Re: [asterisk-users] System() command refuses to execute bash script
On Mon, 2 Mar 2015, Stefan Viljoen wrote:
So the problem was not Asterisk or BASH or permissions, but rather that
it appears that all paths in any System() script must be absolutely, not
relatively, specified.
Not quite.
The 'base' for relative paths would be the 'cwd' (current working
directory) of the Asterisk process.
You can show the cwd for your running Asterisk by:
sudo ls -l /proc/$(pidof asterisk)/cwd
which is a link to the process's cwd.
I suspect if you search your file system ('sudo find / -name
wireless-executed'), you will find 'wireless-executed' -- probably in the
directory shown by the above command.
You can set this in the script that starts Asterisk. I set mine to /tmp/
('cd /tmp/') so I know where any random file access will occur, relatively
speaking.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] [OT] switches
On Wed, 25 Feb 2015, A J Stiles wrote: The limiting factor with a switch carrying IP telephony traffic is not bandwidth, but routing table entries; and even cheap switches nowadays will usually take 1024 entries, if not 4096. Are you referring to the MAC CAM table? Saying 'routing table' and 'switch' in the same sentence seems confusing. Do VOIP devices take more table entries than other Ethernet devices? I.e. more than 1? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] situation with ivr and four-channel gateway
On Wed, 25 Feb 2015, John Kiniston wrote:
I'd recommend using DEVICE_STATE
On your extension 101, Check the DEVICE_STATE of peer SIP/101, If it's
not 'NOT_INUSE' then dial it, Otherwise dial SIP/102
exten =
101,1,ExecIf($[${DEVICE_STATE(SIP/101)}=NOT_INUSE]?Dial(SIP/101,40))
same = n,Dial(SIP/102,40,t)
same = n,Hangup()
Remember to set 'callcounter = yes' in sip.conf.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000--
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Re: [asterisk-users] forcing GSM on certain extensions
On Tue, 30 Dec 2014, Joseph wrote: I'm trying to force GSM when I call on certain extension but I'm getting connected with ulaw Which is not suitable when bandwidth is low and slow. my phone is iax-322 in iax.conf [iaxy-322] According to: http://www.voip-info.org/wiki/view/IAXy http://voxilla.com/2004/03/08/first-looks-digiums-sexy-little-iaxy/ The IAXy was initially released as a ULAW only device with ADPCM 'promised' in a future firmware release. No mention of GSM. Is your experience different? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Want web page to listen to meetme (WebRTC?)
I have a web page to do the usual meetme admin stuff -- mute, kick, etc. Now, the client is asking if they can listen to the meetme -- click and audio comes out the computer speakers. How can this be implemented? Is this a use case for WebRTC? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] About voip gateway
On Mon, 8 Dec 2014, Leonel Florin wrote: Hay friends, I want to know how many simultaneous call can i do throughout a voip gateway from the internet call to the normal telephony network, because i want to see what implementation do i have to do multiple call from internet to differents telephones. Please reply with a few more details of what you are planning on doing. For example: I want my computer to originate 100 simultaneous calls to PSTN subscribers who have 'opted-in' to receive a 60 second political announcement.' If all you want to do is route calls, OpenSIPS may be a better tool. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] day night service toggle
On Fri, 28 Nov 2014, A J Stiles wrote: And note that this really should be done by dialling separate numbers for in and out, because toggle actions are annoying as hell in practice -- it's easier to remember two different numbers, than to remember what state you are currently in. According to NANPA's 'Vertical Service Codes, Code Definitions' (http://www.nanpa.com/number_resource_info/vsc_definitions.html), *72 and *73 are for call forwarding activation and deactivation (respectively). Of course, the OP could use execiftime() instead and just make sure to go to lunch and return at the exact same time every day. The dialplan coding is easier and no pesky codes to remember. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Upgraded to 13 and now Mailbox is empty in sip show peers
Please don't top-post.
On Wed, 19 Nov 2014, Jayson Baker wrote:
This same issue has happened on 1.8 as well. And so far on all 6 of our
systems we upgraded to 13. It must be something simple? How can we
diagnose it?
Coming late to the party, but...
I'd run tcpdump ('sudo tcpdump -A -s 0 port 3306') and see:
1) Are packets flowing back and forth like you'd expect.
2) Can you capture an insert statement so you can apply it in the MySQL
command line client? You may get a meaningful error message or observe
something funky in one of the columns.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000--
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[asterisk-users] AGI and AMI in PHP -- What's current?
I'm writing some code that needs to access AMI in PHP. (I'll probably be doing AGI later as well.) I'm looking at phpagi (2.20), but it hasn't been updated since 2010 and appears to be a bit behind current Asterisk -- No event handler for event 'fullybooted'. What PHP framework/library are you using -- and why? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] AGI and AMI in PHP -- What's current?
On Tue, 18 Nov 2014, Eric Wieling wrote: Other than a few minor patches, we use stock phpagi. Can you spare me a flat spot on my forehead and share the wealth? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] AGI and AMI in PHP -- What's current?
On Tue, 18 Nov 2014, Eric Wieling wrote: diff at http://pastebin.com/wfDR6u0a -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] OT: script to remove leading and trailing silence
Anybody care to share a script or snippet of what they use to remove leading and trailing silence from customer recorded files? I've fiddled with sox to remove the leading; reverse the file; remove the now leading; and reverse the file again, but I'm not really happy with my results. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Debugging issues with setup
On Fri, 24 Oct 2014, Marco Carvalho wrote: I set up a new server for Asterisk with 11 cert 6 on it. I am migrating from a previous server. I have replicated all the configurations, modules and setup that I know of. However, when I tested an outbound call, it didn’t work. Checking the asterisk message log yielded nothing. Any ideas on how I can isolate and trace the issue? I'd fire up the CLI, bump up debug and verbose, enable SIP debugging (if your outbound provider is SIP), and observer the output when making a call. There should be some clues there. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Phone ( Telecom feature )
Please don't top-post. On 08-Oct-2014 10:48 PM, Dania Asi da...@futuretrendsest.com wrote: I am a system integrator and I have a whale clients in UAE , I will not proceed further in dealing with Asterisk because of the lack of support and because of the rude emails. On Thu, 9 Oct 2014, Mitul Limbani wrote: Hey Danni, Having whale client means you ought to ask for paid consulting in first place. After a couple of off-list exchanges, I suspect (no facts) the situation is this: She was sold an Asterisk system for one of her clients from an unscrupulous vendor. When she asked the vendor for support, she was told 'we' (the mailing list) ARE her support. Thus, it was not entirely unreasonable for her to assume that the support she had paid for would be familiar with her configuration and would be eager to provide service since As vendors you are supposed to be interested to expand your business area and as I told you we have a huge database of clients that need such products... (Actual quote.) I chalk up the 'rude and non-professional' accusations as 'unmet expectations' and 'cultural differences.' Again, no facts in evidence, just my interpretation from the off-list exchanges. My apologies in advance if I've misinterpreted the situation. I've copied the parties to the off-list exchanges in case they would care to clarify my assumptions. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Phone ( Telecom feature )
On Wed, 8 Oct 2014, Dania Asi wrote: Dear Mr. Adam, Thank you for you kind words and for judging me. I am a system integrator and I have a whale clients in UAE , I will not proceed further in dealing with Asterisk because of the lack of support and because of the rude emails. I have no idea what is wrong with you people. And I hope you get well soon from whatever is happening to you. You are misjudging this community. It is staffed by volunteers. Their range of experience and knowledge range from rank novices to experts that you couldn't afford to hire. They do it out of the kindness of their hearts and because they have been helped in the past. I've been on this list for about 10 years and continue to be amazed at the knowledge and ingenuity exchanged for 'free.' People are more likely to help if you include sufficient information and demonstrate that you have at least tried to resolve your issue. Nobody is interested in doing your work (be it homework or work-work) for you. Be thankful there are no 'Carl J. Lydick's on this list. He was a brilliant VMS programmer back in the day. If you asked an intelligent question, you would get an amazing answer. If you asked a stupid question or indicated you had not invested any effort, you might receive an obfuscated command that would erase your disks if you were to ignorantly 'cut-n-paste' the command. Asterisk is a fantastic product. It is like a Lego or Erector set. You have all the bits. It is up to you to fulfill the vision in your head. If you are looking for paid support or a canned solution, it is available. You just looked in the wrong place. Maybe it's just a 'cultural misunderstanding.' -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Lost audio on forwarded calls
Asterisk does not need to care. Is it SIP all the way through? Thanks, Steve T On Fri, Oct 3, 2014 at 3:12 PM, Todd R. tjrl...@live.com wrote: OK, been messing with Asterisk for a long time and I have my opinion on where the issues lies but sometimes it's just nice to see what others think that can relate :-) Here goes.. Inbound calls flow like this: Tier 1 Provider (SIP) Asterisk 1.8 Name Brand PBX - Calls work fine Outbound calls flow like this: Name Brand PBX Asterisk 1.8 Tier 1 provider (SIP) - Calls work fine Problem is being reported on that many (not all) calls have no audio when they are forwarded. Example of forwarded call: Inbound call comes in from Tier 1 Provider Asterisk 1.8 Name Brand PBX Name Brand PBX then forwards the call back out to users cell phone: Name Brand PBX Asterisk 1.8 Tier 1 provider No audio a large percentage of the time. It's my opinion that the Asterisk box only sees the forwarded call as a regular outbound call and forwards it on to the Tier 1 provider then to the users cell phone. I don't see how Asterisk even knows or cares if it was forwarded within the Name Brand PBX. The Name Brand PBX is the one making the connection of the inbound and outbound call. All other inbound and outbound calls are fine, audio is only lost when the Name Brand PBX connects the two calls and creates the forward. Thoughts? -- _ -- 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] how to strip +1 out of incoming number
On Thu, Oct 2, 2014 at 10:52 AM, motty cruz motty.c...@gmail.com
wrote:
Hello, our VoIP send us caller ID +1(area)(number) for instance
+16024224334 is there a way to strip +1 out of caller ID?
I've always done it as 2 steps to handle carrier weirdness:
; trim leading +1 from DNIS
same = n, execif($[${DNIS:0:1} =
+]?set(DNIS=${DNIS:1}))
same = n, execif($[${DNIS:0:1} =
1]?set(DNIS=${DNIS:1}))
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
--
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Re: [asterisk-users] How to append the recording file.
On Sun, 28 Sep 2014, Anurag Rana wrote: I am trying to record the call using MixMonitor. ... Now I know that 'a' option is used to append the recording to a file but I couldn't find any example on how to use it? Also if I use 'a' option and file doesn't exist then is it created or it is error? Any suggestions please? Sure. Try it -- faster than waiting for a response. If it depends on the file already existing, add it to 'core show application mixmonitor.' If it creates the file if it doesn't exist, add it to 'core show application mixmonitor.' -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Playback/background audio from MySQL BLOB
For some applications, storing recorded audio (prompts and caller recordings) as a BLOB in MySQL has advantages. So, once I have the audio in the database, how can I play it? Creating temporary files seems so tacky. Is there another way to playback or background audio either by specifying a URL or from a memory buffer (either C or PHP)? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Playback/background audio from MySQL BLOB
On 09/23/2014 02:17 PM, Steve Edwards wrote: For some applications, storing recorded audio (prompts and caller recordings) as a BLOB in MySQL has advantages. On Tue, 23 Sep 2014, Don Kelly wrote: I'm curious about what the advantages are of storing audio in a blob. Wouldn't it be more efficient to store it in a file and just put the filename in the database? Multiple web servers, multiple Asterisk servers, multiple DB servers, synchronizing filesystems vs db, etc. It appears to eliminate some problems, but Asterisk limiting audio playback to files seems like a tough obstacle. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk failed to authenticate device - attack attempt.
On Mon, 8 Sep 2014, motty cruz wrote: I continue to see the following msg on my Asterisk log: [Sep 8 15:34:37] NOTICE[7375]: chan_sip.c:23277 handle_request_invite: Failed to authenticate device 9009sip:9...@196.107.xx.xx;tag=8dd48dd2 First step is to determine the source -- is it coming from your network or from the Internet. 'sip set debug on,' tcpdump, ngrep, wireshark can all be useful. If it is coming from your network, make note of the MAC address. The first 3 octets are the OUI. Google 'OUI Lookup.' This will tell you the manufacturer (or at least who made the board inside the device). This may give you a clue like 'Cisco Linksys LLC' and you may remember you have an old Sipura (which was bought by Linksys, which was bought by Cisco) laying around that somebody may have decided to 're-purpose' without telling you. If it is coming from the Internet, learn a bit about iptables. The best case scenario is that you know everybody that should be accessing your pbx so you can 'whitelist' the good guys and DROP everything else. Some people moan about how they have clients that travel. Unless they travel to China, Russia, North Korea, Crapistan, etc, just block entire regions of the world. That will knock off 90% of your 'attack surface.' Maybe you can limit traffic to just a couple of class C addresses. Finally, mop up the anklebitters with fail2ban. Oh, and nice long 'random' passwords on all of your SIP endpoints and if you can get away from 4 digit extensions, all the better. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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] Pattern Extension not working in Dialplan
Please don't top-post.
On Sun, Sep 7, 2014 at 1:41 PM, Anurag Rana anuragrana31...@gmail.com wrote:
I created a dummy dialplan where I ask the user to enter the age.
[macro-age]
exten = s,1,Background(my/age) ;;Play recorded message to enter age
exten = s,n,WaitExten(10)
exten = _XX,1,Set(AGE=${EXTEN}) ;; this line is not executing, instead
dialplan is terminating with error given below.
exten = s,n,NoOp(${AGE})
exten = s,n,GotoIf($[${LEN(${AGE})} 0]?notEmpty)
exten = s,n,Goto(s,1)
exten = s(notEmpty),n,Background(my/thank-you)
exten = s,n,Wait(1)
On Sun, 7 Sep 2014, John Kiniston wrote:
The first issue I see is you are attempting to insert your pattern match
in the middle of your 's' extension, That's going to break your 's'
extension. The second issue is that you are matching on XX which will
match two digits, You need to match on _X instead if you are attempting
to match on the number 8.
I recommend you look into 'read' instead of trying to do a pattern
match.
A pattern match is a reasonable method. I use pattern matching more often
that the read() application. Try both and see which meets your needs
better.
Are you really defining a 'macro' or is that just the (misleading) name
you chose for your context. Personally, I use gosub() more, but again,
try both :)
I suggest you try 'dialplan show macro-age' to see how Asterisk is
interpreting your dialplan. I suspect it is not what you expect.
In specific, your ordering of '_xx' in the middle of 's' is odd. This
would disrupt the value of the priority in older versions of Asterisk, but
it appears that it does work in modern (I'm using 11) versions.
Also, a label ('notEmpty') belongs to a priority, not an extension.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000--
_
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Re: [asterisk-users] Pattern Extension not working in Dialplan
On Sun, 7 Sep 2014, Steve Edwards wrote:
In specific, your ordering of '_xx' in the middle of 's' is odd. This would
disrupt the value of the priority in older versions of Asterisk, but it
appears that it does work in modern (I'm using 11) versions.
Disregard that. I can't even follow my own advice ('dialplan show
macro-age'). Don't 'intermingle' extensions.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
--
_
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Re: [asterisk-users] Asterisk secure fine tune - stop attack
Please don't top post. On Thu, 4 Sep 2014, motty cruz wrote: Hi A J, believe me, I wish i do as you suggested, however I have a few extensions outside the office with dynamic IPs, so that is not a possibility. Do your few extensions travel to China, Russia, Iran, Iraq, North Korea, etc? (Sorry if I stepped on anybody's toes.) If you configure iptables to drop all and then only allow the few IP address ranges you really need, 90% of the problem is solved. Then use fail2ban to manage the remaining anklebitters. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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] (no subject)
For future reference, a well chosen subject will yield more relevant replies. Better bait == better fish. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Understanding local channels
On Mon, 25 Aug 2014, Patrick Laimbock wrote: https://wiki.asterisk.org/wiki/displa/AST/Introduction+to+Local+Channels s/displa/display/ -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] FYI: Block Comments
On Mon, 25 Aug 2014, Joshua Colp wrote:
how many of you know about templates?
(You may get more replies with a more 'on-target' subject. I lost
interest in 'block comments' but was curious why the thread was still
getting replies.)
Love templates. Use them in extensions.conf, sip.conf, and iax.conf every
day.
Here's an example from extensions.conf:
[party-line](digit-timeout,h,i,max-timeout,pound-main,s)
same = n, agi(write-cdr)
same = n, background(${PROMPTS-PATH}/0116)
...
Where the templates look like:
[digit-timeout](!)
exten = t,1,goto(${CONTEXT},s,1)
[h](!)
exten = h,1,goto(finish-call,h,1)
[i](!)
exten = i,1,goto(${CONTEXT},s,1)
[max-timeout](!)
exten = T,1,goto(max-time,s,1)
[pound-main](!)
exten = #,1,goto(main-menu,s,1)
[s](!)
exten = s,1,verbose(1,[${EXTEN}@${CONTEXT}!${ANI}])
Note that the 's' template has to be the last template specified in the
template list. Also, that '${EXTEN}@${CONTEXT}' makes for a quick
cut-n-paste into the 'dialplan show' CLI command.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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[asterisk-users] Can't hangup channel from CLI
Asterisk 11.11.0 on CentOS 6.5, installed from RPMs. OpenSIPS fronting Asterisk from a Tekelec T9000. I'm accumulating stuck channels. I'm googling now and I recognize that Friday afternoons are the worst time to ask questions, but I'm getting desperate because this is keeping me from rolling a system out to production. (Yup, I know. Who rolls out a system on a Friday afternoon...) They all show (core show channels) that they are running one of my AGIs, so I suspect something changed in processing AGIs from 1.2(!). (I've recently changed from using 'signal()' to using 'sigaction()' if that rings any bells with anyone.) That's puzzling enough, but killing the AGI process from the shell doesn't help. Neither does 'channel request hangup SIP/a.b.c.d-0031' Neither does 'channel redirect SIP/a.b.c.d-0031 default,s,3' where 'default,s,3' is a hangup(). Neither of these commands log any errors. The only other clue may be this message: Autodestruct on dialog '1405009120-21730426@T9000' with owner SIP/a.d.b.c-0031 in place (Method: BYE). Rescheduling destruction for 1 ms for each of the hung channels every 30 seconds or so. I haven't identified what callers are doing to reproduce the error reliably yet. Any clues or suggestions? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] PRI timing settings
PRI intense debug should show all you need to fix this. On Wed, Aug 20, 2014 at 12:13 PM, Jeff LaCoursiere j...@jeff.net wrote: Sadly none of these changes have made any difference. I'll report the resolution for posterity once we find it. Thanks, j On 08/20/2014 10:13 AM, Don Kelly wrote: It’s possible that Sprint is burping on the name. Try first dropping the “1.” Then try dropping the name also, if necessary. --Don *From:* asterisk-users-boun...@lists.digium.com [ mailto:asterisk-users-boun...@lists.digium.com asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jeff LaCoursiere *Sent:* Wednesday, August 20, 2014 10:03 AM *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] PRI timing settings What about the text portion? Should that never be sent? I was indeed sending the '1', and I will remove that to see if it solves my problem, but I also have the company name in there. I feel like a newb asking such questions, but I've never had this issue before :) Company 1NXXNXX Cheers, j On 08/20/2014 09:46 AM, Eric Wieling wrote: NXXNXX is the correct format of CallerID numbers in NANPA. The leading 1 is not part of any NANPA phone number. Toll free “area codes” are also not valid for CallerID. *From:* asterisk-users-boun...@lists.digium.com [ mailto:asterisk-users-boun...@lists.digium.com asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jeff LaCoursiere *Sent:* Wednesday, August 20, 2014 2:41 PM *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] PRI timing settings On 08/20/2014 07:58 AM, Scott L. Lykens wrote: On Aug 19, 2014, at 5:56 PM, Jeff LaCoursiere j...@jeff.net wrote: I wrote earlier today about a new PRI installation in the Caribbean, where all outbound calls are functioning fine *except* calls to Sprint phone numbers, which get rejected immediately as busy. I don’t know what expectations for CLID your carrier might have, or for that matter the upstream carrier, however, we found through our CLEC here in the US that while the CLEC was happy to take e.164 formatted numbers from us as CLID, Global Crossing would reject them further upstream resulting in our calls to many toll frees being rejected. Switching to 10 digit CLID on all outbound calls through that PRI solved the problem. I don’t know if this is your problem but be sure your CLID is in the most simple format possible for your region to help rule it out. sl This makes me curious... what *is* the simplest format possible for NANPA numbers? I'm sure there must be a spec to conform to. Can anyone point me to it? Cheers, j -- _ -- 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] Dispatching calls question
On Wed, 20 Aug 2014, Jerry Geis wrote: I have a question about dispatching calls... If I try to dispatch a call on line 1 using the AMI and I check in my table to see if line 1 is available and it is So I have done my checking now I dispatch my call and at that same time a call comes in on line 1 and now its no longer available for me to make a call, I connect on AMI and my call fails How do I prevent this from happening? Sure I can start at 23 instead of 1 and work down instead of up but eventually the same thing may happen. If you're using something like MySQL, use 'get_lock/release_lock.' If you're using some other database, see what locking features you have available. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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] Error opening file for reading: Permission denied
On Tue, Aug 19, 2014 at 11:36 AM, Mitch Claborn mitch...@claborn.net No, that's not it. The wording is different. Can you run Asterisk via strace? Something like: sudo -u asterisk strace /usr/sbin/asterisk -c -p -U asterisk -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk peer definition registration
Hi, I am using Asterisk currently as a SIP proxy to a SIP provider. Inside my real-time, I would set the SIP credential based on what the user has provided. For example [name] type=peer defaultuser=USER_PROVIDED secret=USER_PROVIDED host=USER_PROVIDED When I reset Asterisk, Asterisk will attempt to register with the sip provider. And if there are sufficiently amount of records with invalid credentials, I'll get blocked by the SIP provider as they might think that I'm brute forcing. Just a question to check if there's any chance I could ask Asterisk not to register when I reset. Or is there any other possible solution for this? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk peer definition registration
Is there a way that I could set the configuration for reloading after ITSP brute force timer expiration? On Sun, Aug 17, 2014 at 3:51 AM, Paul Belanger paul.belan...@polybeacon.com wrote: On Sat, Aug 16, 2014 at 11:21 AM, Steve Ng steveng.1...@gmail.com wrote: Hi, I am using Asterisk currently as a SIP proxy to a SIP provider. Inside my real-time, I would set the SIP credential based on what the user has provided. For example [name] type=peer defaultuser=USER_PROVIDED secret=USER_PROVIDED host=USER_PROVIDED When I reset Asterisk, Asterisk will attempt to register with the sip provider. And if there are sufficiently amount of records with invalid credentials, I'll get blocked by the SIP provider as they might think that I'm brute forcing. Just a question to check if there's any chance I could ask Asterisk not to register when I reset. Or is there any other possible solution for this? No, only reload after your ITSP brute force timer has expired. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] *SOLVED* Re: Anyone have any experience with inbound SIP trunks from Simwood?
On Thu, 7 Aug 2014, A J Stiles wrote: . And my mistake was in sip.conf. The configuration stanza I had named simwood_in_slough should, of course, have been named after the number I had programmed in at the other end of the trunk . *hangs head in shame* It's OK. We're all a little 'slow' from time to time. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk websocket with Nginx 502 Gateway error
Hi all, I am using the following set up: SIPML5 -- Nginx -- Asterisk, where NGINX as a reverse proxy, main purpose is to take in wss and route to Asterisk's ws. However, I am facing this issue recently where Nginx will return 502 gateway error of 8018#0: *24183 upstream prematurely closed connection while reading response header from upstream, client: 116.15.31.xxx, server: asteriskstage.xxx.yy, request: GET / HTTP/1.1, upstream: http://127.0.0.1:8088/ws;, host: asteriskstage.xxx.yy Anyone has any idea why? Here's my nginx config: http://pastebin.com/UU0G3YLh Regards, Steve -- _ -- 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] Internal timing under load is critical ?
On Wed, 30 Jul 2014, babak wrote: According to some recommendations like http://osdial.org/howto/ Internal timing is very critical with Asterisk when it is under load and we must use DAHDI hardware or USB Voice Synch Tool http://www.sangoma.com/accessories/specialty-tools/ But according to my understanding of wiki https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces It seems it is not necessary now. Please tell me your opinions. I'm running Asterisk 11.11.0 on an HP ProLiant DL320e Gen8 E3-1240v2. 1,300 calls with no audio issues. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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] Internal timing under load is critical ?
Please don't top post. Please keep the thread only on the list. On Thursday, July 31, 2014 12:16 AM, Steve Edwards asterisk@sedwards.com wrote: I'm running Asterisk 11.11.0 on an HP ProLiant DL320e Gen8 E3-1240v2. 1,300 calls with no audio issues. On Wed, 30 Jul 2014, babak wrote: 1300 calls include playback voices ? The test scenario was for the first server to originate calls (via call files) to the second server and then 'playback()' a long file. The second server would answer the call and then 'playback()' a long file. Audio was flowing in each direction. Bandwidth was observed using 'iftop' as being in the 70mb to 80mb range in each direction (if I remember correctly). I placed calls from a handset to confirm audio quality. which timing module you are using: res_timing_timerfd.so or res_timing_kqueue.so or res_timing_dahdi.sores_timing_pthread.so I used res_timing_timerfd.so. I'm finally making the leap from 1.2 to the current decade :) I read somewhere that this was the timer to use and it seems to be working fine for me. I don't think the cores got much over 20% to 30% busy. Various failures were observed on the console from running out of file descriptors. This was on a stock CentOS 6.5 install with no tweaks to bump up the max file descriptors. The client only asked for 500 simultaneous calls so no further testing was done. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Limit Asterisk
Please don't top-post. Please trim irrelevant posts. On Thu, 24 Jul 2014, Eduardo Leones wrote: Another question, what audio format I use in MixMonitor to maintain a connection with reasonable quality and reduce the use of I / O disk? I think the question is premature. You have a resource limitation. Until you know what that limitation is, you can't really make intelligent changes. Is it I/O activity or I/O bandwidth? Are you swapping? (Swapping is 'death' to performance.) Are you running out of CPU? If you're planning on transcoding to something as computationally intensive as 729, do you have gobs of excess CPU capacity? If not, you'll just be trading 1 resource limitation for another. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Limit Asterisk
Please don't top-post. On Wed, 23 Jul 2014, Eduardo Leones wrote: In this case SSD disks you think it solves? Don't buy hardware until you've identified (either empirical or calculated) the bottleneck. But... SSDs do rock. I recently observed (via vmstat 5) a Samsung 840 topping out at 460,000 blocks per second. I can remember when 10,000 was big :) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Limit Asterisk
On Wed, 23 Jul 2014, Chris Bagnall wrote: The 840 is a great bit of kit... The 850 is supposed to be shipping next week. It's got 3d VNAND so the chip geometry can be bigger -- higher speeds and greater reliability. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Call Identifier Logging
On Tue, 22 Jul 2014, Steven Wheeler wrote:
Try this:
CDR(userfield) = ${SHELL(asterisk -rx core show channel ${CHANNEL} | grep Call
Identifer | cut -d: -f2 | cut -d[ -f2 | cut -d] -f1 | cut -d\n -f1):0:-1};
Not really interested in this topic, but invoking 6 processes seems a bit
excessive :)
How about something like:
asterisk -rx core show channel SIP/spa841-0003\
| awk '/Call Identifer/ {gsub(/[][]/,); print $3}'
Of course, a dialplan function would be best.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Call Identifier Logging
On Tue, 22 Jul 2014, Steve Edwards wrote:
How about something like:
asterisk -rx core show channel SIP/spa841-0003\
| awk '/Call Identifer/ {gsub(/[][]/,); print $3}'
Or:
asterisk -rx core show channel SIP/spa841-0003\
| awk -F'[][]' '/Call Identifer/ {print $2}'
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Strange Error
On Thu, 3 Jul 2014, Andrew Colin wrote: Does anyone know what this error means and how to fix it? [Jul 3 11:57:27] WARNING[17040] pbx.c: Don't know what to do with 'SIP/ 1) Please choose a more meaningful subject. Lots of errors can be considered strange. (Note that actually, this is a warning, not an error.) 2) Please show a few more lines of console output (with verbose and debug set high) to give us some context. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] recording in mp3
Please don't top-post. Please trim irrelevant posts. From: Tiago Geada mixmonitor has a argument that is a script ran just as the recording is finished. we use this to move the file from ramfs to final destination. you can use it to use sox and convert it... On Thu, 3 Jul 2014, andrew Colin wrote: Can you explain? I'm guessing 'core show application MixMonitor' should give you a good start. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gotoif($[${LEN(${CALLERID(number)})} != 4]?true) doesn't work...
Please don't top post.
On Wed, Jul 2, 2014 at 11:07 AM, Doug Lytle supp...@drdos.info wrote:
Differences between yours and mine:
Yours:
Gotoif($[${LEN(${CALLERID(number)})} != 4]?true)
Mine:
Gotoif($[${LEN(${get-admin-password})} 1]?2:4)
On Wed, 2 Jul 2014, Positively Optimistic wrote:
Thanks Doug!! I appreciate your response.. it sent me looking in the
right direction. The following yielded the results I was looking for.
Gotoif($[${LEN(${CALLERID(number)})} != 4]?true)
Here's a snippet from one of my (1.2) dialplans:
gotoif($[${LEN(${EXTEN})} 10] ?add-npa,${EXTEN},1)
So are the quotes now a requirement? (Quotes makes it look like you are
comparing strings instead of numbers to my old eyes.)
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000--
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Re: [asterisk-users] Attack on Sip server.
Please don't top-post. Please trim posts to the specific post you are replying to. On Fri, 27 Jun 2014, Anurag Rana wrote: Can't use anything which block IP addresses because my system is behind a gateway and attacker gets the address of that gateway. In this way I will end up blocking myself. Please suggest something else. The most effective approach would be to configure your gateway to block all IP addresses and white-list the ones you really need. If you are in control of the endpoints, moving to a non-standard SIP port as previously suggested should be pretty effective. Most script-kiddies won't bother to 'port-scan' to identify the new port number. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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] T1 Card RED ALARM
Remember to always check your cables first. Thanks, Steve T On Tue, Jun 24, 2014 at 1:47 PM, arun kumar arunvsadni...@gmail.com wrote: Thank you Josh for your valuable reply. I will do try changing the server and let you know what happening. ~Arun On Tue, Jun 24, 2014 at 8:39 PM, Josh Metzger joshdmetz...@gmail.com wrote: On Tue, Jun 24, 2014 at 5:25 AM, arun kumar arunvsadni...@gmail.com wrote: Hello All, I have a Digium Wildcard TE410P Quad-Span T1 Card, when I do connect T1 lines it goes in RED. When I do connect the same line on a different Server (Same Model T1 Card) it works fine. How do I examine/diagnose my T1 Card for any hardware failures. I heard about loopback test , how helpful it is? Here are my configuration /etc/zaptel.conf span=1,1,0,esf,b8zs bchan=1-23 dchan=24 Zaptel Configuration == SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) 24 channels configured Thanks ~Arun It could still be some sort of system config issue, even if you think everything is configured the same. Have you tried moving the T1 card from the Bad system to the good system? That will at least help narrow down if it's a bad card / port, or a config issue. -Josh -- _ -- 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
[asterisk-users] WSS over Asterisk
Hi, Have anyone tried using SIPML5 to connect to Asterisk over wss? I'm having the error as shown below Connecting to 'wss://54.xxx.xxx.xxx:8080/ws wss://54.254.228.251:8080/ws' SIPml-api.js?svn=224:1 ==stack event = starting SIPml-api.js?svn=224:1 __tsip_transport_ws_onerror SIPml-api.js?svn=224:1 __tsip_transport_ws_onclose SIPml-api.js?svn=224:1 ==stack event = failed_to_start Where if I'm connecting through ws://54.xxx.xxx.:8080/ws, it works fine. Any idea why? -- _ -- 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] WSS over Asterisk
I am using Asterisk v12.3. As far as DTLS, I understand that applying the following Javascript will temporarily fix for SIPML5 to Asterisk: https://gist.github.com/steve-ng/14b9b88af43f92db1e46 WS works for me, its just wss which I'm stuck currently. On Thu, Jun 12, 2014 at 4:37 AM, Miguel Molina mfmolina-lis...@millenium.com.co wrote: El 11/06/2014 1:52 p. m., Matthew Jordan escribió: On Wed, Jun 11, 2014 at 1:32 PM, William Hetherington w...@willwh.com wrote: Chrome 35 broke all of this you need to be using DTLS now I believe. I had working secure web sockets with asterisk 12.2.x and chrome 34 and then google broke eveything :) I have not yet got around to test out DTLS etc. with chrome 35 Just so I don't waste too much time when I go to test, does anyone know if all that's required for DTLS on the asterisk side is the following in sip.conf? dtlsenable=yes dtlsverify=yes dtlsrekey=60 dtlscafile=/usr/local/share/ca-certificates/myCA.crt dtlscertfile=/etc/ssl/mycert.com.pem dtlssetup=actpass I assume I also need TLS configs in http.conf Signalling is independent of the media; DTLS only affects the media. However, there are known issues with Chrome's negotiation of DTLS and Asterisk - see https://issues.asterisk.org/jira/browse/ASTERISK-22961 -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org It is broken in Chrome (firefox never had SDES) because the WebRTC standard favoured the DTLS SRTP implementation instead of the SDES one. The thing is that although Asterisk supports DTLS implementation, it only supports SHA-1 hashing but both Firefox and Chrome work with SHA-256. The patch proposed in ASTERISK-22961 is an effort to solve this issue. Best 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 -- _ -- 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] Interesting new hack attack
In the past little while, we've seen a wave of attacks on asterisk, via the provisioning. It goes something like this: A. scan for IP phones on the internet, either via spotting something on port 5060, or via the port 80 web interface for the phone. Or, use web sites that scan the internet, and classify the machines, to make your work shorter. B. Once you get into the web GUI, get the URL for provisioning. I haven't checked yet... do any phones actually allow you to set this, or do any display the current value? And, finally, how many phones publish their own MAC address in the GUI? Or, can you suck this out of the returned IP packets? C. Given the URL and the mac, fetch the phones provisioning info, including it's sip account info. Use to best advantage. D. Going further, set up a brute-force probe algorithm, to probe all possible mac addresses for a given phone manufacturer, via http requests. After all, those provisioning web servers are fast and efficient, aren't they? Collect all possible mac addresses and grab the provisioning, and now you have a LOT of sip accounts. Use to best advantage. And, professional hacking organizations seem to also follow these rules: a. wait several months for any history of the above activities to roll off the log files. Treat your phone systems like fine wine vintage. b. Use multiple (hundreds/thousands) of machines scattered over the earth to carry out the above probes, and also to use the accounts for generating international calls. In general, using the SIP account info gleaned from these kinds of efforts is a bit problematic. You see, to effectively use your phone system to place calls, they will have to set up their own phone system to act like a phone, and register to the phone system, and then initiate calls. Trouble is, your phone is usually already registered, but can be bumped off. Your phone will re-register at intervals and bump the hackers, who will again register and bump your phone. This little game of king of the hill may show up in your Asterisk logs. So, these defenses can be employed to stop/ameliorate such hacking efforts: 1. Keep your phones behind a firewall. Travellers, beware! Never leave the default login info of the phone at default! 2. Never use the default provisioning URL for the phone, with it's default URL or password. 3. Use fail2ban, ossec, whatever to stymie any brute force mac address searches. 4. Use your firewalls to restrict IP's that can access web, ftp, etc, for provisioning to just those IP's needed to allow your phones to provision. 5. Keep your logs for a couple years. 6. Change your phone SIP acct passwords now, if you haven't implemented the above precautions yet. If I missed a previous post on this, forgive me. Just thought you-all might appreciate a heads-up. murf -- Steve Murphy ParseTree Corporation 57 Lane 17 Cody, WY 82414 ✉ murf at parsetree dot com ☎ 307-899-5535 -- _ -- 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] les opérations ajouter,supprimer,modifier un client avec jEE et asterisk
On Wed, 7 May 2014, Meriem Abid wrote: salut, je suis entrain de developper une application... You will have better luck if you can post in English. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Reload problems with 1.8.27 and 11.9.0 - Someone else ?
On Thu, May 1, 2014 at 11:08 AM, Administrator TOOTAI ad...@tootai.net wrote: snip As explained in one on my previous message, it's a bug, easily reproducible: take a queues.conf (or sip.conf or iax.conf or voicemail.conf or ...) like this (what is important is the #include): snip NOTICE[3346]: app_queue.c:6811 reload_queue_rules: queuerules.conf has not changed since it was last loaded. Not taking any action. On Tue, 6 May 2014, Rusty Newton wrote: However I'm still confused as to how you are seeing the behavior you are seeing. Any chance the OP is including files from a file system that isn't maintaining atime/ctime/mtime/etc as expected, like NFS? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] how to hangup Local/100 channel
On Mon, 5 May 2014, motty cruz wrote: one of the extensions fall into a loop, I don't know how to hangup that channel -- Goto (autoatten,s,2) -- Sent into invalid extension 's' in context 'autoatten' on Local/200@sipphones-01b2;2 any ideas? If you're asking how to prevent it from happening, how about 'exten = s,2,hangup()?' If you're asking how to hang up the channel while it is in a loop, what have you tried? Does 'channel request hangup' help? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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] how to hangup Local/100 channel
Please don't top post. On Mon, 5 May 2014, motty cruz wrote: Thanks for your support, I was able to soft hangup using hangup request Local/200@users-0001b first, I did core show channels, after stopping this loops I was able to fixed that problem from happening again, On Mon, 5 May 2014, Steve Edwards asterisk@sedwards.com wrote: If you're asking how to prevent it from happening, how about 'exten = s,2,hangup()?' Note that you also could have added the 'missing' priority and reloaded your dialplan and the hangup would have been executed on the next iteration of the loop. Or if you're adventurous, there's always the 'dialplan add extension' command. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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] AGI GET DATA behavior
Le 30/04/2014 13:10, Thorsten Göllner a écrit : Is your script really so simple? Enable agi debugging (agi set debug on) and take look at it when this happens. On Thu, 1 May 2014, Hoggins! wrote: Yes, it is fairly simple, really. The problem is that Asterisk's behavior is not constant : 1 time out of 4 or 5, without ANY change in the behavior of the user, Asterisk simply does not wait for the user input, and returns 0 before the timeout. 0) Please don't top-post. 1) Reduce your script to the smallest example that illustrates your issue. 2) Post your script along with console output (with 'core agi set debug on,' 'core set debug 99,' and 'core set verbose 99') of a successful and a failed call. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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] Open Source Asterisk Polling Solution
On 4/23/2014 12:20 AM, Nick Cameo wrote: I have a strong Java, PHP and SQL background. Will probably need to make a call using AGI or such? On Wed, 23 Apr 2014, James Sharp wrote: You can go AGI, but there are direct ODBC handles available in the dialplan if you build Asterisk properly with the ODBC resources enabled. That'd my personal preference from a performance standpoint. On Wed, 23 Apr 2014, Josh Metzger wrote: I agree that ODBC is the way to go here. It's trivially easy to setup, and equally simple to push database updates via the dialplan. I've used ODBC connectivity with Asterisk in a large and VERY busy call center, and performance was never remotely an issue (call recording is a different story, but that's something else entirely...). There was mention of checking against a DNC list, and ODBC would be good for this as well - just put that into a table and match against it before making your outbound call. And in this corner... I always do database access it an AGI. IMNSHO, any significant chunk of logic or functionality belongs in an AGI. Keep your dialplan lean and mean. I tried database access in the dialplan using the mysql() application years ago, just to confirm I was right and I was :) What an ugly, messy, fragile dialplan. You already know database access in 'real' languages, why would you want to code in a limited and difficult to debug environment? I'm an 'old school' C programmer, so performance is always close to my heart, but not when it makes my job harder. Writing database access in the dialplan avoids creating a process for the AGI, but unless you're processing hundreds or thousands of calls per second, process creation is not going to be a factor. I write my AGIs in C. It is my 'sharpest tool in the toolbox.' If C is not in your 'wheelhouse,' use PHP or coughJava/cough. You (and the next guy who gets to enhance and maintain this application) will be glad you did. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Open Source Asterisk Polling Solution
On Tue, 22 Apr 2014, A J Stiles wrote: ...so absolutely *do not* pay money for a solution, and *do* insist on the Source Code and Modification Rights. Even an obvious and simple solution has value if it exceeds the OP's skill set or the value of his time to implement. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Open Source Asterisk Polling Solution
On Wed, 23 Apr 2014, Steve Edwards wrote: I tried database access in the dialplan using the mysql() application years ago, just to confirm I was right and I was :) What an ugly, messy, fragile dialplan. On Wed, 23 Apr 2014, Doug Lytle wrote: With FuncODBC this is no longer an issue. All of the query logic is handled outside of the dial plan. I took a look and it looks like a step in the right direction, kind of a 'prepared statement' approach and it gets all the ugly quoting nonsense out of the dialplan. The query statement may be out of the dialplan, but the logic of what to do with the returned values remains. The OP stated that he was going to 'will wire it up to the DNC' (the National Do Not Call Registry?) which sounds like a simple 'query the database to see if the key exists' kind of thing for which ODBC seems reasonable. This application should be expanded to include multiple databases so his callers can press 1 to be queued for an agent or 2 to be added to his client's private DNC database. While checking 2 databases is no big deal, a simple 'check-dnc' AGI can hide those details and yield a cleaner dialplan. As the application matures, there may be additional enhancements that would lean towards wishing he had started down the AGI road. If the target list includes (but is not limited to) members of a group (like a church) you could have a situation where the callee is on the DNC, but has opted-in so you have another database to consider. How about checking the database to see the last time they had 'waste they need picked up?' If the 'waste' is charitable donations of clothing or furniture, I suspect most people would be good with just a call or 2 per year. How about letting the 'donor' schedule the number of months until the next call? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Help with a bug
On Wed, 23 Apr 2014, CDR wrote: The case is this: A Record is executed and an immediate Playback follows. Asterisk returns an error, saying that the file does not exist, but a few seconds later, it does. A simple test: exten = *,n,record(foo.wav) exten = *,n,playback(foo) works as expected for me with Asterisk 11.8.1. I notice in the console log you uploaded, you have a file name of '180-industry:sln' The syntax for record says 'filename.format' not 'filename:format' -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] DAHDI loading issue on Asterisk
Hello all, At this point I am at a complete loss as to how to fix my problem. I have gone back and rebuilt the whole system from the ground up but am getting the same results. When I start asterisk with the -f option I do see what the problem appears to be. Unable to load config skinny.conf, Skinny disabled. Unable to open '/dev/dahdi/channel': Permission denied Unable to open channel 1: Permission denied here = 0, tmp-channel = 1, channel = 1 Unable to register channel '1-23' This is because the file permissions are not set correctly on the /dev/dahdi/channel: crw-rw 1 root root 196, 254 Apr 17 15:07 channel Can anyone provide me with a concise step by step guide, or point me to one, on building Asterisk 11, DADHI and all? I have pieced together different components of the process but they seem to be in error. Other than this one problem the Asterisk system works like a champ. I really need to get my wtce43x card working and ISDN PRI setup on this platform. Thanks for your assistance. Steve -- _ -- 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] DAHDI loading issue on Asterisk
Russ,
Here is what I am showing:
asteriskpbx@asterisk-pbx:/etc/udev/rules.d$ cat dahdi.rules
ACTION!=add, GOTO=dahdi_add_end
# DAHDI devices with ownership/permissions for running as non-root
SUBSYSTEM==dahdi, OWNER=asterisk, GROUP=asterisk,
MODE=0660
# Backward compat names: /dev/dahdi/channo
SUBSYSTEM==dahdi_channels,SYMLINK+=dahdi/%m
# Add persistant names as well
SUBSYSTEM==dahdi_channels, ATTRS{hardware_id}!=,
SYMLINK+=dahdi/devices/%s{hardware_id}/%s{local_spanno}/%n
SUBSYSTEM==dahdi_channels, ATTRS{location}!=,
SYMLINK+=dahdi/devices/@%s{location}/%s{local_spanno}/%n
LABEL=dahdi_add_end
# hotplug scripts
SUBSYSTEM==dahdi_devices,
RUN+=%E{DAHDI_TOOLS_ROOTDIR}/usr/share/dahdi/dahdi_handle_device
SUBSYSTEM==dahdi_spans,
RUN+=%E{DAHDI_TOOLS_ROOTDIR}/usr/share/dahdi/dahdi_span_config
Original Message
Subject: Re: [asterisk-users] DAHDI loading issue on Asterisk
From: Russ Meyerriecks rmeyerrie...@digium.com
Date: Fri, April 18, 2014 12:09 pm
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
On Fri, Apr 18, 2014 at 12:22 PM, st...@vanwambeck.net wrote:This is
because the file permissions are not set correctly on the
/dev/dahdi/channel:
Steve,
The default dahdi.rules file specifies the /dev/dahdi/ directory to be
owned by asterisk:asterisk. What's the contents of your
/etc/udev/rules.d/dahdi.rules file?
--
Russ Meyerriecks
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
direct: +1 256-428-6025
Check us out at: www.digium.com www.asterisk.org
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Re: [asterisk-users] DAHDI loading issue on Asterisk
Russ, I think that got it... I had to make a slight change to the command though: root@asterisk-pbx:/etc/dahdi# chown -R asteriskpbx:asteriskpbx /dev/dahdi Astersisk is now showing the dahdi and pri commands. I can see channels now from the pri show channels command. I think I am good to go now. Thanks for the guidance, have a wonderful weekend! Steve Original Message Subject: Re: [asterisk-users] DAHDI loading issue on Asterisk From: Russ Meyerriecks rmeyerrie...@digium.com Date: Fri, April 18, 2014 1:15 pm To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com On Fri, Apr 18, 2014 at 2:58 PM, st...@vanwambeck.net wrote: Russ, Here is what I am showing: asteriskpbx@asterisk-pbx:/etc/udev/rules.d$ cat dahdi.rules ACTION!=add, GOTO=dahdi_add_end # DAHDI devices with ownership/permissions for running as non-root SUBSYSTEM==dahdi, OWNER=asterisk, GROUP=asterisk, MODE=0660 Hmm, this looks right. Does chan_dahdi.so load up properly if you chown -R asterisk:asterisk /dev/dahdi ? -- Russ Meyerriecks Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6025 Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-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] DAHDI loading issue on Asterisk
Russ, OK, I see where the disconnect is now. As I mentioned somewhere along the line there doesn't seem to be a definative guide on building all of the things from scratch outside of the Asterisk. The Definitive Guide 3rd Edition which does touch on the Linux install and making the Asterisk user asteriskpbx. As we now see the DAHDI implementation is looking for the asterisk user. This sure can be a touch confusing. I did push the server through a restart to see if it would come up and indeed it did not. Looking at the /dev/dahdi directory everything is set back to root/root! Time to rebuild the system with the username of asterisk and go home!!! Thanks again for the assist. Steve Oh I see, udev must have been setting the ownership to root because the asterisk user/group doesn't exist on your system. # DAHDI devices with ownership/permissions for running as non-root SUBSYSTEM==dahdi, OWNER=asterisk, GROUP=asterisk, MODE=0660 In the dahdi.rules file, change the owner and group on this line to asteriskpbx so this fix will persist through a module reload. -- Russ Meyerriecks Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6025 Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-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] DAHDI loading issue on Asterisk
Sean, Yes, it is: asteriskpbx@asteriskpbx:~$ lsmod | grep dahdi dahdi 227741 2 oct612x,wcte43x crc_ccitt 12707 1 dahdi asteriskpbx@asteriskpbx:~$ Do you have the kernel module loaded? lsmod | grep dahdi sean -- _ -- 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] Dimensioning asterisk 11
On Thu, 17 Apr 2014, Jerry Geis wrote: I will be using a dell R320 Xeon E5-2420 2G and 4G RAM.also using a SIP trunk with ulaw/alaw codec. no transcoding or anything. Just call a number and play a gsm file. How will you do ulaw - gsm without transcoding? How many calls could I expect to make at the same time? A whole bunch? It's hard to give any specifics without the same hardware and workload. Here's a datapoint to consider -- testing an HP ProLiant DL320e Gen8 v2 E3-1240v3 8GB. 9300 passmarks vs your 7300 passmarks. (And only $880 from Newegg.) 2 hosts, 1 originating calls, 1 running a simple dialplan, but similar to the expected production dialplan. 500 'participants' - 100 meetme conferences with 5 calls in each. 3000 'participants' - 100 confbridge conferences with 30 calls in each. Meetme() is still a 'single thread' application so you're done when you max out 1 CPU core. 500 calls was my goal, so that's where testing stopped. The hosts aren't in production yet, so I don't know if my testing experience will match production experience. I would expect playback() (without transcoding) to be significantly less CPU hungry than meetme() or confbridge(). -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Dimensioning
On Thu, 17 Apr 2014, Jerry Geis wrote:
I was thinking transcoding was through PRI card - not gsm to ulaw. :)
You can convert the GSM files to ULAW using sox. I tend to transcode
everything to WAV (PCM not that funky 'GSM in WAV') because it is
relatively cheap (CPU cycles) to transcode from WAV to ULAW and everything
else in the world understands WAV just fine. If you really need to squeeze
out every last cycle, you can schedule a script to transcode WAVs to ULAWs
as needed.
So if all I am doing is originating calls, and using playback() in the
dialplan - then a system() call on completion I can expect upwards or
3000 concurrent calls?
Based on my unsubstantiated testing on my hosts, that seems like a
reasonable conclusion.
What do you do in the program executed by system()?
How do you actually test to make sure without having 3000 users to call.
Crowdsourcing?
No, it's really pretty simple.
On the 'source' host, I have a call file:
# sample-call-file
channel:sip/test@target
application:playback
data:/tmp/total
# (end of sample-call-file)
And a shell script to create the call files:
# create-calls.sh
cp sample-call-file /tmp/
chmod +x /tmp/sample-call-file
for I in $(seq 1 $1)
do
sudo -u asterisk\
cp /tmp/sample-call-file\
/var/spool/asterisk/outgoing/${RANDOM}
done
# (end of create-calls.sh)
Then, on the 'target' host I have a dialplan snippet:
[public]
exten = test,1, verbose(1,[${EXTEN}@${CONTEXT}])
exten = test,n, set(GROUP()=TEST)
exten = test,n, set(ROOM=0${GROUP_COUNT()})
exten = test,n, meetme(${ROOM:-2}, cd)
; exten = test,n, confbridge(${ROOM:-2})
exten = test,n, hangup()
Then, on the 'source' host, I can create calls with this command:
./create-calls.sh number-of-calls-to-create
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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[asterisk-users] DAHDI loading issue on Asterisk
Hi all, I have a fresh install of Asterisk 11.8.1 and am putting a Digium TE435 4 T1 card in it for ISDN PRI. I can get the card to be recognised by the DAHDI utilities but when I put in the file "chan_dahdi.conf" with either the generated file from samples with what seem to be appropriate settings or with the basic config as outlined on the DAHDI install guide the Asterisk "core show help" display is missing all the "dahdi" and "pri" commands.If I remove the "chan_dahdi.conf" file and restart Asterisk the commands magically reappear. I have gone back and checked on menuselect but don't see anything obvious that I have missed to support this function. I have run out of ideas on how to integrate this. The documentation makes it sound pretty simple but I have been fighting this for a week now with no success. I am not seeing any parse errors from the module reload command:asteriskpbx*CLI module reload chan_dahdi.soasteriskpbx*CLIThe truncated output from "core show help" is:core stop when convenient Shut down Asterisk at empty call volumecore waitfullybooted Wait for Asterisk to be fully booteddata get Data API getdata show providers Show data providers... resencestate change Change a custom presence statepresencestate list List currently know custom presence statesrealtime destroy Delete a row from a RealTime databaserealtime load Used to print out RealTime variables.I can restart the asteriskpbx process without the "chan_dahdi.conf" file and all the dahdi and pri commands are present. The "chan_dahdi.conf" file I am loading is a basic file from the DAHDI instructions. Even the sample file will not correctly load up either.asteriskpbx@asteriskpbx:/etc/asterisk$ cat chan_dahdi.conf[trunkgroups][channels]usecallerid = yeshidecallerid = nocallwaiting = yesusecallingpres = yescallwaitingcallerid = yesthreewaycalling = yestransfer = yescanpark = yescancallforward = yescallreturn = yesechocancel = yesechocancelwhenbridged = yesrelaxdtmf = yesrxgain = 0.0txgain = 0.0group = 1callgroup = 1pickupgroup = 1immediate = noswitchtype = 5esssignalling = pri_cpecontext = incomingechocancel = yeschannel = 1-23 Any suggestions on what I am missing would be greatly appreciated. Steve VanWambeck -- _ -- 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] DAHDI loading issue on Asterisk
Josh, Yes, I only have one span currently connected, the other 3 are looped. With the Asterisk process stopped I do see the OK on the "dahdi_tool" screen. I am not seeing any sort of errors in the /var/log/asterisk directory but when I start asterisk manually with the -f option I do get the following: Unable to open '/dev/dahdi/channel': Permission deniedUnable to open channel 1: Permission deniedhere = 0, tmp-channel = 1, channel = 1Unable to register channel '1-23' Looking at the /dev/dahdi directory I see the following: snip lrwxrwxrwx 1 root root 12 Apr 15 11:30 95 - chan/004/023lrwxrwxrwx 1 root root 12 Apr 15 11:30 96 - chan/004/024drwxr-xr-x 6 root root 120 Apr 15 11:30 chancrw-rw 1 root root 196, 254 Apr 15 11:30 channelcrw-rw 1 root root 196, 0 Apr 15 11:30 ctldrwxr-xr-x 2 root root 80 Apr 15 11:30 devicescrw-rw 1 root root 196, 255 Apr 15 11:30 pseudocrw-rw 1 root root 196, 253 Apr 15 11:30 timerroot@asteriskpbx:/dev/dahdi# cd I compiled the dahdi package under "sudo su", perhaps that is what is wrong??? Steve -- _ -- 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] DAHDI loading issue on Asterisk
(resend in plain text)... Josh, Yes, I only have one span currently connected, the other 3 are looped. With the Asterisk process stopped I do see the OK on the dahdi_tool screen. I am not seeing any sort of errors in the /var/log/asterisk directory but when I start asterisk manually with the -f option I do get the following: Unable to open '/dev/dahdi/channel': Permission denied Unable to open channel 1: Permission denied here = 0, tmp-channel = 1, channel = 1 Unable to register channel '1-23' poll() failed: Interrupted system call Looking at the /dev/dahdi directory I see the following: snip lrwxrwxrwx 1 root root 12 Apr 15 11:30 95 - chan/004/023 lrwxrwxrwx 1 root root 12 Apr 15 11:30 96 - chan/004/024 drwxr-xr-x 6 root root 120 Apr 15 11:30 chan crw-rw 1 root root 196, 254 Apr 15 11:30 channel crw-rw 1 root root 196, 0 Apr 15 11:30 ctl drwxr-xr-x 2 root root 80 Apr 15 11:30 devices crw-rw 1 root root 196, 255 Apr 15 11:30 pseudo crw-rw 1 root root 196, 253 Apr 15 11:30 timer root@asteriskpbx: I compiled the dahdi package under sudo su, perhaps that is what is wrong??? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on OSX
On Sun, 13 Apr 2014, Manu wrote: And I don't believe there's a good synths on debian7. Have you tried the Google TTS? (http://zaf.github.io/asterisk-googletts/) For a 'free' system, it sounds pretty good to me. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] ControlPlayback can not replay complicated file names
On Thu, 10 Apr 2014, Jonathan White wrote: If not sure if I am looking at a bug or expected behaviour as I do not see anything in the documentation. ControlPlayback can not replay complicated file names For example it can replay 1005 but it can not replay 1005-2014-04-08_23:58:17 On Thu, 10 Apr 2014, Eric Wieling wrote: This doesn't fix the issue, but a work around might be to try using file names without the any : in them Seems 'bug-worthy' to me. Here's some more confusion using Asterisk 11.8.1: Play a file with colons in the file name: -- Executing [*@default:3] ControlPlayback(IAX2/6002-1095, /tmp/2014-04-10-13:31:03) in new stack -- IAX2/6002-1095 Playing '/tmp/2014-04-10-13:31:03.slin' (language 'en') -- ControlPlayback seek to offset 0 from end 1) There is no 'slin' file, only a 'wav.' 2) I didn't specify any offset. Note that 'core show application controlplayback' does not mention anything about seek from end. 3) No audio is played. Play a file with quoted colons: -- Executing [*@default:3] ControlPlayback(IAX2/6002-106, /tmp/2014-04-10-13\:31\:03) in new stack -- IAX2/6002-106 Playing '/tmp/2014-04-10-13:31:03.slin' (language 'en') -- ControlPlayback seek to offset 0 from end 1) No slin file. 2) No offset. 3) No audio. 4) No quotes in 'Playing' message. -- Executing [*@default:3] ControlPlayback(IAX2/6002-461, /tmp/foo) in new stack -- IAX2/6002-461 Playing '/tmp/foo.slin' (language 'en') 1) Still no slin file. 2) /tmp/foo.wav is played, audio is heard. So... 1) You should report it as a bug. 2) Eric's right -- you should not use colons in file names. Even if controlplayback() allowed them, they always seem to cause problems somewhere. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] is g729 codec free? or under license???
On 04/09/2014 06:54 PM, Tzafrir Cohen wrote: On Wed, Apr 09, 2014 at 10:19:59AM +0800, Steve Underwood wrote: Hi Jeff, On 04/08/2014 12:13 PM, Jeff Brower wrote: Darrel- The G729 essential patents were *granted* in 1996, but applied for prior to June 8 1995. That means their lifespan is either 20 years from their application date, or 17 years from their grant date, whichever is greater (http://www.uspto.gov/main/faq/p120013.htm). Either way, they expire in 2014. -Jeff Where did you get the cutoff date of June 8 1995, and how does 20 years from that date lead to the last of the patents expiring in 2014? Nobody uses G.729. They use G.729A. The G.729A spec is somewhat later than the original G.729, but I don't know if there are any additional patents which specifically relate to Annex A. You could use G.729 instead, but it roughly doubles the compute needed. If it allows me to avoid the trolls: I'll pay that performance hit. In many caces there are CPU cycles to spare. But the licensing is a hard limit. Well, you do get the benefit of higher quality for your extra compute. G.729 sounds distinctly better than G.729A on a lot of material. There are various things on the web saying the last of the patents on G.723.1, which was around in draft form long before G.729, expires in 2014. However, there seem to be patents related to that codec which don't really expire until some time in 2015. Its really hard to find solid information. The ITU patent database rarely identifies the actual patents being claimed, so its damned hard to look them up. Nice. Regards, Steve -- _ -- 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] VPN SIP Phone | PC Traffic
I did this with SNOM phones and a special firmware a while ago. The trick to get the VPN to extend to the PC port is bridge-utils. Worked very well. On Apr 9, 2014 7:40 AM, Positively Optimistic positivelyoptimis...@gmail.com wrote: We are using vpn routers to connect home users back to our office network. Basically, shipping a mikrotik router that 'calls home' and establishes a vpn connection for the pc and phone that are connected to the mikrotik... user plugs router in, plugs phone and computer into router, and that traffic is encapsulated back to our office... simple and straighforward. We would like to remove the router from the equation... does anyone know of a SIP phone with a built in VPN client that can provide the tunnel for *both the phone and the pc traffic*? It would seem trivial to route a subnet down to the vpn client in the phone, that would be available to devices connected on the PC side of the telephone.. This would be tremendous for an at-home contact center agent..An added benefit would be to limit connections the connection on the PC side of the phone to a specific mac address.. We're aware of the opportunity to use a softphone on the pc with a vpn client. though, we're looking for a physical phone. -- _ -- 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] is g729 codec free? or under license???
Hi Jeff, On 04/08/2014 12:13 PM, Jeff Brower wrote: Darrel- The G729 essential patents were *granted* in 1996, but applied for prior to June 8 1995. That means their lifespan is either 20 years from their application date, or 17 years from their grant date, whichever is greater (http://www.uspto.gov/main/faq/p120013.htm). Either way, they expire in 2014. -Jeff Where did you get the cutoff date of June 8 1995, and how does 20 years from that date lead to the last of the patents expiring in 2014? Nobody uses G.729. They use G.729A. The G.729A spec is somewhat later than the original G.729, but I don't know if there are any additional patents which specifically relate to Annex A. You could use G.729 instead, but it roughly doubles the compute needed. There are various things on the web saying the last of the patents on G.723.1, which was around in draft form long before G.729, expires in 2014. However, there seem to be patents related to that codec which don't really expire until some time in 2015. Its really hard to find solid information. The ITU patent database rarely identifies the actual patents being claimed, so its damned hard to look them up. Regards, Steve -- _ -- 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] IAX2 Trunk Encryption
Wireshark. On Fri, Apr 4, 2014 at 11:13 AM, Elliott W dig...@private-address.infowrote: Ok, I think I am 90%+ there. Note: the configuration or status is the same on both sides unless otherwise noted. I am using RSA keys for authentication and the calls are coming through as authenticated so I'm sure that part works. The peer shows the (E) next to the status in Asterisk Info for the IAX2 peers The trunk configuration contains: encryption=yes So here is my question, Calls stop flowing when I use the directive: forceencryption=yes At the trunk level or higher does not matter, same effect. So my question comes down to, are my calls getting encrypted and why does this directive cause them to fail, AND how can I tell. Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 Trunk Encryption
Have you enabled IAX2 debugging and tried some test calls? Thanks, Steve T On Fri, Apr 4, 2014 at 6:59 PM, Elliott W dig...@private-address.infowrote: That answered my question as to whether it WAS encrypted, I think, and the answer is no, the credentials are but all the rest is not. That just leaves the question of what I need to do to get it encrypted.. Thanks. On Fri, Apr 4, 2014 at 12:59 PM, Steve Totaro stot...@totarotechnologies.com wrote: Wireshark. On Fri, Apr 4, 2014 at 11:13 AM, Elliott W dig...@private-address.infowrote: Ok, I think I am 90%+ there. Note: the configuration or status is the same on both sides unless otherwise noted. I am using RSA keys for authentication and the calls are coming through as authenticated so I'm sure that part works. The peer shows the (E) next to the status in Asterisk Info for the IAX2 peers The trunk configuration contains: encryption=yes So here is my question, Calls stop flowing when I use the directive: forceencryption=yes At the trunk level or higher does not matter, same effect. So my question comes down to, are my calls getting encrypted and why does this directive cause them to fail, AND how can I tell. 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 -- _ -- 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] Need more meetme users -- hitting some limit
On Mon, 31 Mar 2014, Shaun Ruffell wrote: If you're looking to reduce the CPU overhead of processing meetme conferences, this email from awhile ago may be of some help: http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/51750/focus=51777 Thanks for the clue. I can hit my target of 512 on an Intel E3-1240v3 with 'pre-packaged' Asterisk so I'm good for now. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk CLI Banner
On Fri, 28 Mar 2014, Richard Kenner wrote: And this certainly may vary from jurisdiction to jurisdiction. For a (quite dated at this point) discussion of this issue from a US perspective, see http://www.law.berkeley.edu/php-programs/faculty/facultyPubsPDF.php?facID=346pubID=157 The publication (43 pages) is dated 1988. The DMCA (1998) and subsequent legislation may have changed the landscape. My (ignorant) opinion -- just don't. Is it worth the effort to research? Is it worth paying a lawyer to research it and give an opinion that may be worth nothing until it is examined in court? If you want to display something custom, how about a 'wrapper' script that displays a file using 'curl' before handing off to Asterisk -- easier to implement, easier to maintain, no legal BS to consider. Or can you express your creativity by fiddling with ASTERISK_PROMPT? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] (OT) Phones with STP, DHCP, and/or (T)FTP Issues
I remember having to turn off STP or set portfast on some switch ports to some phones due to the boot sequence and timeouts of some phones a long time ago. Does anyone know which phones, if any still suffer from these problems? I am setting up a lab and want to introduce this problem for the class. Thanks, Steve T -- _ -- 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] AMI Proxy
On Mon, Mar 24, 2014 at 6:17 AM, Ishfaq Malik i...@pack-net.co.uk wrote: Just having a quick check to see if anyone is using any AMI proxies and On Mon, 24 Mar 2014, Paul Belanger wrote: All depends on the language you want to use. We used starpy for a while, but ended up rewriting our own version. Currently we're connecting AMI to a message bus and passing events across the bus. What do you see as the advantages of a message bus (dbus?)? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Need more meetme users -- hitting some limit
I'm trying to determine the capacity of my host running Asterisk 11.8.1 on CentOS 6.5. The host is an Intel E3-1240v3 with 8GB RAM, an SSD, and gigabit Ethernet. The primary application will be bridging groups of users using meetme(). I'm using 2 boxes -- 1 to initiate calls using call files (box1), and 1 behaving a bit more like a production box -- bridging calls (box2). The call file on box1 originates a call to box2 and then plays a 2 hour WAV file. The dialplan on box2 drops the call into a meetme, creating the room name from the last 2 digits of the current call count -- distributing the calls into 100 meetmes. When I run a script to create 500 call files on box1, box2 starts complaining at 312 calls, logging 'Unable to open DAHDI pseudo channel: Cannot allocate memory' on the console. From the 'callers perspective' the call is dropped between 'There are currently x other participants in the conference' and the 'beep-beep.' 'top' says Asterisk is only using about 1/2 gigabyte of RAM. 'top' says Asterisk is using about 250% of the CPU (4 physical, 8 logical cores). 'ulimit' (added to /usr/sbin/safe_asterisk in the run_asterisk() function) says the open file limit is 397,006. 'ls -l /proc/$(cat /var/run/asterisk/asterisk.pid)/fd | wc -l' says Asterisk only has 2,194 files open. 'iftop' sees about 24Mb of bandwidth in each direction between the boxes. Using confbridge() I can easily get 3,000 calls (14,869 open files, 180Mb bandwidth), but I'd lose some functionality and have to re-write parts of my application. Any clues of what limit I'm hitting and how to increase it? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Need more meetme users -- hitting some limit
On Fri, 21 Mar 2014, Steve Edwards wrote: The call file on box1 originates a call to box2 and then plays a 2 hour WAV file. The call file on box1 originates a SIP call to box2 and then plays a 2 hour WAV file. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Need more meetme users -- hitting some limit
I found below here: http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe If you have too many conferences, one CPU may not be able to mix all the audio and you will have audio problems even if there are 7+ other CPUs that are essentially idle while waiting for one CPU to mix everything. You should be able to handle 512 conference participants on a modern server system without problem. The current trunk of *DAHDI linux limits the number of open pseudo channels to 512 for this reason*. [1] Thanks, Steve T [1] http://svn.asterisk.org/view/dahdi?view=revisionrevision=9610 The new ConfBridge module in the upcoming Asterisk 1.10 release may not have this limitation. On Fri, Mar 21, 2014 at 11:53 AM, Steve Edwards asterisk@sedwards.comwrote: I'm trying to determine the capacity of my host running Asterisk 11.8.1 on CentOS 6.5. The host is an Intel E3-1240v3 with 8GB RAM, an SSD, and gigabit Ethernet. The primary application will be bridging groups of users using meetme(). I'm using 2 boxes -- 1 to initiate calls using call files (box1), and 1 behaving a bit more like a production box -- bridging calls (box2). The call file on box1 originates a call to box2 and then plays a 2 hour WAV file. The dialplan on box2 drops the call into a meetme, creating the room name from the last 2 digits of the current call count -- distributing the calls into 100 meetmes. When I run a script to create 500 call files on box1, box2 starts complaining at 312 calls, logging 'Unable to open DAHDI pseudo channel: Cannot allocate memory' on the console. From the 'callers perspective' the call is dropped between 'There are currently x other participants in the conference' and the 'beep-beep.' 'top' says Asterisk is only using about 1/2 gigabyte of RAM. 'top' says Asterisk is using about 250% of the CPU (4 physical, 8 logical cores). 'ulimit' (added to /usr/sbin/safe_asterisk in the run_asterisk() function) says the open file limit is 397,006. 'ls -l /proc/$(cat /var/run/asterisk/asterisk.pid)/fd | wc -l' says Asterisk only has 2,194 files open. 'iftop' sees about 24Mb of bandwidth in each direction between the boxes. Using confbridge() I can easily get 3,000 calls (14,869 open files, 180Mb bandwidth), but I'd lose some functionality and have to re-write parts of my application. Any clues of what limit I'm hitting and how to increase it? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Need more meetme users -- hitting some limit
On Fri, 21 Mar 2014, Steve Totaro wrote: I found below here: http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe If you have too many conferences, one CPU may not be able to mix all the audio and you will have audio problems even if there are 7+ other CPUs that are essentially idle while waiting for one CPU to mix everything. You should be able to handle 512 conference participants on a modern server system without problem. The current trunk of DAHDI linux limits the number of open pseudo channels to 512 for this reason. [1] With 312 calls distributed across 100 meetmes, 'top' shows 1 core at 32%, 1 core at 6% and the rest basically idle. So it looks like meetme() is still a single CPU application, but I have plenty of CPU headroom. Coincidentally, 512 is my target. Any clues on how to get 200 more? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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] Need more meetme users -- hitting some limit
On Fri, Mar 21, 2014 at 2:26 PM, Steve Edwards asterisk@sedwards.comwrote: On Fri, 21 Mar 2014, Steve Totaro wrote: I found below here: http://www.voip-info.org/ wiki/view/Asterisk+cmd+MeetMe If you have too many conferences, one CPU may not be able to mix all the audio and you will have audio problems even if there are 7+ other CPUs that are essentially idle while waiting for one CPU to mix everything. You should be able to handle 512 conference participants on a modern server system without problem. The current trunk of DAHDI linux limits the number of open pseudo channels to 512 for this reason. [1] With 312 calls distributed across 100 meetmes, 'top' shows 1 core at 32%, 1 core at 6% and the rest basically idle. So it looks like meetme() is still a single CPU application, but I have plenty of CPU headroom. Coincidentally, 512 is my target. Any clues on how to get 200 more? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 What does the console say for channels when you max out? That limitation has to be in the source code if in fact that is the limit you are bumping into. Thanks, Steve T -- _ -- 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] Need more meetme users -- hitting some limit
On Fri, 21 Mar 2014, Adrian Serafini wrote: Upgrade to 1.4? hehe, I thought you were the self proclaimed 1.2 luddite? I'm a big fan of older releases with 1 year plus of uptime. Yep, that's me :) I'm trying to make the leap from 1.2 to 11.8.1 -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Need more meetme users -- hitting some limit
On Fri, 21 Mar 2014, Paul Belanger wrote: DAHDI has a pseudo channel limit of 512, somebody has already posted how to change it with modprode. Not in this thread, but big thanks for the clue. Googling 'dahdi pseudo channel limit modprobe' showed the secret sauce. I can get 1,000 simultaneous callers in 100 meetmes with only an occasional crackle -- way over my 500 target. Since DAHDI has a default limit of 512 and I was peaking out at 312 callers in 100 meetmes, that implies each caller takes a DAHDI channel and each meetme takes 2. Is that about right? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Need more meetme users -- hitting some limit
On Fri, 21 Mar 2014, Steve Edwards wrote: Googling 'dahdi pseudo channel limit modprobe' showed the secret sauce. Oops. Guess I should complete the thread... You can set the DAHDI pseudo channel limit in /etc/modules.conf: options dahdi max_pseudo_channels=x or you can set it from the command line like: echo x /sys/module/dahdi/parameters/max_pseudo_channels It appears you need 1 DAHDI pseudo channel per caller and 2 pseudo channels per meetme. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Need more meetme users -- hitting some limit
Is there any good documentation on that process? On Fri, Mar 21, 2014 at 3:36 PM, John Novack jnov...@stromberg-carlson.orgwrote: Steve Edwards wrote: On Fri, 21 Mar 2014, Adrian Serafini wrote: Upgrade to 1.4? hehe, I thought you were the self proclaimed 1.2 luddite? I'm a big fan of older releases with 1 year plus of uptime. Yep, that's me :) I'm trying to make the leap from 1.2 to 11.8.1 That is a HUGE leap Watch out for whiplash! John Novack -- Dog is my Co-pilot -- _ -- 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] Need more meetme users -- hitting some limit
On Fri, 21 Mar 2014, Steve Edwards wrote: I'm trying to make the leap from 1.2 to 11.8.1 On Fri, 21 Mar 2014, Steve Totaro wrote: Is there any good documentation on that process? I haven't looked. I know they added a few of variables to the AGI environment Asterisk passes to your AGI on STDIN. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Billing number vs. CallerID number | Asterisk 11.5.1
Gateway computers rejects calls like this. I was informed that their carrier rejects the calls because they cannot accurately bill. It seems pretty silly with voip and number portability. Thanks, Steve T On Mar 17, 2014 5:19 PM, Eric Wieling ewiel...@nyigc.com wrote: Often it is P-Asserted-ID, but depends on the carrier. You should be asking your carrier how to do this. Be careful, if the carrier doesn't like your CID spoofing they might bill the call to a default number on the account. I suspect it is the destination which is rejecting the call because toll free numbers are not considered valid, not your carrier rejecting the call. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Positively Optimistic Sent: Monday, March 17, 2014 4:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Billing number vs. CallerID number | Asterisk 11.5.1 In a multi-tenant environment, we are sending various CallerIDs outbound from asterisk based on who the user is. We have an insurance agency who would like to present a toll free callerid. This works.. unless they're calling a toll free number. In that case, occasionally, the call fails. However, should we send a correctly formatted npanxx of a local number, the call completes. We have been advised that we can send the billing telephone number as a separate header and the call will complete, all-the-while, presenting the toll free number as the caller id. Does anyone know of the correct header required to provide this functionality? -- _ -- 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] Replying to Posts
On Thu, 13 Mar 2014, Ron Wheeler wrote: -1 Prefer top posting. Your preferences are in conflict with the mailing list rules (http://www.asterisk.org/community/discuss), specifically #5. It has to be all one way or the other. This is an English language list. Thus, the natural expectation is top to bottom, left to right, answers follow questions. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Oddity with FFA
On 03/11/2014 12:36 AM, Mike Diehl wrote: Hi all, For the most part, we are finding that Fax for Asterisk works pretty well. However, we have seen some wierdness that we'd like to try to fix. Once in a while, we will get a partial result report for a given fax but when we look at the actual .tiff image, it looks to be complete. This is causing our users to not get a positive acknowledgement when they send the fax. Is there anything we can do to mitigate this? Mike. How do you know the FAX is complete? If a page was received, the sending machine said more pages were to follow, and then it dropped the call, is that a complete FAX? Steve -- _ -- 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] Remote extensions call drops after 20 seconds.
Check here: http://www.freepbx.org/forum/freepbx/installation/incoming-calls-drop-30-seconds-on-fpbx-2-9-0-1-8-7-0 Thanks, Steve Totaro On Mon, Mar 10, 2014 at 4:43 PM, alp...@gmail.com alp...@gmail.com wrote: Yes, well, really is Elastix. Hmmm where I need to pt directmedia=no ? Thanks, On Mon, Mar 10, 2014 at 2:38 PM, Eric Wieling ewiel...@nyigc.com wrote: Try ulaw instead of g729, set directmedia=no I see you are using FreePBX. I cannot help further. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@gmail.com Sent: Monday, March 10, 2014 4:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: and...@telesip.net Subject: Re: [asterisk-users] Remote extensions call drops after 20 seconds. Guys, hi. I have not solved the problem. Outgoing calls to remote extensions drops on 5-20 seconds. Incoming calls work perfectly. Hope you can help me. Attach updated logs: http://pastebin.com/HmKEDAUq Thanks, On Thu, Dec 19, 2013 at 8:48 AM, Eric Wieling ewiel...@nyigc.com wrote: See sip.conf.sample in the Asterisk tarball for documentation of valid settings. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@gmail.com Sent: Wednesday, December 18, 2013 9:30 PM To: and...@telesip.net; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Remote extensions call drops after 20 seconds. I set canreinvite=very in the remote extension, and now the call not drops. Valid solution? On Wed, Dec 18, 2013 at 6:38 PM, Andres and...@telesip.net wrote: On 12/18/13, 3:09 PM, alp...@gmail.com wrote: Hello. I have a problem with the configuration of a remote extensions. Calls are truncated at 20 seconds. I got my my NAT firewall properly configured. Here I attached my debug in CLI: http://pastebin.com/gh34E69f When the call is setup I see your Asterisk retransmitting the SIP/2.0 200 OK packet many times and getting no response. The other end needs to receive the packet and generate an ACK. You need to trace where that packet is going and figure out why it is not reaching its target, or if it is, then why is the ACK not making it back. Thats your problem. Thank you! -- Allan Porras http://allanPorras.com http://www.AllanPorras.com Google Plus: http://goo.gl/BRkbX Twitter: @alpocr http://twitter/alpocr -- Technical Support http://www.cellroute.net -- _ -- 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 -- Allan Porras http://allanPorras.com http://www.AllanPorras.com Google Plus: http://goo.gl/BRkbX Twitter: @alpocr http://twitter/alpocr -- _ -- 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 -- Allan Porras http://allanPorras.com http://www.AllanPorras.com Google Plus: http://goo.gl/BRkbX Twitter: @alpocr http://twitter/alpocr -- _ -- 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 -- Allan Porras http://allanPorras.com http://www.AllanPorras.com Google Plus: http://goo.gl/BRkbX Twitter: @alpocr http://twitter/alpocr -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs
Re: [asterisk-users] Oddity with FFA
Hi Mike, If the sending machine keeps trying it might be the call has been hung up by asterisk before its own acknowledgement message has finished being sent. There have been bugs like this in the past, and people can be pretty casual about making changes which hang up aggressively. A FAX system should really wait for the final DCN message before disconnecting, to ensure both sides have seen what they need. Spandsp does that, but I am not sure about FFA. Regards, Steve On 03/11/2014 03:03 AM, Mike Diehl wrote: Steve, I BELIEVE the fax is complete because the fax image is a form that appears to only be a single page. But, since FFA isn't providing acknowledgement, the sending fax machine is resending the document multiple times! Mike. On Mon, Mar 10, 2014 at 12:49 PM, Steve Underwood ste...@coppice.org mailto:ste...@coppice.org wrote: On 03/11/2014 12:36 AM, Mike Diehl wrote: Hi all, For the most part, we are finding that Fax for Asterisk works pretty well. However, we have seen some wierdness that we'd like to try to fix. Once in a while, we will get a partial result report for a given fax but when we look at the actual .tiff image, it looks to be complete. This is causing our users to not get a positive acknowledgement when they send the fax. Is there anything we can do to mitigate this? Mike. How do you know the FAX is complete? If a page was received, the sending machine said more pages were to follow, and then it dropped the call, is that a complete FAX? Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Temporarily placing confbridge participants on hold - two way muting
Is there a way of temporarily suspending participants in a conference? Say I have 5 users, A,B,C,D,E and I wish A, B and C to have a discussion in the confbridge session that D and E can't hear, is there a way to suspend D and E for a while (whilst they are played music or whatever) and later join them back in? Failing that, I was considering kicking them and using an AGI script to rejoin them to the conference but I wasn't sure how to do that from the script (the rejoin, not the kick)? Any pointers or suggestions welcomed. (In a nutshell it's for a situation where certain participants need to have privacy in the conference from a group of others and it all needs to be driven from an AGI script). Regards Steve The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email 'techni...@brendata.co.uk' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk -- _ -- 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] h extension isn't processed after call file finishes.
On Mon, 17 Feb 2014, Mike Diehl wrote: Is there something I need to do in order to get the h extension to get called? Would the 'g' dial() option help? Proceed with dialplan execution at the current extension if the destination channel hangs up. It won't take you to h, but it may allow you to do what you need to do -- even if the next dialplan priority just says 'goto h.' -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk NAT
On Wed, 19 Feb 2014, Gholamreza Sabery wrote: Hello, a few days ago I sent a question: http://lists.digium.com/pipermail/asterisk-users/2014-February/282241.html but no one answered me! I just want to know is it possible or not? If it were only so easy... Participation in these lists is purely voluntary. You only get a reply if you managed to pique somebody's interest and they feel they have something to offer -- which may be commiseration rather than an answer. Having said all that, there are some incredibly knowledgeable and generous participants who have helped me out of some sticky situations. Think of it like a message in a bottle. You cast it out to sea and you may make an incredible contact. You may not. Something to keep in mind. These lists is largely 'US centric' by which I mean that if you post after the US work day ends (even accounting for 'programmer hours') you are limiting your potential audience. Posting late on a Friday afternoon can be an exercise in futility. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] How does extensions.lua compares to extensions.conf ?
On Wed, 12 Feb 2014, Olivier wrote: How does extensions.lua compares to extensions.conf or extensions.ael on stability, performance and features ? I'm a 1.2 Luddite, but... I used AEL for a system a couple of years ago. Even suffering through some syntactical inconsistencies and parsing bugs and a general lack of meaningful error messages when loading the dialplan, the result was a much more maintainable system. It was very refreshing being able to program in a 'real' programming language rather than something reminiscent of a deck of punch cards :) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Lots of calls, less memory
On Mon, 10 Feb 2014, Justin Sherrill wrote: We're running Asterisk 1.8 on a 32-bit Debian machine, and it has been fine for some time now. But! We've got such a incoming call volume over the few weeks that we'll have Asterisk occasionally restart itself. My hunch is that it is in part memory pressure. I'm a 1.2 Luddite, but... I suspect it's not RAM. I have a CentOS 32bit box with 2GB that has 350 calls right now (last night's peak was 420), most in meetme conferences. How many concurrent calls are you handling? How much RAM is Asterisk consuming on your box? Any obese AGIs? I run tons of AGIs, but I write them in C. Does 'vmstat 5' show swapping? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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
