Re: [asterisk-users] (no subject)
On 9 Feb 2015, at 15:32, Francisco Leonardo Mota francisco.m...@rnp.br wrote: Submission. Thanks, Uh, no problem?.. 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] (no subject)
If you're using a redhat based distro, have you checked SELinux? Try disabling (will require a server reboot) Regards Ish On 3 September 2014 20:41, Steve Edwards asterisk@sedwards.com wrote: 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 -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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)
Asterisk is not started. Start asterisk or look at the logs if there is any issues . Try asterisk -vvvgc and debug From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anthony Azzopardi Sent: Wednesday, September 03, 2014 11:57 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] (no subject) Hello asterisk-users, Just compiled and installed 11.12.0 however when I try to connect with rasterisk I get: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) It seems that asterisk.ctl is not created. -- _ -- 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)
Did you start the Asterisk server? jg -- _ -- 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] (no subject)
thanks for your response
with the code below i can't get the extenssions 223
exten = 529,1,Answer()
exten =
529,n,MixMonitor(test_num-${CALLERID(num)}_name-${CALLERID(name)}_${EXTEN}_UID-${UNIQUEID}.wav|av(0)V(0))
exten = 529,n,Dial(SIP/223)
exten = 529,n,Hangup()
i can get my number only with uniqueid
test_num-0661xx_name-_529_UID-1376564701.1204.wav
any help please
thanks and regards
2013/8/13 Positively Optimistic positivelyoptimis...@gmail.com
Define it as a variable, use the variable to define the filename
Ex.
exten =
529,n,Set(monfile=num-${CALLERID(num)}_name-${CALLERID(name)}_${EXTEN}_UID-${UNIQUEID})
exten = 529,n,MixMonitor(/var/spool/disa/${monfile}.wav,,)
hello list,
i have asterisk 1.4 installed i use MixMonitor to record all the inboud
calls with the code below my question how can i do to save alse the sip
extenssion 223
exten = 529,1,Answer()
exten = 529,n,MixMonitor(test_${UNIQUEID}.wav|av(0)V(0))
exten = 529,n,Dial(SIP/223)
exten = 529,n,Hangup()
thanks and regards
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Re: [asterisk-users] (no subject)
Define it as a variable, use the variable to define the filename
Ex.
exten =
529,n,Set(monfile=num-${CALLERID(num)}_name-${CALLERID(name)}_${EXTEN}_UID-${UNIQUEID})
exten = 529,n,MixMonitor(/var/spool/disa/${monfile}.wav,,)
hello list,
i have asterisk 1.4 installed i use MixMonitor to record all the inboud
calls with the code below my question how can i do to save alse the sip
extenssion 223
exten = 529,1,Answer()
exten = 529,n,MixMonitor(test_${UNIQUEID}.wav|av(0)V(0))
exten = 529,n,Dial(SIP/223)
exten = 529,n,Hangup()
thanks and regards
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Re: [asterisk-users] (no subject)
On Friday 12 April 2013, Thomas Perron wrote: Basic Dial Plan Why is this plan not engaging the line exten = 105,n,Dial(SIP/voipvoip.com/1703501) and dialing the 703 number? The logs and debug dont show any problems [incoming] exten = 44,1,Answer() exten = 44,n,Wait(1) exten = 44,n,Playback(beep) exten = 44,n,Goto(105,105,1) ; ; [105] exten = 105,1,Wait(2) exten = 105,n,Playback(hello-world) exten = 105,n,Dial(SIP/voipvoip.com/1703501) exten = 105,n,Hangup() Have you included the [105] context within the default context for the extension from which you are dialling 105? If 44 from the outside world is failing to trigger it, then it's possible that Asterisk is seeing the first 105 in Goto(105,105,1) as a priority rather than a context,extension,priority. Rename the [105] context to start with a letter. -- AJS Answers come *after* questions. -- _ -- 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)
check this out http://msnbc.msn.com-report6.us/finance/-- _ -- 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)
On Monday 30 July 2012, akhilesh chand wrote: Hi, I'm not able to configure 8 port card whenever I configure it is showing fatal: error inserting wct4xxp(/lib/modules/2.6.18-128.el5/dahdi/wct4xxp/wct4xxp.ko):unknown symbol in module, or unknown parameter It sounds as though you need to recompile DAHDI-Linux. (Did you compile it before you acquired this card?) Just download the latest DAHDI package Source Code, and compile and install it. If you didn't compile your own kernel from Source Code, then you will also need the package kernel-devel (on Fedora / CentOS) or linux-headers (on Ubuntu). -- AJS Price Engines Ltd. DDI: 01283 707058. -- AJS Answers come *after* questions. -- _ -- 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)
Thanks ajs On Monday, July 30, 2012, A J Stiles wrote: On Monday 30 July 2012, akhilesh chand wrote: Hi, I'm not able to configure 8 port card whenever I configure it is showing fatal: error inserting wct4xxp(/lib/modules/2.6.18-128.el5/dahdi/wct4xxp/wct4xxp.ko):unknown symbol in module, or unknown parameter It sounds as though you need to recompile DAHDI-Linux. (Did you compile it before you acquired this card?) Just download the latest DAHDI package Source Code, and compile and install it. If you didn't compile your own kernel from Source Code, then you will also need the package kernel-devel (on Fedora / CentOS) or linux-headers (on Ubuntu). -- AJS Price Engines Ltd. DDI: 01283 707058. -- AJS Answers come *after* questions. -- _ -- 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] (no subject)
Hi sam,
Have solved the problem with your advice. Call drop in 10 seconds without
disconnecting a-party call. Thank you very much.
[TB]
exten =_X.,1,Wait(${INCOMING_WAIT})
exten =_X.,2,Verbose(TB)
exten =_X.,3,Answer()
exten =_X.,4,Set(mainLoop=0)
;exten =_X.,5,Set(TIMEOUT(absolute)=5)
exten =_X.,5,Playback(/var/callagent/prompts/monitor/thanks)
exten = _X.,6,Dial(DAHDI/7/
09501032209,100,L(3[:1][:3000])g)
exten =_X.,7,Noop(${DIALEDTIME})
exten =_X.,8,Goto(TB,_X.,1)
exten =_X.,n,Hangup()
Cheers
Vinod Dharashive
Sent from BlackBerry® on Airtel
-Original Message-
From: Sam Govind govoi...@gmail.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Wed, 7 Sep 2011 11:53:33
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] (no subject)
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Re: [asterisk-users] (no subject)
See absolute timeout. I think yours' a complex thing to achieve I guess
absolute timeout may be the thing that can help. In older versions
absoluteTimeoute(n) could take you to exten T when time n elapsed. now I
guess funtion Timeout() is used as replacement.
here's an excerpt from somewhere:
; limit calls to ex-girlfriend to 300 seconds
exten = 123,1,Set(TIMEOUT(absolute)=300)
exten = 123,2,Dial(${EX-GIRLFRIEND})
exten = T,1,Playback(im-sorry)
exten = T,2,Playback(vm-goodbye)
exten = T,3,Hangup( )
Also see if Dial() command options L(x:y:z), or S(x) work out for you when
combined with option g.
On Wed, Sep 7, 2011 at 7:42 AM, Vinod Dharashive vdharash...@gmail.comwrote:
Hi team,
I am trying to find solution to hangup b-party call after 1 min with out
disconnecting the call of a-party but following dial plan which is
disconnect both the calls.
Please suggest me the solution.
[TB]
exten = _X.,1,Wait(${INCOMING_WAIT})
exten =_X.,2,Verbose(TB)
exten =_X.,3,Answer()
exten = _X.,4,Set(mainLoop=0)
exten = _X.,5,Set(TIMEOUT(absolute)=10); set time in milliseconds
exten = _X.,6,Playback(/var/callagent/prompts/monitor/thanks)
exten = _X.,7,Dial(DAHDI/7/
09501032209,10,S(60))
exten = _X.,8,Noop(${DIALEDTIME})
exten =_X.,9,Goto(TB,_X.,1)
exten =_X.,n,Hangup()
Thanks
Vinod Dharashive
Sent from BlackBerry® on Airtel
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Re: [asterisk-users] (no subject)
you running GSM FWTs with asterisk ? On Mon, Apr 25, 2011 at 6:51 AM, Abid Saleem abid_aster...@hotmail.comwrote: HI, I am trying to setup a Class 4 termination setup using a kind of channel hunting scenerio. I have some SIP DID numbers assigned from the local telecom provider for termination. MY call comes from my wholesale client and lands on a switch, then it is routed to asterisk. I want asterisk to route this call to my local DID provider on the next available channel with DID number as the new Caller ID. This is just like GSM gateway that recieves the call and then re-originates the call using the next available SIM card number. Can someone help me how can I configure Asterisk to perform this? Thanks Abid. -- _ -- 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 -- Regards: (Muhammad υѕмαη ) -- _ -- 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)
Anyone going to remove this spammer/scammer? 2010/12/19 Dmitry Kupchinetsky dkupchinet...@hotmail.com: http://www.barenakedbabies.com/shop/images/images.html -- _ -- 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] (no subject)
On Sat, Oct 16, 2010 at 4:35 PM, Dan Journo d...@keshercommunications.comwrote: Hi, Does anyone know where this is suddenly coming from? -- _ -- 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 Where what is suddenly coming from? Cheers - The Mick -- _ -- 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)
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James A. Shigley Sent: Friday, July 16, 2010 11:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] (no subject) Ok I have a queue that is working perfectly. The problem is when one of the agents is outside the building on an external phone line (say a cell phone). My telco hangs up on the call . I think the telco is hanging up on these calls because there is no CID attached. (I know my telco wont connect calls without ANI, so that is what it is my assumption) So first I need to prove my assumption is right. How can I check if those calls are being sent with caller ID. Because all I see on console output for the phone call is this -- Channel 0/8, span 3 got hangup, cause 50 (sometimes cause 1 or 27 instead) -- Nobody picked up in 1000 ms -- Hungup 'DAHDI/56-1' It doesn't show where it actually tried to dial or not. I know it works because if I sent it to the in house number it calls that number and if someone answers it they get the person who is on hold in the queue. It only fails on outside the building calls. So where do I check to see if it is or isn't attaching caller ID. Let's assume I'm right and the CID is the issue; What config and/or context do I need to change so that the when a queue tries to place a call to an agent there is caller ID? James Shigley -- 1. obviously it did dial, otherwise you wouldn't get nobody picked up 2. in your dialplan, put this line before queue Exten = 1,1,Set(CALLERID(num)=201212) - change 1,1 to context appropriate values and 201212 to a proper DID for your location. Do these this a post a CLI output with verbose set to 5 or higher. -- _ -- 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
On Fri, Jul 9, 2010 at 12:57 PM, Mike Ely mike...@amyskitchen.net wrote: Has anyone figured out how to detect the actual cellphone answer rather than the bogus one sent by the cell carrier? *CLI core show application AMD -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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)
On Fri, Mar 19, 2010 at 3:13 AM, Zeeshan Zakaria zisha...@gmail.com wrote: Fail2ban is a must. I was a victim of such attacks, and have implemented some other measures too, but fail2ban is a must have with the link posted by Matt which describes how to set it up for asterisk. Make sure you put your own ip address in ignore list otherwise it can block you too. You may also consider to use BFD (Brute Force Detection) [1] as your tool for log analysis. We have a detailed tutorial [2] on how to install and configure BFD, using Asterisk rules [3] for SIP and IAX protocols. Our approach is not to use iptables but to block the communication with the attacker using route del -host $ATTACK_HOST reject. To unban a specific IP we will use a manual command like route del -host $ATTACK_HOST reject. This is not probably not the best method but it works for us till now. Best regards, Ioan. [1] - http://www.rfxn.com/projects/brute-force-detection/ [2] - http://www.modulo.ro/Modulo/ro/Articole/Securitate_pentru_servere_Asterisk.html [3] - http://www.modulo.ro/Modulo/downloads/tools/tenora.bfd.tar.gz -- _ -- 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)
On 19/03/10 1:19 PM, Adrian Marsh wrote: Hello, I’m looking for some advice on securing Asterisk. Have a look at fail2ban: http://www.voip-info.org/wiki/view/Fail2Ban+%28with+iptables%29+And+Asterisk -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- 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)
On Fri, 19 Mar 2010, Adrian Marsh wrote: I’m looking for some advice on securing Asterisk. My first step will be to strengthen the passwords in use, and for the hardphones to restrict by IP address, but that still leaves the softphone quite widely open. Asterisk doesn't differentiate between a hard phone and a soft phone. You can restrict by IP address for soft phones as well. Does Asterisk 1.6 have anything in it that can automatically block out an attacking IP, say if it receives several 20 or so failed attempts from that IP in x minutes? I'm a 1.2 Luddite, so I can't speak for 1.6. I think any brute force or DOS security policy needs to be implemented external to Asterisk. I don't think there are any AMI events you could listen to. I think you are limited to what you can scrounge out of a log file. How about setting up a couple of honey-pot SIP accounts with obvious passwords and in the context fire off a user event? Then you could listen for the event via AMI. Any other suggestions? Repost with a meaningful subject -- a blank subject labels you as a newbie who is probably not worth the time of members with relevant experience. -- 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)
Fail2ban is a must. I was a victim of such attacks, and have implemented some other measures too, but fail2ban is a must have with the link posted by Matt which describes how to set it up for asterisk. Make sure you put your own ip address in ignore list otherwise it can block you too. On 2010-03-18 8:45 PM, Matt Riddell li...@venturevoip.com wrote: On 19/03/10 1:19 PM, Adrian Marsh wrote: Hello, I’m looking for some advice on securing Asteri... Have a look at fail2ban: http://www.voip-info.org/wiki/view/Fail2Ban+%28with+iptables%29+And+Asterisk -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- 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] (no subject)
If you read your message all the way to the end, and every posting, you will discover exactly how to do that on your own. asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users nasar mahmud wrote: Please descard me from the asterisk users list...thanks (Abu Nasar Mahmud) Checked by AVG - www.avg.com Version: 9.0.733 / Virus Database: 271.1.1/2660 - Release Date: 01/31/10 14:35:00 -- Dog is my co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
After doing a little research on this, the answer is a limited yes. Asterisk has 6 logging files to be used. If you aren't using all 6, you could designate any unused files to a context and use the log application to feed that specific log file. Since you would be doing this in a custom fashion, you could manually roll that log with a system command at the top of the context. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mickael ropars Sent: Tuesday, October 20, 2009 3:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] (no subject) All, I want to know if it's possible to create a log file per context? and each time a context is restarted a ne x log file is created. regards Mickael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
On Tue, 20 Oct 2009, mickael ropars wrote:
I want to know if it's possible to create a log file per context? and
each time a context is restarted a ne x log file is created.
This is not clear to me. Contexts are not restarted. What are you trying
to log?
Asterisk has the system() application which will execute any arbitrary
Linux command line so you can do pretty much anything.
Asterisk doesn't have the native ability to create log files as I think
you described. How would you handle 2 calls entering the same context at
effectively the same time? There are race conditions to consider both
for file creation and writing.
Maybe this will give you some ideas:
[wildcard-test]
exten = _!,1, verbose(1,[${CONTEXT}:${EXTEN}])
exten = _!,n, system(logger -i -p local0.info -t
${CONTEXT} ${CALLERID(num)} entered context)
exten = _!,n, answer()
exten = _!,n, hangup()
exten = _x,4, playback(demo-congrats)
exten = _x,n, system(logger -i -p local0.info -t
${CONTEXT} ${CALLERID(num)} finished)
exten = _x,n, hangup()
exten = h,2,system(logger -i -p local0.info -t
${CONTEXT} ${CALLERID(num)} hung up)
exten = h,n,hangup()
This will log every entry to the context to syslogd. You can configure
syslogd (/etc/syslog.conf) to separate the log entries as desired.
This is pretty inefficient -- it creates at least 4 processes (2 on entry,
2 on hangup) for every call.
I had an application several years ago that required logging how long each
caller was in each context. I used resetcdr(w) and enhanced
cdr_addon_mysql.c. When the call finished, I executed an AGI that added up
the cdrs and rated the call.
If you post questions with meaningful subject lines, you may attract the
interest of someone who has solved your exact problem and you make it
easier for the next guy to research.
--
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] (no subject)
- ameu...@yahoo.fr wrote: I have to develop a VoIP application. I need to know how to use Java APIs to communicate to my client application with asterisk. I tried looking for some answers based upon your subject but nothing came up. This may be what you're looking for: http://lmgtfy.com/?q=asterisk+java+api --Tim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
On 19 Mar 2009, at 15:08, ameu...@yahoo.fr wrote: I have to develop a VoIP application. I need to know how to use Java APIs to communicate to my client application with asterisk. Ok. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
use ami http://www.voip-info.org/wiki/view/Asterisk+manager+Example%3A+Java or Ajam http://www.voip-info.org/wiki/view/Aynchronous+Javascript+Asterisk+Manager+(AJAM) 2009/3/19 ameu...@yahoo.fr I have to develop a VoIP application. I need to know how to use Java APIs to communicate to my client application with asterisk. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Asterisk user number: 1099 Linux user: #443184 shazaum.googlepages.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
Right On Mon, Feb 23, 2009 at 9:07 PM, Lê Văn Hòa ho...@inet.vn wrote: ko gui nua -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
What asterisk cli shows when you soft hangup these channels Shariq On Fri, Sep 5, 2008 at 11:55 PM, Bill Andersen [EMAIL PROTECTED]wrote: V 1.4 When I do a show channels I get the following. CLI show channels Channel Location State Application(Data) SIP/7110-b495d3b0[EMAIL PROTECTED]:2Up Page(Local/[EMAIL PROTECTED]Local/71 SIP/7110-afd286e0[EMAIL PROTECTED]:2Up Page(Local/[EMAIL PROTECTED]Local/71 2 active channels 2 active calls I need to kill these SIP channels, but the only thing I have found when searching is the soft hangup solution - which simply doesn't do anything to these channels. CLI soft hangup SIP/7110-b495d3b0 CLI soft hangup SIP/7110-afd286e0 CLI show channels Channel Location State Application(Data) SIP/7110-b495d3b0[EMAIL PROTECTED]:2Up Page(Local/[EMAIL PROTECTED]Local/71 SIP/7110-afd286e0[EMAIL PROTECTED]:2Up Page(Local/[EMAIL PROTECTED]Local/71 2 active channels 2 active calls Can someone suggest a better way of getting rid of these channels? My solution so far has been to restart Asterisk... not a good solution. Thanks Bill ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
Hi - I'm trying to install a fresh copy of asterisk on a 64bit platform. I'm using CentOs 5.1 and all the latest builds of zaptel, libpri and asterisk. When I try to build Asterisk this is the error I'm getting. src/add.c:1: error: CPU you selected does not support x86-64 instruction set You may not have the right sources for your kernel. You may have the 32-bit sources instead of the 64-bit ones. What kind of CPU is it? - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
Use SendDTMF. --- On Thu, 7/3/08, Neha Punia [EMAIL PROTECTED] wrote: From: Neha Punia [EMAIL PROTECTED] Subject: [asterisk-users] (no subject) To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com Date: Thursday, July 3, 2008, 10:29 AM Hi I m making a call from one asterisk server to an asterisk client The call gets completed but I want it to send dtmf signals The dialplan I have made for this is like exten = 205,1,Answer exten = 205,n,Wait(15) exten = 205,n,Playback(dtmf-1) exten = 205,n,Wait(20) but it does not send any dtmf signal where is the problem?? CAUTION - Disclaimer * This e-mail contains PRIVILEGED AND CONFIDENTIAL INFORMATION intended solely for the use of the addressee(s). If you are not the intended recipient, please notify the sender by e-mail and delete the original message. Further, you are not to copy, disclose, or distribute this e-mail or its contents to any other person and any such actions are unlawful. This e-mail may contain viruses. Infosys has taken every reasonable precaution to minimize this risk, but is not liable for any damage you may sustain as a result of any virus in this e-mail. You should carry out your own virus checks before opening the e-mail or attachment. Infosys reserves the right to monitor and review the content of all messages sent to or from this e-mail address. Messages sent to or from this e-mail address may be stored on the Infosys e-mail system. ***INFOSYS End of Disclaimer INFOSYS***___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
But if I m using this SendDTMF it does not send anything
I m using it like this in extension.conf
exten = 205,1,Answer
exten = 205,n,Wait(20)
exten = 205,n,Playback(dtmf-1)
exten = 205,n,Wait(20)
exten = 205,n,SendDTMF(9)
exten = 205,n,Wait(5)
exten = 205,n,Read(digito)
exten = 205,n,SayDigits(${digito})
exten = 205,n,Hangup
on the console it only shows tht the call completed and no message about the
DTMF and in the log files it shows like :
Jul 3 17:21:01 DEBUG[896] chan_sip.c: Stopping retransmission on '[EMAIL
PROTECTED]' of Request 102: Match Found
Jul 3 17:21:27 DEBUG[896] chan_sip.c: Setting NAT on RTP to 0
Jul 3 17:21:27 DEBUG[896] chan_sip.c: Outgoing Call for 205
Jul 3 17:21:27 DEBUG[896] chan_sip.c: (Provisional) Stopping retransmission
(but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found
Jul 3 17:21:27 DEBUG[896] chan_sip.c: Acked pending invite 102
Jul 3 17:21:27 DEBUG[896] chan_sip.c: Stopping retransmission on '[EMAIL
PROTECTED]' of Request 102: Match Found
Jul 3 17:21:27 DEBUG[896] chan_sip.c: build_route: Contact hop: sip:[EMAIL
PROTECTED]
Jul 3 17:21:47 DEBUG[896] chan_sip.c: * Detected inband DTMF '1'
Jul 3 17:22:18 DEBUG[896] chan_sip.c: Stopping retransmission on '[EMAIL
PROTECTED]' of Request 102: Match Found
Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is '(null)'
Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is '(null)'
Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is '205'
Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is 'default'
Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is 'SIP/3001-008d8ce0'
Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is '(null)'
Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is 'Hangup'
Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is '(null)'
Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is '2008-07-03 17:21:27'
Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is '2008-07-03 17:21:27'
Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is '2008-07-03 17:22:23'
Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is '56'
Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is '56'
Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is 'ANSWERED'
Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is 'DOCUMENTATION'
Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is '(null)'
Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is '1215085887.0'
Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is '(null)'
Jul 3 17:22:23 DEBUG[896] chan_sip.c: update_call_counter(205) - decrement
call limit counter
Jul 3 17:22:23 NOTICE[896] pbx_spool.c: Call completed to SIP/3001
Jul 3 17:22:23 DEBUG[896] chan_sip.c: Stopping retransmission on '[EMAIL
PROTECTED]' of Request 103: Match Found
Jul 3 17:22:24 DEBUG[896] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]'
Jul 3 17:23:57 DEBUG[896] chan_sip.c: Stopping retransmission on '[EMAIL
PROTECTED]' of Request 102: Match Found
Jul 3 17:24:09 DEBUG[896] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]'
Jul 3 17:25:47 DEBUG[896] chan_sip.c: Stopping retransmission on '[EMAIL
PROTECTED]' of Request 102: Match Found
Jul 3 17:25:54 DEBUG[896] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]'
It says detected inband dtmf 1 but says nothing about 9.
Am I doing anything wrong in the extension.conf.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benjamin Jacob
Sent: Thursday, July 03, 2008 5:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] (no subject)
Use SendDTMF.
--- On Thu, 7/3/08, Neha Punia [EMAIL PROTECTED] wrote:
From: Neha Punia [EMAIL PROTECTED]
Subject: [asterisk-users] (no subject)
To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com
Date: Thursday, July 3, 2008, 10:29 AM
Hi
I m making a call from one asterisk server to an asterisk
client
The call gets completed but I want it to send dtmf signals
The dialplan I have made for this is like
exten = 205,1,Answer
exten = 205,n,Wait(15)
exten = 205,n,Playback(dtmf-1)
exten = 205,n,Wait(20)
but it does not send any dtmf signal
where is the problem??
CAUTION - Disclaimer *
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INFORMATION intended solely
for the use of the addressee(s). If you are not the
intended recipient, please
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message. Further, you are not
to copy, disclose, or distribute this e-mail or its
contents to any other person and
any such actions are unlawful. This e-mail may contain
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not liable for any damage
you may sustain as a result of any virus in this e-mail.
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Infosys reserves
Re: [asterisk-users] (no subject)
The number one skill for setting up asterisk is learn how to communicate since it's a communication application :P As for your problem looks like you are trying to use the wrong span for dial out. On Thu, Jul 3, 2008 at 8:50 AM, Bikrish Amatya [EMAIL PROTECTED] wrote: Hello everybody I have configures asterisk server and i am using TE220P digium card. Here is the content of the /etc/zaptel.conf file ### span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 span=2,2,0,ccs,hdb3 bchan=32-46,48-62 dchan=47 loadzone= in defaultzone = in the content of /etc/asterisk/zapata.conf is as follow [channels] context=incoming switchtype=national ;pridialplan=national usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes rxgain=0.0 txgain=0.0 immediate=no callprogress=no callerid=asreceived group=1 channel=1-15,17-31 # output of zttool is as follow #9474; Alarms Span #9474; #9474; RED T2XXP (PCI) Card 0 Span 1 #9474; OK T2XXP (PCI) Card 0 Span 2 #9474; Output of cat /prox/zaptel/1 is as follow Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 HDB3/CCS RED 1 TE2/0/1/1 Clear (In use) RED 2 TE2/0/1/2 Clear (In use) RED 3 TE2/0/1/3 Clear (In use) RED 4 TE2/0/1/4 Clear (In use) RED 5 TE2/0/1/5 Clear (In use) RED 6 TE2/0/1/6 Clear (In use) RED 7 TE2/0/1/7 Clear (In use) RED 8 TE2/0/1/8 Clear (In use) RED 9 TE2/0/1/9 Clear (In use) RED 10 TE2/0/1/10 Clear (In use) RED 11 TE2/0/1/11 Clear (In use) RED 12 TE2/0/1/12 Clear (In use) RED 13 TE2/0/1/13 Clear (In use) RED 14 TE2/0/1/14 Clear (In use) RED 15 TE2/0/1/15 Clear (In use) RED 16 TE2/0/1/16 HDLCFCS (In use) RED 17 TE2/0/1/17 Clear (In use) RED 18 TE2/0/1/18 Clear (In use) RED 19 TE2/0/1/19 Clear (In use) RED 20 TE2/0/1/20 Clear (In use) RED 21 TE2/0/1/21 Clear (In use) RED 22 TE2/0/1/22 Clear (In use) RED 23 TE2/0/1/23 Clear (In use) RED 24 TE2/0/1/24 Clear (In use) RED 25 TE2/0/1/25 Clear (In use) RED 26 TE2/0/1/26 Clear (In use) RED 27 TE2/0/1/27 Clear (In use) RED 28 TE2/0/1/28 Clear (In use) RED 29 TE2/0/1/29 Clear (In use) RED 30 TE2/0/1/30 Clear (In use) RED 31 TE2/0/1/31 Clear (In use) RED I am new to asterisk and googled around , configured the asterisk server. Now when i make a call from outside , it give me busy tone.. and when i call from softphone .. it shows me as show below -- Executing [EMAIL PROTECTED]:1] Dial(SIP/bikrish-09b21980, Zap/g1/600833) in new stack [Jul 3 19:14:34] WARNING[6018]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/bikrish-09b21980' status is 'CONGESTION' I am not able to figure out the problem. Any kind of help would be appericiated. Thanking you bikrish ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
On Thu, 3 Jul 2008, Alex Balashov wrote: C F wrote: The number one skill for setting up asterisk is learn how to communicate since it's a communication application :P Oh, if only more newbie posters on this list would heed that advice. ) How about rejecting emails that don't have a subject? ) How about rejecting top posted replies? ) How about rejecting posts to -dev until the poster's account is more than a couple of days old? Or until they've earned a couple of karma points? Or a challenge/response confirming this post is about changing the C source code? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
On Thu, 3 Jul 2008, Alex Balashov wrote: Steve Edwards wrote: On Thu, 3 Jul 2008, Alex Balashov wrote: C F wrote: The number one skill for setting up asterisk is learn how to communicate since it's a communication application :P Oh, if only more newbie posters on this list would heed that advice. ) How about rejecting emails that don't have a subject? ) How about rejecting top posted replies? ) How about rejecting posts to -dev until the poster's account is more than a couple of days old? Or until they've earned a couple of karma points? Or a challenge/response confirming this post is about changing the C source code? I would say the main thing that is needed is a grammar and spelling checker, followed by some degree of nominal assessment of conceptual integrity and coherence. The latter may be impossible to implement, but the former would be beneficial. But deciphering posts from our non-English-speaking members is half the challenge/fun :) Seriously though, good for them for trying. I wouldn't. What are you if you speak 3 languages? Trilingual. What are you if you speak 2 languages? Bilingual. What are you if you only speak 1 language? American :) Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
Steve Edwards wrote: On Thu, 3 Jul 2008, Alex Balashov wrote: Steve Edwards wrote: On Thu, 3 Jul 2008, Alex Balashov wrote: C F wrote: The number one skill for setting up asterisk is learn how to communicate since it's a communication application :P Oh, if only more newbie posters on this list would heed that advice. ) How about rejecting emails that don't have a subject? ) How about rejecting top posted replies? ) How about rejecting posts to -dev until the poster's account is more than a couple of days old? Or until they've earned a couple of karma points? Or a challenge/response confirming this post is about changing the C source code? I would say the main thing that is needed is a grammar and spelling checker, followed by some degree of nominal assessment of conceptual integrity and coherence. The latter may be impossible to implement, but the former would be beneficial. But deciphering posts from our non-English-speaking members is half the challenge/fun :) Seriously though, good for them for trying. I wouldn't. What are you if you speak 3 languages? Trilingual. What are you if you speak 2 languages? Bilingual. What are you if you only speak 1 language? American :) I'm trilingual, but English is by far my best language. If I had to write a post on a technical mailing list in one of the other languages, I would certainly take the time to ensure that it sounds reasonably coherent. I cannot fault people for poor/limited English. But there is a difference between someone who tried and someone who didn't, and it is reflected in the overall level of culture that comes across in the substance of their post, the formulation of their thoughts, and so on. Somebody that *both* speaks/writes English poorly -- *and* uses incomprehensible, Philistine gibberish (excuse me, AOLer short-hand) -- deserves what they earn. There seems to be a remarkable coincidence of these two proclivities as often as not. -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
Alex Balashov wrote: Steve Edwards wrote: On Thu, 3 Jul 2008, Alex Balashov wrote: Steve Edwards wrote: On Thu, 3 Jul 2008, Alex Balashov wrote: C F wrote: The number one skill for setting up asterisk is learn how to communicate since it's a communication application :P Oh, if only more newbie posters on this list would heed that advice. ) How about rejecting emails that don't have a subject? ) How about rejecting top posted replies? ) How about rejecting posts to -dev until the poster's account is more than a couple of days old? Or until they've earned a couple of karma points? Or a challenge/response confirming this post is about changing the C source code? I would say the main thing that is needed is a grammar and spelling checker, followed by some degree of nominal assessment of conceptual integrity and coherence. The latter may be impossible to implement, but the former would be beneficial. But deciphering posts from our non-English-speaking members is half the challenge/fun :) Seriously though, good for them for trying. I wouldn't. What are you if you speak 3 languages? Trilingual. What are you if you speak 2 languages? Bilingual. What are you if you only speak 1 language? American :) I'm trilingual, but English is by far my best language. If I had to write a post on a technical mailing list in one of the other languages, I would certainly take the time to ensure that it sounds reasonably coherent. I cannot fault people for poor/limited English. But there is a difference between someone who tried and someone who didn't, and it is reflected in the overall level of culture that comes across in the substance of their post, the formulation of their thoughts, and so on. Somebody that *both* speaks/writes English poorly -- *and* uses incomprehensible, Philistine gibberish (excuse me, AOLer short-hand) -- deserves what they earn. There seems to be a remarkable coincidence of these two proclivities as often as not. -- Alex Bilingual, Trilingual, -lingual does not necessarily include English as one of the languages. It is for some a great effort just trying to write in English, never mind the effort of knowing colloquialism, etc. So not being fluent, not being able to be as coherent as a native English speaker would, does not make me or someone else eligible for an answer. No wonder so many think that monolingual people with English as their only language are arrogant Yes, diatribes and flames are accepted //Peter ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
On Fri, 4 Jul 2008, Peter Lindquist wrote: Steve Edwards wrote: But deciphering posts from our non-English-speaking members is half the challenge/fun :) Seriously though, good for them for trying. I wouldn't. What are you if you speak 3 languages? Trilingual. What are you if you speak 2 languages? Bilingual. What are you if you only speak 1 language? American :) Bilingual, Trilingual, -lingual does not necessarily include English as one of the languages. It is for some a great effort just trying to write in English, never mind the effort of knowing colloquialism, etc. So not being fluent, not being able to be as coherent as a native English speaker would, does not make me or someone else eligible for an answer. No wonder so many think that monolingual people with English as their only language are arrogant Yes, diatribes and flames are accepted Boy, did you miss the mark. I am a monolingual American. I was giving non-English-speakers props for trying and poking fun at myself and my countrymen. Lighten up. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
Alex Balashov wrote: ) How about rejecting emails that don't have a subject? That is an excellent idea. If a person doesn't have enough clue to use a subject, then we're really just feeding the beast when we indulge the question with an answer. And the archived version of that question/answer are pretty useless, too. Thx. b. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
the subject of this thread has been on this list way too many times just search the archives. On 5/23/08, Joseph L. Casale [EMAIL PROTECTED] wrote: In the setup tutorial @ http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation it states the potential issue regarding setting up UniqueID as the primary key, but doesn't state how to rectify this? What is the proper way to make sure this is done right? Also, has anyone built a simple front end for non technical folk to utilize for accessing the data simply for overview when billing etc is not important (small company)? Thanks! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
http://www.soft-switch.org/unicall/mfcr2/ch02.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
This may be more helpful as far as Asterisk implementation. Sorry I cannot be of more help, I have never dealt with this tech. http://www.voip-info.org/wiki/view/Asterisk+MFC+R2 Thanks, Steve Totaro On Mon, Apr 28, 2008 at 9:06 AM, Arthur [EMAIL PROTECTED] wrote: http://www.soft-switch.org/unicall/mfcr2/ch02.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
Again, a reply to my reply. Note to self: stop hitting send before completing thoughts. Maybe if you ask the telco to turn off the SLA blocking. It may not solve the underlying issue but it may allow you to continue inbound and outbound without service interruption providing it does not drop any active calls as well. Make sure you get a helpful tech on the phone. Many times they will just dismiss you with we cannot do that even though they may be able to. Thanks, Steve Totaro On Mon, Apr 28, 2008 at 9:12 AM, Steve Totaro [EMAIL PROTECTED] wrote: This may be more helpful as far as Asterisk implementation. Sorry I cannot be of more help, I have never dealt with this tech. http://www.voip-info.org/wiki/view/Asterisk+MFC+R2 Thanks, Steve Totaro On Mon, Apr 28, 2008 at 9:06 AM, Arthur [EMAIL PROTECTED] wrote: http://www.soft-switch.org/unicall/mfcr2/ch02.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
Make sure you get a helpful tech on the phone. Many times they will just dismiss you with we cannot do that even though they may be able to. i always say if you pay your bills you should get the support you diserve. every provider is almost always willing to help out his clients if they express their needs with precision. one more thing : nothing compares to having a friend working at the providers company so get yourself one. Again, a reply to my reply. Note to self: stop hitting send before completing thoughts. you shoudl add something like this to your base code .. if finish-email == 'yes': keyboard.hit(enter) else: keyboard.write(text) :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
On Mon, Apr 28, 2008 at 9:32 AM, Arthur [EMAIL PROTECTED] wrote: Make sure you get a helpful tech on the phone. Many times they will just dismiss you with we cannot do that even though they may be able to. i always say if you pay your bills you should get the support you diserve. every provider is almost always willing to help out his clients if they express their needs with precision. one more thing : nothing compares to having a friend working at the providers company so get yourself one. You are preaching to the choir. I have dealt with all the big and many of the small players here in the US. I always say people that do the right thing and work hard will be rewarded but more often than not, they are taken advantage of. This is not Utopia, these guys at the telcos are overworked, work in a monolithic bureaucracy, and many probably hate their jobs. They love to close tickets ASAP since that is how they are evaluated. As soon as I get a good helpful tech, I get their DID and praise the heck out of them (almost to the point of brown nosing) and CC their supervisor (with their permission of course). Normal support channels get me answers like we cannot do that, or we can but it will take about two weeks. Again, a reply to my reply. Note to self: stop hitting send before completing thoughts. you shoudl add something like this to your base code .. if finish-email == 'yes': keyboard.hit(enter) else: keyboard.write(text) :) True, true, but coffee tends to stave off incomplete or incoherent postings. Sometimes I look at posting made at the end of the day or before the caffeine kicks in and they make no sense whatsoever :) Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
On Fri, 2008-02-22 at 10:38 +0530, sandeep wrote:
for example:
dial to a extension(123).if the user didnot pick the call, caller
should get a ivr script(Enter 1 to to dial operator and 2 to go to
voicemail)
If caller press 1 it should dial to the operator,else if he dials 2 it
should go to the voicemail of calle's extension.
It's really pretty easy.
; Call the SIP peer, let the phone ring for 20 seconds
exten = 123,1,Dial(SIP/some_sip_peer,20)
; Play the press-1-or-press-2 prompt, get one digit
; from the caller, and save it to a variable called
; ${option}
exten = 123,n,Read(option,press-1-or-press-2,1)
; If the caller enters 1, send the call to the [some_context] context,
; to the operator extension, priority 1
exten = 123,n,GotoIf($[${option} = 1]?some_context,operator,1)
; Otherwise, send the call to voicemail
exten = 123,n,VoiceMail([EMAIL PROTECTED])
I haven't actually taken the time to test this in my own dialplan, but
it should work. Obviously you'll want to change the name of the SIP
peer you're dialing, as well as the location of the operator extension.
--
Jared Smith
Community Relations Manager
Digium, Inc.
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Re: [asterisk-users] (no subject)
vi /etc/asterisk/extensions.conf On Fri, Feb 22, 2008 at 12:08 AM, sandeep [EMAIL PROTECTED] wrote: hi, how to write a advanced dial plan for example: dial to a extension(123).if the user didnot pick the call, caller should get a ivr script(Enter 1 to to dial operator and 2 to go to voicemail) If caller press 1 it should dial to the operator,else if he dials 2 it should go to the voicemail of calle's extension. thanks sandeep. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
Check your extensions.conf On Jan 1, 2008 11:33 AM, lists65 [EMAIL PROTECTED] wrote: ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
Andrew Joakimsen wrote: Check your extensions.conf Hahahahaha! Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
[EMAIL PROTECTED] wrote: Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to our customer. The only experience we've had is a very frustrating one trying to load the IP software on a Cisco 7970G and so we assume that if we have to go through that for all 80 phones, we'll probably commit suicide :) Thanks We have used Cisco and Aastra, can't comment on Polycom or Snom. I cannot recommend Cisco, good sound quality but that's it. Ridiculously overpriced, too few usable features, incredibly awkward to manage. Aastra have good sound quality, reasonable price, configs are plain text and not to hard to work with. We have the 9133i as our basic phone and 480i in the Call Centre for the soft buttons. Both can be fed from the same config templates. We used to use Grandstream but quality and support issues have driven us away. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
We agree with Drew and no longer use Grandstream. We have used a few Polycom, (best voice quality, hardest to configure). I have heard good things about Snom but never used them. We standardized on Aastra. Good build, sound quality, and feature set. Easy to configure or upgrade and good pricing. If you try Snom please share your thoughts. At present we are sticking with Aastra due to good results and user feedback. Jim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Drew Gibson Sent: Wednesday, October 31, 2007 11:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] (no subject) [EMAIL PROTECTED] wrote: Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to our customer. The only experience we've had is a very frustrating one trying to load the IP software on a Cisco 7970G and so we assume that if we have to go through that for all 80 phones, we'll probably commit suicide :) Thanks We have used Cisco and Aastra, can't comment on Polycom or Snom. I cannot recommend Cisco, good sound quality but that's it. Ridiculously overpriced, too few usable features, incredibly awkward to manage. Aastra have good sound quality, reasonable price, configs are plain text and not to hard to work with. We have the 9133i as our basic phone and 480i in the Call Centre for the soft buttons. Both can be fed from the same config templates. We used to use Grandstream but quality and support issues have driven us away. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
What is the issue with the Grandstream? We are getting tired of Cisco issues, so we have started looking at Grandstream and they seem to be pretty good. The Polycom work well, but they seem to die after about a year or so. We bought 20 of them about 2 years ago and 7 of them have died or had buttons stop working so we had to replace them. I haven't had a single Cisco do that and we have probably 100 of them. Jim Houser wrote: We agree with Drew and no longer use Grandstream. We have used a few Polycom, (best voice quality, hardest to configure). I have heard good things about Snom but never used them. We standardized on Aastra. Good build, sound quality, and feature set. Easy to configure or upgrade and good pricing. If you try Snom please share your thoughts. At present we are sticking with Aastra due to good results and user feedback. Jim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Drew Gibson Sent: Wednesday, October 31, 2007 11:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] (no subject) [EMAIL PROTECTED] wrote: Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to our customer. The only experience we've had is a very frustrating one trying to load the IP software on a Cisco 7970G and so we assume that if we have to go through that for all 80 phones, we'll probably commit suicide :) Thanks We have used Cisco and Aastra, can't comment on Polycom or Snom. I cannot recommend Cisco, good sound quality but that's it. Ridiculously overpriced, too few usable features, incredibly awkward to manage. Aastra have good sound quality, reasonable price, configs are plain text and not to hard to work with. We have the 9133i as our basic phone and 480i in the Call Centre for the soft buttons. Both can be fed from the same config templates. We used to use Grandstream but quality and support issues have driven us away. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
We have used the Grandstream GPX2000, HT503 and GXW4104 gateways. Quality is in all cases are on the lower end. The quality I refer to is buggy software and poor call quality. I have been involved with Telecom since the early 80s and dealt with a lot of phone systems. The Grandstream phones just plain feel cheap. Real Walmart quality, not professional business class equipment. The phone functioned ok and was super easy to setup but complaints of echo and poor volume levels were common. They may be better as we have not used them in over 6 months. We have recently used their gateways due to good pricing and their economics fit our solution base well but ran into issues with them. I believe their gateways will get improved as both are new and on early firmware releases. However, we got upset with poor support. Either no call back at all or a useless email a day later with little to no information to help solve our issue. In Grandstream's defense it may be we are just too small to matter and that's ok. We prefer to go elsewhere and deliver product that when the average user picks it up to talk on it they say this is quality stuff. Asterisk is as talented as the firm that programs it BUT the phone is crucial in the end user's system satisfaction. Regardless of what you put in the back room the phone IS the device that sets the impression to your client if you are delivering a quality solution. We would do Cisco because it is high quality but we don't care to fight with the configuration or licensing issues. We would do Polycom, and probably will, but have not had the time to jump to through the hoops needed to acquire good enough pricing to make money selling them. We feel Aastra is a good compromise in delivering quality product to make the customer happy with their decision while still making us to make some sort of small profit for our time. It's solid and provides a quality feel and function. This said, Grandstream is not junk and this is not meant to be a Grandstream rant. I would like to apologize if I jumped in too quick sounding that way. Grandstream is just the lower end of quality and should be deployed in applications where the client is willing to accept that. That's not our marketplace. If you want easy to configure, low cost, slam dunk Asterisk deployments then Grandstream works. But the end result will not be as good if you build a system with Cisco, Polycom, Snom, or Aastra. We've even tested Avaya 46XX phones on Asterisk. They sound GREAT! Probably one of the best. We just can't get Asterisk to light the messaging waiting light on the phone. Arrggg! You need to decide what your marketplace offering is and what your clients are willing to accept. If call quality is the most important then our testing shows nobody beats Polycom or Avaya. Someday we are going to beat the Avaya message waiting light issue. If quality of deskset feel is the most important factor them Avaya and Cisco stand out as best. We will not put configuration into a factor simply because the customer uses the tool we are expected to configure it to their needs. We won't sell them any device based on it being easier for us to configure. I would like to hear what people say about Snom as their sets look very nice. Sorry for the novel, all I really wanted to express is Grandstream is cheap, look before you jump. Good luck on your decision... Jim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peder @ NetworkOblivion Sent: Wednesday, October 31, 2007 11:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] (no subject) What is the issue with the Grandstream? We are getting tired of Cisco issues, so we have started looking at Grandstream and they seem to be pretty good. The Polycom work well, but they seem to die after about a year or so. We bought 20 of them about 2 years ago and 7 of them have died or had buttons stop working so we had to replace them. I haven't had a single Cisco do that and we have probably 100 of them. Jim Houser wrote: We agree with Drew and no longer use Grandstream. We have used a few Polycom, (best voice quality, hardest to configure). I have heard good things about Snom but never used them. We standardized on Aastra. Good build, sound quality, and feature set. Easy to configure or upgrade and good pricing. If you try Snom please share your thoughts. At present we are sticking with Aastra due to good results and user feedback. Jim [EMAIL PROTECTED] wrote: Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard
Re: [asterisk-users] (no subject)
Honestly, Its my opinion that the Aastra phones are very lacking in the firmware department. If they could get that sorted out I wouldn't mind using them. But for now there are too many NAT issues mostly caused because they use an OLD version of Broadcom CallCtrl. Why they use an ancient version is beyond me but the phones dont even have a NAT keepalive option. They promise updates to their firmware but then they only fix minor bugs. Grandstream are ok. But as others have said their support is very lacking. I've had products of theirs behave very oddly like operate and refuse to apply any settings no matter what and not allow a factory reset... paperweight. I'd personally use Polycom in the situations where there's no NAT and the Linksys SPA-phones where you do have NAT. On 10/29/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to our customer. The only experience we've had is a very frustrating one trying to load the IP software on a Cisco 7970G and so we assume that if we have to go through that for all 80 phones, we'll probably commit suicide :) Thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
We have Cisco 9760 for executives and Aastra 9112i for everybody else. We started with Grandstream, don't remember the model, cost around $80 USD but it had bad audio quality and echo problems (running asterisk 1.09). The quality of construction felt poor, like a toy phone. We replaced them with the Aastra for double the cost and the quality improved dramatically. Audio quality was much better and echo problems all but eliminated. This phone also feels more solid. There are a few areas that are not perfect; the speaker phone is good not excellent and we have had to replace a couple of phones because they have stopped working. Over all I would say not bad for the price especially if they are for general use. We had to upgrade from the Aastra phones for our executives because they needed very good audio for both handset and speaker phone. We are using Cisco 9760's for them and have had no problems with quality. Plus they have a very solid feel. My question to the list is: As I need to add phones I am considering buying used Cisco 9760's. Is there any difference with the 9760G? I have heard that the 9761's have even better audio quality. Our main requirement is audio quality, our users do not need a lot of features on their phones. Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peder @ NetworkOblivion Sent: Wednesday, October 31, 2007 12:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] (no subject) What is the issue with the Grandstream? We are getting tired of Cisco issues, so we have started looking at Grandstream and they seem to be pretty good. The Polycom work well, but they seem to die after about a year or so. We bought 20 of them about 2 years ago and 7 of them have died or had buttons stop working so we had to replace them. I haven't had a single Cisco do that and we have probably 100 of them. Jim Houser wrote: We agree with Drew and no longer use Grandstream. We have used a few Polycom, (best voice quality, hardest to configure). I have heard good things about Snom but never used them. We standardized on Aastra. Good build, sound quality, and feature set. Easy to configure or upgrade and good pricing. If you try Snom please share your thoughts. At present we are sticking with Aastra due to good results and user feedback. Jim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Drew Gibson Sent: Wednesday, October 31, 2007 11:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] (no subject) [EMAIL PROTECTED] wrote: Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to our customer. The only experience we've had is a very frustrating one trying to load the IP software on a Cisco 7970G and so we assume that if we have to go through that for all 80 phones, we'll probably commit suicide :) Thanks We have used Cisco and Aastra, can't comment on Polycom or Snom. I cannot recommend Cisco, good sound quality but that's it. Ridiculously overpriced, too few usable features, incredibly awkward to manage. Aastra have good sound quality, reasonable price, configs are plain text and not to hard to work with. We have the 9133i as our basic phone and 480i in the Call Centre for the soft buttons. Both can be fed from the same config templates. We used to use Grandstream but quality and support issues have driven us away. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
What is the use case? Linksys, Polycom, Snom, and Aastra all have their strengths and weaknesses. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, October 29, 2007 10:42 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] (no subject) Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to our customer. The only experience we've had is a very frustrating one trying to load the IP software on a Cisco 7970G and so we assume that if we have to go through that for all 80 phones, we'll probably commit suicide :) Thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
Stay away from Cisco they just don't work for the price, if it would be in the price range of a Grandstream phone I would tell you go for it, but at the current price its just not worth it. Aastra, Polycom or linksys all work for me. Never tried Snom before. On 10/29/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to our customer. The only experience we've had is a very frustrating one trying to load the IP software on a Cisco 7970G and so we assume that if we have to go through that for all 80 phones, we'll probably commit suicide :) Thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
I've had experience with Linksys and Polycom. Either one is easy enough to provision. Took me a while to understand how to provision Polycom. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, 30 October 2007 3:42 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] (no subject) Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to our customer. The only experience we've had is a very frustrating one trying to load the IP software on a Cisco 7970G and so we assume that if we have to go through that for all 80 phones, we'll probably commit suicide :) Thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
Motherboard with SATA RAID1 support That's a mulit-port SATA controller with RAID in the driver (software). 256 MB RAM Use a little more RAM. digium PRI/E1 card Is there any reason you aren't using Sangoma cards? 1. If I use Software RAID, what would be the impact to my deployment? ( problems that I have to face with regard to the call flow ) None. 2. If I use Hardware based RAID 1, what would be the impact to the system? A PCI slot. 3. According to your practical experiance what is the ideal solution among both options? Software RAID works fine. -- Thanks Regards, Vidura Senadeera, Network Engineer, Debug Solutions Sri Lanka. Tel - +94114520036 Mobile - +9466596 Web - www.debug.lk ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] (no subject)
Hi Guy,. you should at least put a subject any way follow this link http://nerdvittles.com/index.php?p=134 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Mon, 11 Jun 2007 18:36:54 +0530 Subject: [asterisk-users] (no subject) Hi, please help me in developing and reading Text through IVR application using asterisk. can any one help me at highlevel on this, other than using SPANDSP application. Regards K.Rajesh. _ Tried the new MSN Messenger? It’s cool! Download now. http://messenger.msn.com/Download/Default.aspx?mkt=en-in ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ With Windows Live Hotmail, you can personalize your inbox with your favorite color. www.windowslive-hotmail.com/learnmore/personalize.html?locale=en-usocid=TXT_TAGLM_HMWL_reten_addcolor_0607___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] (no subject)
That made all the difference! Thanks again! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Ruggles Sent: Wednesday, May 30, 2007 6:26 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] (no subject) Thanks; I have made the change and I will try it tomorrow! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cristian N. Bradiceanu Sent: Wednesday, May 30, 2007 4:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] (no subject) Hi, Please take a look at http://www.voip-info.org/wiki/index.php?page=Asterisk+v1.2+upgrade+to+v1.4+g otchas iax.conf The new threading model is great, but the default of 10 threads is way too low. Symptoms include total loss of audio until the channel is hung up. in general section, add: iaxthreadcount = 200 in general section, add: iaxmaxthreadcount = 1000 Hope this helps. Regards, Cristian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
Hi,
Please take a look at
http://www.voip-info.org/wiki/index.php?page=Asterisk+v1.2+upgrade+to+v1.4+gotchas
iax.conf The new threading model is great, but the default of 10 threads is
way too low. Symptoms include total loss of audio until the channel is hung
up.
- in general section, add: iaxthreadcount = 200
- in general section, add: iaxmaxthreadcount = 1000
Hope this helps.
Regards,
Cristian
On 5/30/07, David Ruggles [EMAIL PROTECTED] wrote:
Need some help with IAX trunking.
I've got six systems:
AsteriskM (main)
___|
| || | |
Asterisk1 Asterisk2 Asterisk3 Asterisk4 Asterisk5
AsteriskM has two Sangoma A102 2 Port T1 cards in it, the other Asterisk
boxes are using ztdummy for timing, they are all using IAX trunking.
My calls come in over Sip or Zap to asteriskm and are routed to one of the
asteriskN servers based on load. The routing is done by a small AGI script
that gets the current load from a monitoring machine and then changes the
priority. Dialplan snippet:
--- Snippet ---
exten = _X.,1,AGI(manager.agi)
exten = _X.,100,Dial(IAX2/asterisk1/${EXTEN})
exten = _X.,200,Dial(IAX2/asterisk2/${EXTEN})
exten = _X.,300,Dial(IAX2/asterisk3/${EXTEN})
exten = _X.,400,Dial(IAX2/asterisk4/${EXTEN})
exten = _X.,500,Dial(IAX2/asterisk5/${EXTEN})
--- Snippet ---
This works fine for a few calls. I'm using the SIPp package to generate a
10-25 simultaneous call load. Every once in a while I starting seeing
loads
of error messages on AsteriskM's console:
chan_iax2.c:7054 socket_process: Peer 'asterisk4' is now UNREACHABLE!
Time:
2
chan_iax2.c:6216 socket_read: Out of idle IAX2 threads for I/O, pausing!
chan_iax2.c:909 __schedule_action: Out of idle IAX2 threads for
scheduling!
chan_iax2.c:7054 socket_process: Peer 'asterisk4' is now REACHABLE! Time:
134
chan_iax2.c:6216 socket_read: Out of idle IAX2 threads for I/O, pausing!
That is just a small example, I may have 50-100 of these type of messages
scroll very quickly. If I give the system a minute everything goes back to
normal.
I would like some one who is very knowledgeable about IAX to assist me
with
this problem. If someone knows a lot about IAX optimization and is willing
to work with me I would be willing to pay for their time.
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200 [EMAIL PROTECTED]
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RE: [asterisk-users] (no subject)
Thanks; I have made the change and I will try it tomorrow! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cristian N. Bradiceanu Sent: Wednesday, May 30, 2007 4:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] (no subject) Hi, Please take a look at http://www.voip-info.org/wiki/index.php?page=Asterisk+v1.2+upgrade+to+v1.4+g otchas iax.conf The new threading model is great, but the default of 10 threads is way too low. Symptoms include total loss of audio until the channel is hung up. * in general section, add: iaxthreadcount = 200 * in general section, add: iaxmaxthreadcount = 1000 Hope this helps. Regards, Cristian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
You seem to have misplaced your message/comment/question. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
Todd- Asterisk wrote: Hello everyone! I'm planning on setting up a new system shortly and can't pick the right card... We will have 2 or 3 lines coming in and 7 extensions (GXP2k's). Should I just get 2 or 3 X100P cards? Or do I need the Sangoma A20200 or even the A20200D (Echo cancelation)... I was thinking I'd use a Dell 2.0 GHz machine as the server... If anyone has suggestions as to the benifits/problems of each card choice, I'd love to hear it. thanks Todd In my humble opinion, X100P's are only good for one line (and barely that). They don't work as well as the TDM400s do, and having more than one X100 card in a system is an unnecessary bombardment of interrupts. For a 2-3 line setup I would strongly suggest looking at a TDM400 or the Sangoma A200. I have used both and have been happy with both. I use a TDM400 at home and have managed to remove almost all echo with the use of fxotune and adjusting the gains. I'm using a Sangoma A200 with the on-board echo canceler for a phone system at work and have been very happy with it. The only complaint of echo on this system is on an occasional incoming call and only for the first second or two. If money is tight and you are willing to tune echo out of your system by hand, use the TDM400. If you are willing to spend the cash and don't want to have to deal with constant tweaking to remove echo, get the A200d (and make sure you download the latest drivers from sangoma). -Dave ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
At 05:23 AM 12/14/2006, you wrote: Should I just get 2 or 3 X100P cards? Or do I need the Sangoma A20200 or even the A20200D (Echo cancelation)... When I started down this path I choose the TDM04 and have always had occasional echo issues, not bad and not often, but it annoys the wife and one of these days I'll sell the TDM04 and replace it with the A20002D so I have hardware echo cancellation. Someone else a few months back said the same thing about all the small business installations he did because he just didn't want to have complaints and the extra $300 was a small price to pay for peace of mind. All that said, I don't have the Sangoma card yet and have never seen one so I could be blowing smoke! Ira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
I have been using the sangoma A200 with echo cancelation and I have been real happy. - Original Message - From: Todd- Asterisk [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, December 14, 2006 3:23 PM Subject: [asterisk-users] (no subject) Hello everyone! I'm planning on setting up a new system shortly and can't pick the right card... We will have 2 or 3 lines coming in and 7 extensions (GXP2k's). Should I just get 2 or 3 X100P cards? Or do I need the Sangoma A20200 or even the A20200D (Echo cancelation)... I was thinking I'd use a Dell 2.0 GHz machine as the server... If anyone has suggestions as to the benifits/problems of each card choice, I'd love to hear it. thanks Todd ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
You might want to take a look at the new 4 port FXO from Grandstream I haven't had one yet to evaluate but assuming it works it is very price competative and off-loads all the analog (TDM) stuff from your PC Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada I have been using the sangoma A200 with echo cancelation and I have been real happy. - Original Message - From: Todd- Asterisk [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, December 14, 2006 3:23 PM Subject: [asterisk-users] (no subject) Hello everyone! I'm planning on setting up a new system shortly and can't pick the right card... We will have 2 or 3 lines coming in and 7 extensions (GXP2k's). Should I just get 2 or 3 X100P cards? Or do I need the Sangoma A20200 or even the A20200D (Echo cancelation)... I was thinking I'd use a Dell 2.0 GHz machine as the server... If anyone has suggestions as to the benifits/problems of each card choice, I'd love to hear it. thanks Todd ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
We have placed 2 X 4 port cards in a Dell 2950 and it worked well - even when it was recording 50% of the calls. PaulH On Fri, 2006-11-24 at 11:54 +0600, Imran M Yousuf wrote: Dear Users, I am fairly new to Digium and Asterisk. I wanted to know that if I use the Digium product THREE Digium Wildcard TE412Ps’ (Quad E1 Card) how many calls can I handle simultaneously. I want to use the cards with the following Configurations: Intel® Xeon™ 3.00GHz/800MHz, 2M Processor 1GB PC2-3200 DDR2 SDRAM (2x512MB + 2X1GB) memory Integrated Dual Channel Ultra320 SCSI Adapter NC7781 Single Port PCI-X embedded NIC Hot plug drive cage - Ultra3 (6X1) High Speed IDE CD-ROM Drive 72.8GB Pluggable Ultra320 SCSI 15,000 rpm (1) Universal Hard Drive Asterisk Business Edition 3 X TE412P I have a requirement of handling 350 Calls using a single Server and please note the Server will used to transferring the call only. Other Servers will handle gateway Negotiation and Billing. This server will SIMPLY be a Gateway. Please let me know if this configuration too high or too low. If anybody has better solution please let me know that as well. Thank you, waiting eagerly for a response. Imran M Yousuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
Add a subject next time.
Are you behind a firewall where the Asterisk server is located? Have
forward ports 5060 and 1 - 2 UDP to the asterisk server?
On 11/10/06, Stas Khromoy [EMAIL PROTECTED] wrote:
i am sure this came up before
but all my searches are not resulting in anything usefull
trying to setup a grandstream phone
to connect to an asterisk server
now i am outside the network (home)
on my side
settings on the phone seem to be correct
id and password, astersik server ip, port
in pf.conf
# SIP (TCP)
voip_tcp = 5060
# SIP, IAX2, IAX, RTP, MGCP (UDP)
voip_udp = {5060, 4569, 5036, 20001, 2727}
---
on the server side
same thing
plus
voip_users = ip from where i am connecting
--
can't seem to find anything else that should be opened on either side
to allow connection
--
i guess, help ?
--
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--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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Re: [asterisk-users] (no subject)
You might want to repost it with a subject or you miss a lot of people seeing or opening it up. -- Original message -- From: "Scott Pinhorne" [EMAIL PROTECTED] Hi All I would greatly appreciate some advice or some direction as to where to go next. I have a provider passing me incoming calls via my Session Border Controller. I am able to pass them calls fine but coming in fails with a 407 Authentication Fail error. In my sip.conf I have an entry for the provider but am not asking for a user/pass so I would expect the calls to come in and then pass to the context specified in extensions.conf: [iplcr-gw] type=peer host=xx.xx.xx.xx nat=no dtmfmode=inband context=from-iplcr insecure=invite canreinvite=yes disallow=all allow=ulaw,alaw I have tried different insecure= methods but am still getting the same error. Does anyone know what else could be causing the error or suggest some other things I should try? Many Thanks Scott ---BeginMessage--- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End Message--- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] (no subject)
I am going to reply inline as you asked many questions I have two questions. Sure, you do!! First I am running a t400p with three fxo ports signalling fxs (inbound CO lines). I have six polycom 501's. The problem is the amount of time the call setup takes. I have done this with Mitel phones before with a t-1 and had the same problem. My customer always complains about the call setup time. Am I doing something wrong or is this how it is. It takes up to five seconds to pickup or start ringing the CO. I would be happy to supply fake ringback if anyone knows how to do that. You can add the r command to your Dial string to fake ringback However the CLI is your friend in this case. How long does it take after dialing on the polycom for the console to reflect it is dialing? If it is right away, you may have a problem with your CO lines, and how many digits it is expecting before placing the call. Second Problem is SIP Polycom phone line programming, I have read many contradicting things. How should it be provisioned to allow multiple incoming calls. How many lines,calls per line and the rest of the bull, Iknow loaded question. I am using kewl start on those three lines by the way I set 6 lines per key on my Polycoms that works for 501 and 601. Calls just keep coming IN!!! . ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
[EMAIL PROTECTED] wrote: Hi, Looking for good rates for UK Landline Mobile. Plus Saudi Arabia, UAE, India Pakistan. This is a -biz question, not -users. Also, do you realize how bad it makes you look that you can't even bother to put a subject on your mail? B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (no subject)
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
The only issues you could potentially run into is if all the modules
are FXS and they all needed to ring simultaneously... your power
supply may not be suited to handle to voltage requirements.
Sean
Ninneman, Tj wrote:
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Hey everybody,
Is it alright to run two TDM400s on the same machine? If it is,
how would one differentiate between the channels on each card? So,
if I?m running strait FXS and my first card is fxsks 1-4, would the
second be fxsks 5-8? Would there be any interrupt problems?
Any help would be great!
Thanks!
Tj
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RE: [Asterisk-Users] (no subject)
Hi Tj, yes, you can run two TDM400s (or more) on the same cpu, and the channels are 1 to 4, and 5 to 8. (also, you can set one or more groups for outgoing calls). Interrupts are the main issue. As far as possible avoids that the cards share interruptions. cheers Fabio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Ninneman, Tj Enviado el: Miercoles, 28 de Junio de 2006 12:54 p.m. Para: asterisk-users@lists.digium.com Asunto: [Asterisk-Users] (no subject) Hey everybody, Is it alright to run two TDM400s on the same machine? If it is, how would one differentiate between the channels on each card? So, if I'm running strait FXS and my first card is fxsks 1-4, would the second be fxsks 5-8? Would there be any interrupt problems? Any help would be great! Thanks! Tj ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (no subject)
Sure, but if one needs that many, much better off to use the Sangoma A200 No MB problems and up to 24 channels. John Novack Fabio wrote: Hi Tj, yes, you can run two TDM400s (or more) on the same cpu, and the channels are 1 to 4, and 5 to 8. (also, you can set one or more groups for outgoing calls). Interrupts are the main issue. As far as possible avoids that the cards share interruptions. cheers Fabio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Ninneman, Tj Enviado el: Miercoles, 28 de Junio de 2006 12:54 p.m. Para: asterisk-users@lists.digium.com Asunto: [Asterisk-Users] (no subject) Hey everybody, Is it alright to run two TDM400s on the same machine? If it is, how would one differentiate between the channels on each card? So, if I'm running strait FXS and my first card is fxsks 1-4, would the second be fxsks 5-8? Would there be any interrupt problems? Any help would be great! Thanks! Tj ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (no subject)
[EMAIL PROTECTED] could be a better start for beginners (but beware, the installation CD will format your HD without asking). http://asteriskathome.sourceforge.net/ On Tue, 2006-04-25 at 10:47 +0800, rommel malana wrote: Goodday, I'm an opensource fanatic and I have already installed asterisk in my mandriva linux. Actually, I'm also planning to install the asterisk management portal for GUI of asterisk. If anyone could help me guide in installing this. Thanks a mill for the help.. -Rommel- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (no subject)
--- rommel malana [EMAIL PROTECTED] wrote: Goodday, I'm an opensource fanatic and I have already installed asterisk in my mandriva linux. Actually, I'm also planning to install the asterisk management portal for GUI of asterisk. If anyone could help me guide in installing this. Thanks a mill for the help.. -Rommel- Rommel, You should read the book Asterisk: The future of telephony (I believe is the name). There is a PDF of it available online. Do a google search and you should find it. Dovid __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (no subject)
Please make sure to write a subject line. Thank You On 4/24/06, rommel malana [EMAIL PROTECTED] wrote: Goodday, I'm an opensource fanatic and I have already installed asterisk in my mandriva linux. Actually, I'm also planning to install the asterisk management portal for GUI of asterisk. If anyone could help me guide in installing this. Thanks a mill for the help.. -Rommel- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] (no subject)
Hi Im also new but you should know very well all the interfaces you are going to connect the sistem, the number of users you'll have (hardware requeriments), know a lot about the soft/hardphones you'll use and download the asterisk handbook or the big one (i don't remember the name) Good luck Jose Manuel Cortes David X Semestre Ingenieria Electronica PONTIFICIA UNIVERSIDAD JAVERIANA De: [EMAIL PROTECTED] en nombre de Savvas Gavriel Enviado el: Mié 15/03/2006 15:12 Para: asterisk-users@lists.digium.com Asunto: [Asterisk-Users] (no subject) Hi, to all, i am new in the list and i am interest to deploy a sistem with asterisk i have a PC with a Suse Linux 8.2 and also i have Dialogic VFX card with 4 analog port. From where a can get Dialogic Driver for linux. From ware a mast beging to resolve the problem the project to implement VoIP Gateway. Savvas. _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (no subject)
AFIAK, they can't - we would like to do the same thing, but it's not possible with patching the source. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On 10-Mar-06, at 7:56 PM, btb wrote: can the default voicemail folders (old, work, friends, etc.) be changed? for example, i'd like to configure asterisk so that there are only folders called friends and old. thanks -ben ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (no subject)
See http://www.iaxtel.com/setup.html 2005/12/2, P.G.C.K. Nirukshitha [EMAIL PROTECTED]: Dear Sir I have configured two asterisk Boxes.Then I need to communicate these asterisk boxes via the IAX.It is better if you can help me to configure two boxes to communicate via asterisk. Thanks Nirukshitha Gamage -- This e-mail and any attachments are intended for the above named recipient(s) only and may be privileged. This message and any attachments has been scanned for viruses and dangerous content by ITABS Lanka Mail Scanner, and is believed to be clean. Although measures have been taken to ensure that this e-mail and attachments are free from any virus we advise that, in keeping with good computing practice, the recipient should ensure they are actually virus free. Please note that this message has been sent over public networks which may not be a 100% secure communications medium and ITABS Lanka cannot be held responsible for its integrity. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Giovanni Miano ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (no subject)
See http://www.iaxtel.com/setup.html 2005/12/2, Lakmal [EMAIL PROTECTED]: Hi all, I have configured two asterisk Boxes.Then I need to communicate these asterisk boxes via the IAX.It is better if you can help me to configure two boxes to communicate via asterisk Thanks, Ishanka. - Original Message - From: Branko Samardzic [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, December 02, 2005 10:43 AM Subject: [Asterisk-Users] IAX trunking frequency parameter works only oninitiator side Hi, I am experimenting with trunkfreq parameter. When it is 20ms I can see both parties in IAX session sending IAX frames every 20ms. When I modify this parameter to 40ms then I can see that only server that initiated IAX connection works properly (i.e. sends IAX frames every 40ms while other side still sends IAX frames at 20ms per frame rate). I disabled jitter buffers on both sides and I use speex codec. Here is tcp dump of IAX traffic: 23:26:45.972072 IP SERVER_A.62142 SERVER_B.4569: UDP, length 58 23:26:45.976295 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:45.996264 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.006742 IP SERVER_A.62142 SERVER_B.4569: UDP, length 58 23:26:46.016270 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.036254 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.047891 IP SERVER_A.62142 SERVER_B.4569: UDP, length 58 23:26:46.056248 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.076286 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.091255 IP SERVER_A.62142 SERVER_B.4569: UDP, length 58 23:26:46.096262 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.116243 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.127494 IP SERVER_A.62142 SERVER_B.4569: UDP, length 58 23:26:46.136242 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 SERVER_A initiates connection while SERVER_B answers. SERVER_A iax.conf file === [SERVER_B] disallow=all allow=speex jitterbuffer=no dropcount=2 maxjitterbuffer=200 maxexcessbuffer=100 minexcessbuffer=60 jittershrinkrate=1 trunkfreq=40; How frequently to send trunk msgs (in ms) context = foo secret=zYX9VUt auth=md5 type=friend host=SERVER_B_IP_ADDRESS trunk=yes SERVER_B iax.conf file === [SERVER_B] disallow=all allow=speex jitterbuffer=no dropcount=2 maxjitterbuffer=200 maxexcessbuffer=100 minexcessbuffer=60 jittershrinkrate=1 trunkfreq=40; How frequently to send trunk msgs (in ms) context = default secret=zYX9V auth=md5 type=friend host=SERVER_B_IP_ADDRESS trunk=yes Any idea as to why trunking frequency is not symmetrical? Any help is appreciated ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This e-mail and any attachments are intended for the above named recipient(s) only and may be privileged. This message and any attachments has been scanned for viruses and dangerous content by ITABS Lanka Mail Scanner, and is believed to be clean. Although measures have been taken to ensure that this e-mail and attachments are free from any virus we advise that, in keeping with good computing practice, the recipient should ensure they are actually virus free. Please note that this message has been sent over public networks which may not be a 100% secure communications medium and ITABS Lanka cannot be held responsible for its integrity. -- This e-mail and any attachments are intended for the above named recipient(s) only and may be privileged. This message and any attachments has been scanned for viruses and dangerous content by ITABS Lanka Mail Scanner, and is believed to be clean. Although measures have been taken to ensure that this e-mail and attachments are free from any virus we advise that, in keeping with good computing practice, the recipient should ensure they are actually virus free. Please note that this message has been sent over public networks which may not be a 100% secure communications medium and ITABS Lanka cannot be held responsible for its integrity. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Giovanni Miano ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:
Re: [Asterisk-Users] (no subject)
I am having the same problem, but on both PSTN and a Voicepulse Connect IAX line. PSTN rings clicks dead air, then rings and connects, IAX just clicks, has dead air, rings and connects. Don't have a clue on how to fix it though. Greg Roger Johnsen wrote: I have a Wildcard TDM400P card being used with Asterisk. For some reason, incoming PSTN calls are getting delayed before they ring through on the Asterisk PBX to an extension. The calling party hears an initial ring tone and then a click noise, at which point it will then actually starts to ring the target extension. I had done some research and saw similar problems that seemed to relate to caller ID so turned it off but still had the same one ring, click delay problem. I've turned it back on and verified the behavior is the same so I'm not sure this is the problem. Essentially I'd like the call to ring through immediately and not perform this one ring click until the call is routed to the correct line. Has anyone else seen this and is there a way to fix the problem? Please let me know if I can provide any additional information. -Roger ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (no subject)
Jonathan k. Creasy wrote: 0930155701|cfg |3|00|0004f2022609.cfg could not be downloaded, getting next file. Any ideas? I attached the config files, I got them from somewhere else. The phone isn't finding the config file as the above log entry shows. The config file consists of the mac address of the phone with a .cfg appended. Doug ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (no subject)
It could potentially be both. I would look at your extensions.conf first
though. What does the extension entry for that context look like.
For instance I have an entry in my extensions.conf for dialing outside
lines (outside being from asterisk to my PBX and then onto the outside
world from there). The entry looks like this:
[to-analog]
exten = _9XXX.,1,Dial(ZAP/G1/${EXTEN})
exten = _9XXX.,2,Congestion
exten = _9XXX.,103,Hangup
To dial a PBX extension the entry would look almost the same:
[to-pbx-extension]
exten = _9XXX.,1,Dial(ZAP/G1/${EXTEN:1})
exten = _9XXX.,2,Congestion
exten = _9XXX.,103,Hangup
Hope this helps,
-Matt
On Wed, 2005-09-14 at 11:46 -0300, Pablo Allietti wrote:
hi all, i have a box with a te110p and a pbx siemens... connect both
with a e1.
with a xten soft i can call extensions numbers in my office example 100
102 etc. but when i truy to go outside with the 9 before the call rings
in the first extensions (100). this is a asterisk problem? or a pbx
problem?
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Re: [Asterisk-Users] (no subject)
unless you show us some config files, I doubt that anybody can help you... On Wednesday 14 September 2005 16:46, Pablo Allietti wrote: hi all, i have a box with a te110p and a pbx siemens... connect both with a e1. with a xten soft i can call extensions numbers in my office example 100 102 etc. but when i truy to go outside with the 9 before the call rings in the first extensions (100). this is a asterisk problem? or a pbx problem? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Insert Subject Here
Flobi wrote: I've been messing with it for a couple weeks with MySQL. It seems pretty good to me though I have had a couple crashes. I cane' say for sure that the crashes were directly related to RealTime though. Also, I'm still using CVS HEAD 2005-09-06 which was right before the beta release, I think. Flobi, please use the subject line. Its there for a reason. Secondly, if your system is crashing, how do you expect us to help debug the problem if you don't provide any info? Like backtrace's etc.. Read doc/README.backtrace for more info. -Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (no subject)
RTFM prashant yadav wrote: Hi, I m trying to install [EMAIL PROTECTED] after installing and logging in as root password i made network connections using netconfig command there i gave ip address as provided by my network provider it displays the ip address I m SORRY to ask that how can i access the net GUI if u can understand my problem plz let me know the commands and some information regarding the problem thanking you Prashant Yadav http://adworks.rediff.com/cgi-bin/AdWorks/sigclick.cgi/www.rediff.com/signature-home.htm/[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] (no subject)
I use BINK to burn ISO Images and it works great. Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Tuesday, August 30, 2005 11:09 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] (no subject) On Tue, Aug 30, 2005 at 05:27:37PM -0400, Mark Phillips wrote: Sounds to me like you copied the file to a disk rather than burn an ISO image. A common mistake folks make especially if they've never done an iso before. But then also wrote: What tools are you using? I prefer k3b. It rocks But also complicates the procedure when you want a simple ISO image burning. Hence the confusion with burning of the disk's files. cdrecord dev=whatever iso.image eject -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (no subject)
Sounds to me like you copied the file to a disk rather than burn an ISO image. A common mistake folks make especially if they've never done an iso before. What tools are you using? I prefer k3b. It rocks Mark prashant yadav wrote: having problems with installing [EMAIL PROTECTED] i downloaded the asteriskathome-1.5.iso file from asteriskathome.sourceforge.net link burned it on a cd but it is not booting what seems to be the problem hoping for a quick reply http://adworks.rediff.com/cgi-bin/AdWorks/sigclick.cgi/www.rediff.com/signature-home.htm/[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (no subject)
On Tue, Aug 30, 2005 at 05:27:37PM -0400, Mark Phillips wrote: Sounds to me like you copied the file to a disk rather than burn an ISO image. A common mistake folks make especially if they've never done an iso before. But then also wrote: What tools are you using? I prefer k3b. It rocks But also complicates the procedure when you want a simple ISO image burning. Hence the confusion with burning of the disk's files. cdrecord dev=whatever iso.image eject -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (no subject)
Depends on the phon you are using. Park will do that, and you should use park. On 8/28/05, bodra [EMAIL PROTECTED] wrote: Hi all i am developing a client for the asterisk that controls ur phone from an Xp c# application what functions in Asterisk that will allow you to put someone on hold but not park calls and bring them back, without using flash hook cuz it will be a button in that application Powered by Hellacious Riders - http://www.hriders.com Want to be able to access your mail via POP 3? Please view: http://www.hriders.com/web_page.cfm?web_pageID=94 for more info. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (no subject)
On Monday 15 Aug 2005 15:19, Tom Tobias wrote: I am using the correct version of pwlib(1.5.2) and openh323(1.12.2) for the stable asterisk build. Both packages configure and compile with no problems. However when compiling chan_h323 from the asterisksource/channels/h323 directory I get this error. It's not an error, it's a warning and it can be ignored. Chan-h323.h:31: warning; ‘sockaddr_in bindaddr’ defined but not used arc r libchanh323.a ast_h323.o There have been posts of similar messages but none with the specific syntax as the one above. Those posts have mentioned commenting some part of the source code in order to build. I have combed the Makefile and corrected any variables pointing to invalid directories. I have put the lines suggested into /etc/profile. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (no subject)
On Sat, Aug 13, 2005 at 08:10:03AM -0800, Cliff Savage wrote: The digium board will be in the same box. Does this mean: Channel 4 to incoming phone line. Channel 1 to DSL modem? Or DSL modem to the incoming line...and then the pass thru port on the DSL modem goes to Channel 4? Will this even work? I'd hate to have to switch to a cable modem. ADSL should not bother PSTN as long as you use a proper filter. In our case a proper filter was supplied by the phone company when we installed the ADSL line. We Happily use Asterisk with an FXO card and an ADSL connection from the same phone line. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (no subject)
I am having some problems with faxing in asterisk. I have a TE100P which is taking my PRI. This seems to be working fine. I also have a TDM400P with 2 FXS. Again card seems to be working fine, I can dial from phones attached to these to ports and everything seems to work fine. I have 2 DID's for my two fax machines that dial the Zap port for the FXS on the TDM400P. The problem comes in when I attempt to send or recv a fax on a fax machine attached to the TDM400P. Most of the time the fax will terminate part way through the transmission, however it will occasionally go through just fine but slow. I trying to resolve the situation I have tried many things. I first removed echo cancellation on both my FXS ports and my PRI. That didn't seem to have an effect at all other than more echo on voice calls. I do notice on my asterisk console that it seems to be disabling the echo cancellation when it hears the fax tones anyway. I then contact Digium, which won't official support faxing of course, but gave me some things to check. 1. No interrupt sharing. I disabled all unnecessary devices, like USB, etc, and had verfiied that the Digium hardware was on all on its own IRQ. This seemed to have not effect. 2. Disable Hyperthreading. I did this and it seemed to have no effect. 3. Ensure your not getting any NMI's. Which I did, and I was not accumulating NMIs.. 4. Runhdparm -t /dev/[Hard Drive Device] and notice if you hear crackles, pops, or loss of audio. I did get substantial interference with the audio when running this. Digum had me run the zttest application, which showed every couple of interations I would get a reading of 99.3 or 99.4 where they say anything below a 99.89 is bad. I can fax from FXS to FXS on the same card just fine. Which I think is an interesting point. Maybe it has to do with data passing across the bus to the t1 card? The only recommendation that Digium had left was to run the Hard Drive in DMA mode. Since I currently have a SATA drive in this system this isn't possible. I am willing to try running a PATA style drive in the system so DMA can be turned on. Number 4 above seemed to agree that it is possibly the SATA HD causing I/O issues on the sytem. Installing a new HD will be a time consuming process so I thought I would post to the list first to see if anyone else had any ideas/experiences. It's highly unlikely that you can do anything more to resolve the problem. It is the same problem that many of us have been discussing on this list for months regarding the TDM card and missed frames. I can say that folks are trying to narrow down the root cause and hopefully will have a fix in the next several weeks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (no subject)
Why not co-lo at a place like TxLink so you don't have to pay for public Internet bandwidth when all you're going to be using it for mostly is to get to someone like TxLink to do voip termination? VoIP termination with multiple carriers for LCR is relatively easy. VoIP origination is much harder, and that's where I'd see some high value in those DS3 line cards. Like you, I'm anxious for their arrival as well, but will wait until they've been GA for a bit as I wouldn't want to be a guinea pig with a DS3's worth of revenue riding on one card that just became GA. On 5/19/05, M O [EMAIL PROTECTED] wrote: BJ, BJ Weschke [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Do Both! :) Re: Telecom SIP termination vs. DS3 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 Did I miss pricing/availability announcements from Digium on that DS3 card somewhere? No idea. You can contact them if you dont know what you missed :) I wasn't aware they were going to be GA in less than 3 weeks from now. From my standpoint, I am just so anxious and confident that the Digium DS3 Channelized Voice PCI Card, whenever I get my order of DID #'s and test my configuration of Asterisk, that I am willing to prepay, or have available to Digium, whatever $$$ they want for the card. I am EVENTUALLY going to need it anyways, so I dont mind prepaying wheather or not it is available today! My knowledge of their product offering is no different than yours. But I fully intend on purchasing it :)! We are starting off with a 100Mbps burstable bandwith, though exspensive to start, after 30 days of usage, my bandwidth costs will look like $25K. Going off the top of head for a Sangoma DS3 Card @ $6000 per card, If I got 2 of them for $12,000 total, I eliminate, almost, that $25,000 per month bandwidth cost to me. So if Digiums DS3 Channelized Voice PCI card costs, around what Sangomas costs, $6,000, (JUST AS A EXAMPLE FOR THIS POST), $12,000 for 2 Digium DS3's in 1 month, I will save almost $10,000 AUTOMATICALLY and ever month thereafter! :) Come on Txlink DID #'s. Come on Digium with the DS3 Channelized Voice PCI card. Then all Digium would have left to do is create a board or work with someone on getting Radio Waves into your computer. :) Sincerely, SoftwareRadioGuy __ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (no subject)
On 4/27/05, Andre Normandin [EMAIL PROTECTED] wrote: Does anyone know what the [WARNING: . Changethread: Can't change device '**Unknown**'] line means below.. I just set verbosity to level 5, and noticed that error everytime a voicemail is left.. Everything seems to work ok, and I have no idea how long that error has been there, but I'm just curious if it is something important :-) Looks like the call is coming out of voicemail and then going somewhere else or you have an exten _. defined that is catching a hangup, post your extensions.conf for further analysis. Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (no subject)
Funny, they sell these old cards.. it seems like they are selling refurbs as new.. ... anyways RMA is on its way, would be nice if they would send one as a replacement first, so that we could continue our work and don't have to delay it. They can, its called cross-shipment, but they need a credit card number to ensure they get your return shipment. You have to ask for it. That's the way I did it. Thanks for the info .. Regarding the refurbs, if you or I were owners of digium, how would we handle a backstock of older (possibly refurb) cards when its somewhat known the old cards work fine in some systems? (And, we don't have a clue which systems/motherboards the cards worked fine in.) Well I guess one can also add location, as each phone provider will have different class 5 switches one connects to with different signaling basically .. But still, they should fix the problem once and for all with allowing flash firmware upgrades... hell even Adaptec can manage that .. For the TDM card its not a flash firmware issue. Based on what others have reported, etc, the Rev E/F card had a missing trace on the circuit board (others observed an added jumper a month or so after the card came out), the Rev H card (and maybe other rev's before that) had added components on the circuit board, plus the fxo modules apparently changed since the originals came out. The added jumper had something to do with module slot 1. Personally, it wouldn't bother me a bit if I received a replacement card with an added jumper as long as the card worked as expected. So, digium probably has a back-stock of earlier rev levels that might work just fine, but adding a jumper would not change the Rev level reported to the system. Without knowing specifically what was changed on each Rev, there is no way to guess at how refurb'ed cards should be handled. I hope the current Rev is stable, etc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
