% of the time and also, incoming calls through the X100P cards to the SIP phones are the same. The only problem is that every once in a while, without any odd circustances that I can see, the call just drops and the remote user is gone. The box running asterisk isn't under heavy load, so I can't see why this is happening. I am not using g.729 or 723, just plain old ulaw, which I have got enabled in sip.conf Cheers, Steve -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by The CCIS.net MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calls dropping off
Steve, Since I have a rather short memory and receive about 250 posting per day, I don't have a clue what has/hasn't been suggested. Here's a couple: 1. in logger.conf turn on debug, watch /var/log/asterisk/debug for size, and and hints relative to the dropped calls 2. look at /var/log/asterisk/messages for hints 3. if the problem occurs frequently enough, start a ping from the * box to one or more of the sip phones to verify you're not loosing net connections at the time of the dropped call (Spanning Tree Protocol can mess with your infrastructure without you knowing it, as one example) 4. look in /var/log/asterisk/cdr-csv/Master.csv file to see if any hints in the cdr data 5. post a relavent definition from sip.conf so we have a clue how you've defined a phone, as well as a relative Dial section from extensions.conf and zapata.conf 6. I don't recall which sip phones you're using, but some have internal logging capabilities. If your's do, turn it on and look for hints. 7. Download ethereal and sniff the asterisk nic interface, ensure you stop it right after a failure. If you need help doing the protocol analysis, then let me know. Rich I would have thought that if that was the problem, we couldn't makle or receive calls at all, or that we at least couldnt use all 3 Zap cards at the same time, but we can. The problem only happens every so often, but recently it's getting more and more frequent... management are starting to get pissed :/ No more ideas? I've tried everything else people have mentioned. Cheers, Steve On Mon, Feb 02, 2004 at 01:03:01PM -0500, Bill Hamel wrote: Hi, Have you checked for IRQ conflicts ? -b Quoting Steve Foy [EMAIL PROTECTED]: Hi, On Mon, Feb 02, 2004 at 06:04:40PM +0100, Philipp von Klitzing wrote: Steve, this really is a FAQ. You need add to EACH (!) sip user something like disallow=all allow=ulaw allow=alaw allow=gsm I do have that in my sip.conf. I am using ulaw. Calls from the SIP phones through Asterisk and out one of my X100P cards are working 95% of the time and also, incoming calls through the X100P cards to the SIP phones are the same. The only problem is that every once in a while, without any odd circustances that I can see, the call just drops and the remote user is gone. The box running asterisk isn't under heavy load, so I can't see why this is happening. I am not using g.729 or 723, just plain old ulaw, which I have got enabled in sip.conf Cheers, Steve -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by The CCIS.net MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calls dropping off
Right... It just happened there now, this came up: Feb 5 14:34:18 WARNING[1133742896]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 3 (Response) I'm not sure if that's related to it, but it's the only thing that came up when the call got cut off. Here's the generic sip.conf stuff [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) allow=all allow=GSM allow=G729 allow=iLBC allow=SpeeX; Allow all codecs allow=ulaw Here's a sip.conf declaration: ; Andy [108] type=friend username= secret= host=dynamic dtmfmode=rfc2833 callerid=Andy McAlister 108 context=internal [EMAIL PROTECTED] qualify=yes canreinvite=no And the relevant extension.conf bit: ;Andy exten = 108,1,Dial(SIP/108,15) exten = 108,2,Playback(int-voicemail/108) exten = 108,3,Voicemail(s108) exten = 108,102,Playback(int-voicemail/108) exten = 108,103,Voicemail(s108) Any insight vastly appreciated! Cheers, Steve On Thu, Feb 05, 2004 at 06:33:06AM -0600, Rich Adamson wrote: Steve, Since I have a rather short memory and receive about 250 posting per day, I don't have a clue what has/hasn't been suggested. Here's a couple: 1. in logger.conf turn on debug, watch /var/log/asterisk/debug for size, and and hints relative to the dropped calls 2. look at /var/log/asterisk/messages for hints 3. if the problem occurs frequently enough, start a ping from the * box to one or more of the sip phones to verify you're not loosing net connections at the time of the dropped call (Spanning Tree Protocol can mess with your infrastructure without you knowing it, as one example) 4. look in /var/log/asterisk/cdr-csv/Master.csv file to see if any hints in the cdr data 5. post a relavent definition from sip.conf so we have a clue how you've defined a phone, as well as a relative Dial section from extensions.conf and zapata.conf 6. I don't recall which sip phones you're using, but some have internal logging capabilities. If your's do, turn it on and look for hints. 7. Download ethereal and sniff the asterisk nic interface, ensure you stop it right after a failure. If you need help doing the protocol analysis, then let me know. Rich I would have thought that if that was the problem, we couldn't makle or receive calls at all, or that we at least couldnt use all 3 Zap cards at the same time, but we can. The problem only happens every so often, but recently it's getting more and more frequent... management are starting to get pissed :/ No more ideas? I've tried everything else people have mentioned. Cheers, Steve On Mon, Feb 02, 2004 at 01:03:01PM -0500, Bill Hamel wrote: Hi, Have you checked for IRQ conflicts ? -b Quoting Steve Foy [EMAIL PROTECTED]: Hi, On Mon, Feb 02, 2004 at 06:04:40PM +0100, Philipp von Klitzing wrote: Steve, this really is a FAQ. You need add to EACH (!) sip user something like disallow=all allow=ulaw allow=alaw allow=gsm I do have that in my sip.conf. I am using ulaw. Calls from the SIP phones through Asterisk and out one of my X100P cards are working 95% of the time and also, incoming calls through the X100P cards to the SIP phones are the same. The only problem is that every once in a while, without any odd circustances that I can see, the call just drops and the remote user is gone. The box running asterisk isn't under heavy load, so I can't see why this is happening. I am not using g.729 or 723, just plain old ulaw, which I have got enabled in sip.conf Cheers, Steve -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by The CCIS.net MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options
RE: [Asterisk-Users] Calls dropping off
Steve Foy wrote: Right... It just happened there now, this came up: Feb 5 14:34:18 WARNING[1133742896]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 3 (Response) I'm not sure if that's related to it, but it's the only thing that came up when the call got cut off. Here's the generic sip.conf stuff [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) allow=all allow=GSM allow=G729 allow=iLBC allow=SpeeX; Allow all codecs allow=ulaw Here's a sip.conf declaration: ; Andy [108] type=friend username= secret= host=dynamic dtmfmode=rfc2833 callerid=Andy McAlister 108 context=internal [EMAIL PROTECTED] qualify=yes canreinvite=no And the relevant extension.conf bit: ;Andy exten = 108,1,Dial(SIP/108,15) exten = 108,2,Playback(int-voicemail/108) exten = 108,3,Voicemail(s108) exten = 108,102,Playback(int-voicemail/108) exten = 108,103,Voicemail(s108) Any insight vastly appreciated! Cheers, Steve Hmm.. From memory while back I think I had a similar problem. Try to: bind= YOUR IP ADDRESS. Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calls dropping off
Still no luck, calls are still dropping off about the same amount as before. Any more ideas!? On Fri, Jan 30, 2004 at 05:14:27PM +, Steve Foy wrote: Thanks, I'll try that and see how it goes. Cheers, Steve On Fri, Jan 30, 2004 at 11:46:05AM -0500, Bill Hamel wrote: Try adding it to the phones involved so it looks like this: ; Shirley [100] type=friend username=xxx secret=xxx host=dynamic dtmfmode=rfc2833 callerid=Shirley O'Neill 100 context=internal [EMAIL PROTECTED] qualify=yes canreinvite=no -b Quoting Steve Foy [EMAIL PROTECTED]: Bill, On Fri, Jan 30, 2004 at 08:19:51AM -0500, Bill Hamel wrote: Shot in the dark here ... Do you have: canreinvite=no Set in sip.conf for the SIP phones in question ? No, I don't. All I have in sip.conf is the general stuff like: [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) allow=all allow=GSM allow=G729 allow=iLBC allow=SpeeX; Allow all codecs allow=ulaw and then about 10 friends like this: ; Shirley [100] type=friend username=xxx secret=xxx host=dynamic dtmfmode=rfc2833 callerid=Shirley O'Neill 100 context=internal [EMAIL PROTECTED] qualify=yes -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by The CCIS.net MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Calls dropping off
Steve Foy wrote: Still no luck, calls are still dropping off about the same amount as before. Any more ideas!? On Fri, Jan 30, 2004 at 05:14:27PM +, Steve Foy wrote: Thanks, I'll try that and see how it goes. Cheers, Steve On Fri, Jan 30, 2004 at 11:46:05AM -0500, Bill Hamel wrote: Try adding it to the phones involved so it looks like this: ; Shirley [100] type=friend username=xxx secret=xxx host=dynamic dtmfmode=rfc2833 callerid=Shirley O'Neill 100 context=internal [EMAIL PROTECTED] qualify=yes canreinvite=no -b Quoting Steve Foy [EMAIL PROTECTED]: Bill, On Fri, Jan 30, 2004 at 08:19:51AM -0500, Bill Hamel wrote: Shot in the dark here ... Do you have: canreinvite=no Set in sip.conf for the SIP phones in question ? No, I don't. All I have in sip.conf is the general stuff like: [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) allow=all allow=GSM allow=G729 allow=iLBC allow=SpeeX; Allow all codecs allow=ulaw and then about 10 friends like this: ; Shirley [100] type=friend username=xxx secret=xxx host=dynamic dtmfmode=rfc2833 callerid=Shirley O'Neill 100 context=internal [EMAIL PROTECTED] qualify=yes -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by The CCIS.net MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I would add: reinvite=no in addition to canreinvite=no. It may do the trick. Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Calls dropping off
Hi! I would add: reinvite=no in addition to canreinvite=no. It may do the trick. There is no such parameter as reinvite=. Use canreinvite= only. Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calls dropping off
On Mon, Feb 02, 2004 at 05:28:38PM +0100, Philipp von Klitzing wrote: I would add: reinvite=no in addition to canreinvite=no. It may do the trick. There is no such parameter as reinvite=. Use canreinvite= only. Didn't think so. I don't understand why this is happening, the server isn't heavily loaded or anything like that... Grr.. -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calls dropping off
Hi, On Mon, Feb 02, 2004 at 06:04:40PM +0100, Philipp von Klitzing wrote: Steve, this really is a FAQ. You need add to EACH (!) sip user something like disallow=all allow=ulaw allow=alaw allow=gsm I do have that in my sip.conf. I am using ulaw. Calls from the SIP phones through Asterisk and out one of my X100P cards are working 95% of the time and also, incoming calls through the X100P cards to the SIP phones are the same. The only problem is that every once in a while, without any odd circustances that I can see, the call just drops and the remote user is gone. The box running asterisk isn't under heavy load, so I can't see why this is happening. I am not using g.729 or 723, just plain old ulaw, which I have got enabled in sip.conf Cheers, Steve -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calls dropping off
Do you have busydetect=yes and/or callprogress= in zapata.conf? If so set them to no. On Mon, 2004-02-02 at 11:10, Steve Foy wrote: Hi, On Mon, Feb 02, 2004 at 06:04:40PM +0100, Philipp von Klitzing wrote: Steve, this really is a FAQ. You need add to EACH (!) sip user something like disallow=all allow=ulaw allow=alaw allow=gsm I do have that in my sip.conf. I am using ulaw. Calls from the SIP phones through Asterisk and out one of my X100P cards are working 95% of the time and also, incoming calls through the X100P cards to the SIP phones are the same. The only problem is that every once in a while, without any odd circustances that I can see, the call just drops and the remote user is gone. The box running asterisk isn't under heavy load, so I can't see why this is happening. I am not using g.729 or 723, just plain old ulaw, which I have got enabled in sip.conf Cheers, Steve -- Go to http://www.digium.com/index.php?menu=documentation and look at the Unofficial Links section. This section has links to a wide variety of 3rd party Asterisk related pages. My page is the Asterisk Resource Pages. BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Calls dropping off
Philipp von Klitzing wrote: Hi! I would add: reinvite=no in addition to canreinvite=no. It may do the trick. There is no such parameter as reinvite=. Use canreinvite= only. Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Well... Googling... Few months ago produced that option. When was that option dropped? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calls dropping off
On Mon, Feb 02, 2004 at 11:16:16AM -0600, Eric Wieling wrote: Do you have busydetect=yes and/or callprogress= in zapata.conf? If so set them to no. I did have busydetect=yes, I've just commented it out. I don't see that busydetect would cause problems, as the call does get answered and could last several minutes before it gets dropped :( All I have in zapata.conf now is: [channels] signalling=fxs_ks context=incoming echocancel=yes echocancelwhenbridged=yes echotraining=yes threewaycalling=yes transfer=yes musiconhold=default usecallerid=yes callerid=Outside Line 1 channel=1 callerid=Outside Line 2 channel=2 callerid=Outside Line 3 channel=3 -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calls dropping off
Hi, Have you checked for IRQ conflicts ? -b Quoting Steve Foy [EMAIL PROTECTED]: Hi, On Mon, Feb 02, 2004 at 06:04:40PM +0100, Philipp von Klitzing wrote: Steve, this really is a FAQ. You need add to EACH (!) sip user something like disallow=all allow=ulaw allow=alaw allow=gsm I do have that in my sip.conf. I am using ulaw. Calls from the SIP phones through Asterisk and out one of my X100P cards are working 95% of the time and also, incoming calls through the X100P cards to the SIP phones are the same. The only problem is that every once in a while, without any odd circustances that I can see, the call just drops and the remote user is gone. The box running asterisk isn't under heavy load, so I can't see why this is happening. I am not using g.729 or 723, just plain old ulaw, which I have got enabled in sip.conf Cheers, Steve -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by The CCIS.net MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Calls dropping off
On Mon, 2004-02-02 at 11:21, Senad Jordanovic wrote: Philipp von Klitzing wrote: Hi! I would add: reinvite=no in addition to canreinvite=no. It may do the trick. There is no such parameter as reinvite=. Use canreinvite= only. Well... Googling... Few months ago produced that option. When was that option dropped? It wasn't dropped. People are just getting a bit more strict on making sure the advice given is correct. It was part of a rant recently that it has never been part of the config but referenced enough that it has caused problems. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Calls dropping off
On Mon, 2004-02-02 at 11:21, Senad Jordanovic wrote: Philipp von Klitzing wrote: Hi! I would add: reinvite=no in addition to canreinvite=no. It may do the trick. There is no such parameter as reinvite=. Use canreinvite= only. Well... Googling... Few months ago produced that option. When was that option dropped? It wasn't dropped. People are just getting a bit more strict on making sure the advice given is correct. It was part of a rant recently that it has never been part of the config but referenced enough that it has caused problems. -- Steven Critchfield [EMAIL PROTECTED] It's also showing up on the wiki: http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD Should there be some sort of check on config files when they are parsed so * barks when it sees either an invalid option or syntax? - Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Calls dropping off
Hi! It's also showing up on the wiki: http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD Where? ;- Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Calls dropping off
Philipp von Klitzing wrote: Hi! It's also showing up on the wiki: http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD Where? ;- Philipp Interesting...! Mysteriously... reinvite has EDITED it self in above URL to canreinvite in space in few hours... :) Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calls dropping off
Steve Foy wrote: Hi, I've got a fairly working Asterisk setup, with a few minor glitches, one of which is very very irritating. Sometimes, during a call, the remote end just drops off. We're using software SIP phones (SJPhone) connecting to * then out through analogue lines with X100P cards. There is nothing in the logs and nothing on the console, the call just seems to 'go away'! Enable 'sip debug' at the CLI and send some detailed log file. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calls dropping off
Shot in the dark here ... Do you have: canreinvite=no Set in sip.conf for the SIP phones in question ? Ciao, -b Quoting Steve Foy [EMAIL PROTECTED]: Hi, I've got a fairly working Asterisk setup, with a few minor glitches, one of which is very very irritating. Sometimes, during a call, the remote end just drops off. We're using software SIP phones (SJPhone) connecting to * then out through analogue lines with X100P cards. There is nothing in the logs and nothing on the console, the call just seems to 'go away'! Can anyone shed any light on this? This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calls dropping off
On Fri, Jan 30, 2004 at 01:18:29PM +0100, Olle E. Johansson wrote: Enable 'sip debug' at the CLI and send some detailed log file. It's very difficult to catch the logs when this happens, it doesn't happen all the time, and I'm hardly ever on the phone so, it would be even less likely to happen to me. Is there a way I can get the sip debug lines to get piped out to a file with timestamps? -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calls dropping off
Bill, On Fri, Jan 30, 2004 at 08:19:51AM -0500, Bill Hamel wrote: Shot in the dark here ... Do you have: canreinvite=no Set in sip.conf for the SIP phones in question ? No, I don't. All I have in sip.conf is the general stuff like: [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) allow=all allow=GSM allow=G729 allow=iLBC allow=SpeeX; Allow all codecs allow=ulaw and then about 10 friends like this: ; Shirley [100] type=friend username=xxx secret=xxx host=dynamic dtmfmode=rfc2833 callerid=Shirley O'Neill 100 context=internal [EMAIL PROTECTED] qualify=yes -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calls dropping off
Try adding it to the phones involved so it looks like this: ; Shirley [100] type=friend username=xxx secret=xxx host=dynamic dtmfmode=rfc2833 callerid=Shirley O'Neill 100 context=internal [EMAIL PROTECTED] qualify=yes canreinvite=no -b Quoting Steve Foy [EMAIL PROTECTED]: Bill, On Fri, Jan 30, 2004 at 08:19:51AM -0500, Bill Hamel wrote: Shot in the dark here ... Do you have: canreinvite=no Set in sip.conf for the SIP phones in question ? No, I don't. All I have in sip.conf is the general stuff like: [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) allow=all allow=GSM allow=G729 allow=iLBC allow=SpeeX; Allow all codecs allow=ulaw and then about 10 friends like this: ; Shirley [100] type=friend username=xxx secret=xxx host=dynamic dtmfmode=rfc2833 callerid=Shirley O'Neill 100 context=internal [EMAIL PROTECTED] qualify=yes -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by The CCIS.net MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calls dropping off
Thanks, I'll try that and see how it goes. Cheers, Steve On Fri, Jan 30, 2004 at 11:46:05AM -0500, Bill Hamel wrote: Try adding it to the phones involved so it looks like this: ; Shirley [100] type=friend username=xxx secret=xxx host=dynamic dtmfmode=rfc2833 callerid=Shirley O'Neill 100 context=internal [EMAIL PROTECTED] qualify=yes canreinvite=no -b Quoting Steve Foy [EMAIL PROTECTED]: Bill, On Fri, Jan 30, 2004 at 08:19:51AM -0500, Bill Hamel wrote: Shot in the dark here ... Do you have: canreinvite=no Set in sip.conf for the SIP phones in question ? No, I don't. All I have in sip.conf is the general stuff like: [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) allow=all allow=GSM allow=G729 allow=iLBC allow=SpeeX; Allow all codecs allow=ulaw and then about 10 friends like this: ; Shirley [100] type=friend username=xxx secret=xxx host=dynamic dtmfmode=rfc2833 callerid=Shirley O'Neill 100 context=internal [EMAIL PROTECTED] qualify=yes -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by The CCIS.net MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
