Re: [asterisk-users] SPA112 flapping
Friday, June 17, 2016, 11:56:34 PM, Mike wrote: > I've got a device that seems to become unreachable for about 2 minutes, every > hour. From what I can tell, it isn't due to network or server issues. Any > ideas? The default registration time in spa112 is 1 hour. If registering is slow in your infrastructure, this can be the reason. -- Best regards, Gergomailto:csi...@gmail.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
[asterisk-users] The plain old PBX functionality
Hi, back in the old analog telephony days there was digital PBX-es and digital system phonesets. This phonesets have had many individual illuminatable buttons connected with extensions. The PBX can show on the buttons if some extension is ringing (blinks) or busy (constant light), and the user can transfer the call with one touch (pressing one of this button). I search this functionality in Asterisk. What versions, and what extension functions (or other settings), and what VoIP phones can do this? -- Best regards, Gergo mailto:csi...@gmail.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
[asterisk-users] The plain old PBX functionality
Hi, back in the old analog telephony days there was digital PBX-es and digital system phonesets. This phonesets have had many individual illuminatable buttons connected with extensions. The PBX can show on the buttons if some extension is ringing (blinks) or busy (constant light), and the user can transfer the call with one touch (pressing one of this button). I search this functionality in Asterisk. What versions, and what extension functions (or other settings), and what VoIP phones can do this? -- Best regards, Gergomailto:csi...@gmail.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
[asterisk-users] ISDN outgoing caller id
Hi, is anybody out there who can set the outgoing caller id on ISDN (CAPI or misdn) channels? I've tryed everything what I found in forums, os voip-info.com but no luck. I use a fritz card with CAPI in my first installation (1 BRI), and a hfc 4 port bri card with misdn on other. The first installation have p-t-mp configuration, the second one is p-t-p. Both configuration is EuroISDN in Hungary. So, can anybody help me? -- Best regards, Gergo mailto:csi...@gmail.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] ISDN outgoing caller id
Tuesday, August 27, 2013, 8:41:18 PM, Patrick wrote: On 08/27/2013 08:04 PM, Gergo Csibra wrote: Hi, is anybody out there who can set the outgoing caller id on ISDN (CAPI or misdn) channels? I've tryed everything what I found in forums, os voip-info.com but no luck. I use a fritz card with CAPI in my first installation (1 BRI), and a hfc 4 port bri card with misdn on other. The first installation have p-t-mp configuration, the second one is p-t-p. Both configuration is EuroISDN in Hungary. So, can anybody help me? Have you checked with your Telco if they allow you to change the callerid? If yes, are you setting the callerid to a number that you are allowed to use? You can't just set callerid to any number you like. You must own the number which you want to set callerid to. I have no problem setting the callerid on outgoing calls via chan_capi to one of the numbers that the telco assigned to me. Yes, of course I want to set our assigned numbers, becuse the called party sees Unknown now. -- Best regards, Gergomailto:csi...@gmail.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] Dialplan MySQL inserted ID
Tuesday, August 20, 2013, 5:47:24 PM, Gareth wrote:
On 20/08/13 14:53, Jonas Kellens wrote:
Hello,
how can I obtain the inserted ID after having inserted a row with
MySQL in the dialplan ?
exten = s,n,MYSQL(Query resultid ${connid} INSERT INTO myTable SET
C1=${ARG1}, C2=${ARG2},
timestamp=${STRFTIME(${EPOCH},,%Y-%m-%d_%H:%M:%S)})
I need to know the ID of the newly inserted row.
You could write an AGI script in something like php or perl and get it
to write to the mysql database instead. It can then set a variable which
the dialplan can pick up.
meh...
SELECT LAST_INSERT_ID()
--
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Gergomailto:csi...@gmail.com
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Re: [asterisk-users] Dialplan MySQL inserted ID
Tuesday, August 20, 2013, 6:08:19 PM, Jonas wrote:
On 08/20/2013 06:03 PM, Gergo Csibra wrote:
Tuesday, August 20, 2013, 5:47:24 PM, Gareth wrote:
On 20/08/13 14:53, Jonas Kellens wrote:
Hello,
how can I obtain the inserted ID after having inserted a row with
MySQL in the dialplan ?
exten = s,n,MYSQL(Query resultid ${connid} INSERT INTO myTable SET
C1=${ARG1}, C2=${ARG2},
timestamp=${STRFTIME(${EPOCH},,%Y-%m-%d_%H:%M:%S)})
I need to know the ID of the newly inserted row.
You could write an AGI script in something like php or perl and get it
to write to the mysql database instead. It can then set a variable which
the dialplan can pick up.
meh...
SELECT LAST_INSERT_ID()
Hello,
can I echo this variable ?
Like : exten = s,n,NoOp(${LAST_INSERT_ID()})
No, this is a mysql query, so:
exten = s,n,MYSQL(Query resultid ${connid} INSERT INTO myTable SET
C1=${ARG1}, C2=${ARG2},
timestamp=${STRFTIME(${EPOCH},,%Y-%m-%d_%H:%M:%S)})
exten = s,n,MYSQL(Query resultid ${connid} SELECT LAST_INSERT_ID())
exten = s,n,NoOp(${resultid})
first is your original insert query, next you must read the
last_insert_id() mysql function with an other query, then you can
echo the resultid variable which contains the last inserted id.
--
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Gergomailto:csi...@gmail.com
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Re: [asterisk-users] Top Posting
Sunday, December 30, 2012, 5:13:30 PM, Patrick wrote: On 12/30/2012 04:26 PM, Ron Wheeler wrote: I participate in a lot of lists and top posting is now the norm since people want to see quickly if the message is worth reading. Isn't it a bit of a stretch to extrapolate your experience with your lists to top posting being the norm? I am subscribed to several lists and bottom posting, proper trimming and commenting inline is the norm there. Actually the norm is determined by the list rules. If the list rules say one must use bottom posting then one should use bottom posting. If someone does not like that then don't subscribe, find another source to ask a question (the forum, LUG, hire a consultant) or just bottom post. Questions come before answers. Answers come after questions. -1 against changing rule #5. Complaining about top posting on a list where's no moderation, no sanction if somebody top posting is pointless. but... -1 against changing rule #5. -- Best regards, Gergomailto:csi...@gmail.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] mISDN and 1.8
Monday, September 26, 2011, 11:33:50 PM, Kristijan wrote: Gergo, why do you want to use mISDN? Use Dahdi. Or do you want to use a very exotic isdn card which is only supported by mISDN? tell us more. Well, I want to use one channel driver for all my installations. Now I have to reinstall a machine with an AVMfritz card. But I have 3 other machine with HCF-S cards. My long time experience with mISDN_v1 is, v1 has major echo and fax problems. because the audio signals are transported very unsynchronic because of the kernel driver architecture which was made for data transmission, and originally not for voice and fax applications. misdn echo cancellation works for me. mISDN_v2 with chan_lcr has vital improvements for voice. especially together with OSLEC as echo canceller. but still has some issues with fax transmission. And since Dahdi supports the important BRI card based on HFC-8S/4S colonge chip and via http://code.google.com/p/zaphfc/ also the cheap HFC-S cards. No reason anymore to use mISDN. Yes I've tested this too, but it's very undocumented. I like how-tos, after how-to-s i read the install.txt or readme.txt, but for zaphfc there's nothing. Or the version... on the page you linked states zaphfc works with dahdi-2.1.0.4 2.2.0. The updates list speaks about 2.3.0.1, and asterisk.org says the actual version of dahdi is 2.5.0.1. How can I trust in a driver, which has so many contradictions only in version numbers? :) -- Best regards, Gergomailto:csi...@gmail.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
[asterisk-users] mISDN and 1.8
Hi, are there anybody, who using the chan_misdn included with Asterisk v1.8? If yes what mISDN version used v1 or v2? Yes, I can read on many pages for mISDN2 I need to use chan_lcr, but this informations are 2-3 years old, and I can't imagine asterisk v1.8 chan_misdn works only with linux kernel v2.6.24 which is quite old. -- Best regards, Gergo mailto:csi...@gmail.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] mISDN and 1.8
Monday, September 26, 2011, 7:20:10 PM, Kevin wrote: On 09/26/2011 11:35 AM, Gergo Csibra wrote: Hi, are there anybody, who using the chan_misdn included with Asterisk v1.8? If yes what mISDN version used v1 or v2? Yes, I can read on many pages for mISDN2 I need to use chan_lcr, but this informations are 2-3 years old, and I can't imagine asterisk v1.8 chan_misdn works only with linux kernelv2.6.24 which is quite old. chan_misdn only supports mISDN version 1. This means I need to hack a v2.6.24 kernel remove the mISDNv2 and install mISDN v1? -- Best regards, Gergomailto:csi...@gmail.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] pickupgroup
Thursday, August 4, 2011, 9:35:50 PM, Dan wrote:
Is there any technical reason for this 63 group limit?
Currently the number of callgroups/pickupgroups is limited to 63, because the
variable type that holds the group bits is a long long.
Ah, so it's a major change to get it to hold a larger number?
I was hoping I could just change the following line:-
if ((x 63) || (x 0)) {
OMG.
Group bits means, there is a 64 bit long register, and for the
pickupgroup 1 is the first bit set, for the pg2 is the second and so
on. Imagine some extension can be in more than 1 pickupgroup. In this
situation more than 1 bit set in that variable. There's no x here.
--
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Gergomailto:csi...@gmail.com
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Re: [asterisk-users] Master.csv file limit
Friday, July 8, 2011, 3:13:01 PM, deeps wrote: What is the maximum size limit of Master.csv file and what happens when it reaches limit? That is a text file. Only limited by filesystem. -- Best regards, Gergomailto:csi...@gmail.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] Why PRI not BRI ?
Sunday, May 29, 2011, 10:57:00 AM, virendra wrote: I have stupid question but I want to know it. Why we use the PRI insted of BRI ? Just for the sake of number of lines or any thing else ? Yes, because of much more channels. But if you need only 2 or 4 channels BRI is cheaper. From 10-12 channels becomes PRI cheaper. And the other reason: if you use traditional channel banks instead of VoIP phones that uses PRI also. And why SIP is used for making calls rather then IAX? Even we know IAX takes 1 channel for making calls? Maybe because of almost every VoIP phones knows SIP. I personally didn't meet any IAX VoIP phone. -- Best regards, Gergomailto:csi...@gmail.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] fail-over server
Tuesday, February 8, 2011, 5:07:29 PM, Vieri wrote: How can I minimize this time lapse? Can Asterisk notify all SIP clients in its sip.conf that they need to acknowledge being on-line or not (thus forcing re-registration in my scenario)? If you have two identical servers online, it is better to make a HA sollution. Sorry, I haven't made HA Asterisk yet, I can not help more. -- Best regards, Gergomailto:csi...@gmail.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] Asterisk Performance
Tuesday, February 1, 2011, 11:22:30 PM, Juan wrote: I would like to handle about 250 simultaneous (calls agents only) calls with PRI or a SIP trunk with the following configuration Dell R710 Dual Intel® Xeon® X5650, 2.66Ghz, 12M Cache,Turbo, HT, 1333MHz or Single Intel® Xeon® X5650, 2.66Ghz, 12M Cache,Turbo, HT, 1333MHz Memory 12GB, 1333MHz RAID 1 - 1 Tb X 2 Is that possible?? This is an overkill machine for that :) -- Best regards, Gergomailto:csi...@gmail.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] SPA3102 FAX not working
Thursday, September 9, 2010, 4:32:29 PM, Gopalakrishnan wrote: I am sending FAX from one extension to another extension. I am not able to send. Preferred Codec:G711u You forget to mentoin where do you live? In some countries the G711a codec and in onther countries the G711u codec useable. -- Best regards, Gergomailto:csi...@gmail.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] Asterisk and RAID
Wednesday, August 4, 2010, 6:02:45 PM, Danny wrote: R5 would use 3 out of 4. You can have R5 across 10 drives too. Yes, the writes will be slow, but it possible. -- Best regards, Gergomailto:csi...@gmail.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] ISDN - SIP
Friday, June 11, 2010, 12:27:08 AM, Tzafrir wrote: On Fri, Jun 11, 2010 at 12:19:37AM +0200, Gergo Csibra wrote: Okay. There's some problems with mISDN v2: I'm unable to compile zaphfc, because there's no source for it. mISDN v2 works with hfcpci too? Certainly there is. It's also part of the standard dahdi-extra patch. See http://git.tzafrir.org.il/?p=dahdi-extra.git;a=tree http://svn.debian.org/viewsvn/pkg-voip/dahdi-linux/trunk/debian/patches/dahdi_linux_extra OK. Last time I checked (2009. dec) there wasn't :) I downloaded dahdi-extra snapshot, and dahdi from asterisk.org, untared, I have two directories: dahdi-extra dahdi-linux-complete-2.3.0.1+2.3.0 What's next? I don't understand where to start make with MODULES_EXTRA and SUBDIRS_EXTRA parameters, and how can I configure drivers... -- Best regards, Gergomailto:csi...@gmail.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] Y-cords - What are they ?
Tuesday, July 6, 2010, 8:11:46 PM, bruce wrote: Can someone please explain what Y-cords are available out there and how they can be used with Aastra or other VoIP phones? Maybe with or WITHOUT headsets? Isn't a Y-cord traded for soft Barge in these days? I think Y-cords only for PSTN. Or there're Y-cords for twisted pair ethetnet too, but that not a good idea. Usualy VoIP phones includes a mini 2 port switch to use one switch port for a phone and a PC. -- Best regards, Gergomailto:csi...@gmail.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] ISDN - SIP
Thursday, June 10, 2010, 11:19:16 PM, Philipp wrote: i am using asterisk 1.6.2 with mISDN and a passive AVM Fritz!card on CentOS 5.5. The only thing, i want to do is a call-redirection from an isdn-call to my mobile via sip-account. Unless you are using mISDN v2: Do yourself a favour and switch to CAPI with chan_capi and fcpci. mISDN v1 is a guarantee for headaches (and unstable systems). Okay. There's some problems with mISDN v2: I'm unable to compile zaphfc, because there's no source for it. mISDN v2 works with hfcpci too? -- Best regards, Gergomailto:csi...@gmail.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
[asterisk-users] This is a test, hijack this
Hello Asterisk, This is only a test, because I can't start new thread in this list... -- Best regards, Gergo mailto:csi...@gmail.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] This is a test, hijack this
Wednesday, March 24, 2010, 9:42:52 PM, Dave wrote: On 03/24/2010 03:56 PM, Miguel Molina wrote: Gergo Csibra escribió: This is only a test, because I can't start new thread in this list... If you can send an email, you can start a new thread on this list. What's the point of all this? He was probably having the same problem I've had where I can reply to exiting threads fine but any time I send a fresh email to start a new thread it never goes through. Murphy's law being what it is this email that he suspected would never go through... did. Your diagnosis is perfect :) Normally I read lists in digest mode, with mime-digest I can reply to individual messages. I use another e-mail address to do this. But I can't start new thread with this (other) e-mail address. With this I can (as we seen :)). -- Best regards, Gergomailto:csi...@gmail.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] This is a test, hijack this
Wednesday, March 24, 2010, 10:35:20 PM, Karl wrote: Gergo is using Gmail, He is also using an IMAP client to author his messages (as I do). No. I don't use any IMAP thing :) I use The Bat! but only to download messages with POP3, and I send messages through my ISP's SMTP server. -- Best regards, Gergomailto:csi...@gmail.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] subject: 1.4 vs 1.6
Wednesday, February 24, 2010, 3:56:50 PM, David wrote: On Wed, Feb 24, 2010 at 9:20 AM, Juan Sandro juan.san...@hotmail.com wrote: Hi Guys We are using asterisk 1.4 on all of our platforms for a while now. Some of our partners recommended to use asterisk 1.6 in order to improve overall stability and performance. Can someone please let me know if you have a such experience? Also, do you have any other negative or positive comments on 1.6 If it isn't broke, don't 'fix' it. There are benefits to 1.6, like dramatically enhanced SIP support, much faster dialplan processing, easier faxing, changes to dialplan syntax, and lots of other features. I would say the improvement of going to 1.6 is only if you are trying to expect more from the same gear, or want the new features. If you're not actually having problems, don't change anything. Yes, and check this page: http://www.asterisk.org/asterisk-versions as you can see, the 1.4 version is LTS, and the 1.6 isn't, but the upcoming 1.8 will be LTS too. So don't change to 1.6 :) -- Best regards, Gergomailto:csi...@gmail.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] Static IP
Wednesday, February 17, 2010, 10:40:18 PM, Steve wrote: Congratulation. Doesn't it feel great to help yourself rather than bothering the mailing list with questions that have nothing to do with Asterisk? And it only took you 17 minutes! Much better than cool dude :) -- Best regards, Gergomailto:csi...@gmail.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] call transfer
Tuesday, February 16, 2010, 12:55:12 PM, cool wrote: call comes it should be received by extenion 2000, n if person wants to talk to Sales, receptionist should put the caller on hold than connect to Sales i.e exten 2001, while on hold the caller should hear music on hold,now sale exten can take his call n talk to it.same with Accounts ext 2002. ... what to next To have call transfer in your asterisk setup, YOU need to read some documentation. Start here: http://www.voip-info.org -- Best regards, Gergomailto:csi...@gmail.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] how to allow some extenstions to call outside and some extensions cant call outside
Friday, February 12, 2010, 9:57:42 PM, cool wrote: how to allow some extenstions to call outside and some extensions cant call outside. i am attaching sipand extensions.conf thx Put the extensions into different contexts, and create outside call extensions only in the allowed context. Remember to create outside calls for emergency numbers in the other context too. -- Best regards, Gergomailto:csi...@gmail.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] Virtual Asterisk Installation
Wednesday, January 20, 2010, 11:41:48 PM, Michiel wrote: Forget about virtualization! ... Virtualisation is nice for test-setups, but thats it. for any real job it's a major pain in the ass and makes stuff bork beyond imagination. Well. Why do you use computer? There're slide-rule. You can calculate anything with that... -- Best regards, Gergomailto:csi...@gmail.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] Virtual Asterisk Installation
Thursday, January 21, 2010, 12:53:09 AM, Jeff wrote: On Thu, 21 Jan 2010, Gergo Csibra wrote: Wednesday, January 20, 2010, 11:41:48 PM, Michiel wrote: Forget about virtualization! ... Virtualisation is nice for test-setups, but thats it. for any real job it's a major pain in the ass and makes stuff bork beyond imagination. Well. Why do you use computer? There're slide-rule. You can calculate anything with that... Pretty crappy analogy. Just because you *can* do something doesn't mean it is production ready. Yes. It was an exaggeration. But saying virtualisation isn't for any real job is ROTFL. Every computer system is bigger than a PC is virtualised. Yes for asterisk virtualisation is not an option because of context switching, but for a webserver, other file and application server or database is OK. -- Best regards, Gergomailto:csi...@gmail.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] You won't help me anymore?
Sunday, January 10, 2010, 11:24:22 AM, hadi wrote: You are not willing to help me anymore ? Why do you think this? -- Best regards, Gergomailto:csi...@gmail.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] PBX Extension Help
Friday, January 1, 2010, 7:12:54 PM, Alex wrote: On 01/01/2010 01:06 PM, Warren Selby wrote: Also, shouldn't the .php script be located in /var/lib/asterisk/agi-bin? Fact. And on a live channel must use AGI instead of DeadAGI. And man should not topposting on a maillist... -- Best regards, Gergomailto:csi...@gmail.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] CDR_MYSQL 1.4 Database Structure
Wednesday, December 30, 2009, 6:48:37 PM, Robert wrote: Tilghman Lesher wrote: On Wednesday 30 December 2009 10:52:48 Robert Broyles wrote: Just curious if anyone has successfully patched cdr_addon_mysql to use accept the latest cdr fields from 1.4 ... namely: 'start', 'answer', 'end'? Seems logical that the cdr_mysql addon should be updated to reflect the current cdr. And for backwards compatibility it can still accept 'calldate'. The MySQL driver contains all of the same information, albeit in a slightly different form. Calldate is the same as start, calldate plus duration minus billsec is the same as answer, and calldate plus duration is the same as end. Generally, we do not make design changes in the middle of a release cycle, especially given that such changes would break a great many existing systems. Given that there's no security reason why we would need to make such a change, it is out of the question. While you're certainly welcome to make such a change on your own systems, such a change will not be committed in the 1.4 addons. In the 1.6 series and forward, we've changed the mysql driver to scan the table metadata and adapt the queries to the table structure. Therefore, you could, in fact, use 'start', 'answer', and 'end' in the 1.6 series, as you suggested, above, and it would work perfectly well. On the other hand, if you kept the legacy structure, that would work, too. Thanks for the reply. So my next question is could I take the cdr_mysql from 1.6's addons and use it in 1.4? I don't think so. But you can define more columns, and an insert trigger which calculates the missing fields as defined in Tilghman's reply. -- Best regards, Gergomailto:csi...@gmail.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] Recording the Calls to a USB Drive
Thursday, December 24, 2009, 5:41:46 PM, Danny wrote: Just my opinion; unless you are recording long or many long calls, you should record to your local drive, then copy the files to the USB drive. Asterisk is a very good tool - you don't need to mess it up by introducing an easy point of failure. Yes. I do this since 3 years and work very well. -- Best regards, Gergomailto:csi...@gmail.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] Asterisk depend on postgresql files?
Wednesday, December 23, 2009, 11:17:39 AM, ABBAS wrote: when compiling asterisk with Postgresql we need to specify directory where the postgresql is installed. I need to know once asterisk is ready to use(ie compiled and installed ). Do it still refer the postgresql files that are not part of asterisk ? By deleting the postgresql installation folder will it effect asterisk functionality? Well, for compiling generally you need to install -dev packages (postgresql-dev or something), and client packages. After compilation you can delete -dev packages, but the client packages must be installed for database access. -- Best regards, Gergomailto:csi...@gmail.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
[asterisk-users] dahdi 1.6 zaphfc bristuf what???
Hi, something very strange here. I've downloaded asterisk 1.6.1.11 and dahdi-linux-complete-2.2.0.2+2.2.0. My linux kernel is 2.6.30.9. After compiling and installing dahdi it found my hfc based card, but at reboot it says driver should be 'zaphfc' but is actually 'hfcpci'. Ok. This hfcpci comes from mISDN integrated in linux kernel. How can I get zaphfc? I've tried to download an compile bristuff, but it downloads asterisk 1.2 or 1.4 and do some other weird things. I want to use 1.6 with hfc based card. Why is this so complicated? -- Best regards, Gergomailto:csi...@gmail.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] Eicon Diva 2.01 PCI Passive BRI ISDN card
Saturday, April 4, 2009, 3:13:12 PM, Puskás wrote:
Got it working with Asterisk 1.2 installed on the same PC as Asterisk 1.4 [
in
different directorys and username of course :) ] . Using isdn4linux kernel
module and Dial(Modem/ttyI0/1234567:${EXTEN}) command.
Használj MISDN-t, és ne toppostolj.
Use MISDN, and do not toppost!
--
Best regards,
Gergomailto:csi...@gmail.com
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Re: [asterisk-users] faxing via linksys SPA3102 half page goes through
Wednesday, March 4, 2009, 5:43:13 PM, Joseph wrote: On 03/04/09 15:56, Gergo Csibra wrote: Wednesday, March 4, 2009, 3:22:59 PM, Joseph wrote: FAX Passthru Codec: G711u for me FAX works better with G711a Can you folks compare my setting below with your settings and let me know if something differ. I was experimenting with echo in the past and might have triggered something :-/ I know many settings have nothing to do with the fax but something must have trigger this problem, so I'm listing all the settings. Well, I use Linksys PAP2, but the settings mainly the same. Audio Configuration Preferred Codec:G711u Use Pref Codec Only:No This is the only difference, I use G711a and pref codec only: yes FAX Disable ECAN:No How about that FAX Disable ECAN:No was is the default to Yes or No This was default no, and I use this setting. -- Best regards, Gergomailto:csi...@gmail.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] faxing via linksys SPA3102 half page goes through
Wednesday, March 4, 2009, 3:22:59 PM, Joseph wrote: FAX Passthru Codec: G711u for me FAX works better with G711a -- Best regards, Gergomailto:csi...@gmail.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] top posting again [was: Re: CDR Design]
Friday, December 5, 2008, 2:49:59 PM, Andrew wrote: Address added to spam filter. Please do NOT e-mail me again. A: Because it messes up the order in which people normally read text. Q: Why is top-posting such a bad thing? A: Top-posting. Q: What is the most annoying thing in e-mail? -- Best regards, Gergomailto:[EMAIL PROTECTED] ___ -- 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] Cheapest 4 port FXO
Sunday, October 26, 2008, 12:31:16 PM, Hans wrote: On Sat, 2008-10-25 at 11:54 -0600, Joseph L. Casale wrote: I need to increase reliability at an office as SIP/Internet provider outages are causing some issues. What would be the least expensive analogue card that people are using reliably? If its for reliability, i wouldn't recommend x100p's Have a look at ata's. Either four sipura/linksys/cisco 3102 or their eight port version. You can put those tiny boxes directly behind your phone/fax. Well, there're linksys SPA400 what have 4 FXO. -- Best regards, Gergomailto:[EMAIL PROTECTED] ___ -- 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] chan_misdn troubles
Tuesday, September 23, 2008, 11:57:00 AM, Thanos wrote: Hello I have just set up Asterisk Asterisk 1.4.21.2 on a CentOS 5.2 machine. I am using the OpenVox B200P ISDN card. My problem is that even though chan_misdn module seems to be loaded correctly with Asterisk (I can see it using 'module show' command) the misdn commands are not available to me in the CLI so I cannot tell if my box is correctly interfacing with the ISDN card Any ideas what can be going wrong ? ... cd ../mISDN-1_1_7_2/ What kernel version you use? Newer linux kernels (2.6.24) works only with new (and beta) 1.1.8 misdn. -- Best regards, Gergomailto:[EMAIL PROTECTED] ___ -- 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
[asterisk-users] how to detect pickup...
Hello asterisk-users, My SIP phones are in pickupgroup, and if some of them ringing from other phone can pick up with *8 as usual. But I want to know if this happen. I've tried the a extension, but seems not working. Any other idea? -- Best regards, Gergo mailto:[EMAIL PROTECTED] ___ -- 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
[asterisk-users] Executing dialplan after the call normaly ended
Hi, The Dial command has the g option, voip-info.org says: If the g option is specified, and the called party hangs up before the calling party, then Dial continues execution at priority n+1. and this works well. But I need to continue the execution if the caller hangs up first too. What do I need to do? -- Best regards, Gergomailto:[EMAIL PROTECTED] ___ -- 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] How to detect if SIP extension BUSY?
Wednesday, February 13, 2008, 2:56:42 PM, Michiel wrote: On 13:14, Wed 13 Feb 08, Gergo Csibra wrote: Well? Is it impossible to detect BUSY on SIP channels? not in stock 1.2 Bristuff has a function for it, and russell created a function for it in current trunk that is also available as patch to 1.4 So you have 2 possibilities: - install 1.4 with russell's patch applied asterisk + function from russell: http://www.voip-info.org/wiki/view/Asterisk+func+Devstate OK. I've tried Russel's devstate function and also SIPPEER(ext:curcalls) function but they have the same problem. Devstate shows only called SIP extensions as BUSY, but caller is not, and SIPPEER(ext:curcalls) shows only call count 1 on called extension, on caller is still 0. So if I make a call for example from sipext1 to sipext2 the devstate shows NOT_INUSE on sipext1 and BUSY on sipext2, but in real world they both BUSY, also with curcalls, on sipext1 the call count is 0, and on sipext2 is 1. And in dialplan I can only test the CALLERID(num) or CALLERID(name) but they are user setable, so if some user changes it, I must to modify my dialplan. Another way? -- Best regards, Gergomailto:[EMAIL PROTECTED] ___ -- 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] How to detect if SIP extension BUSY?
Saturday, February 9, 2008, 10:29:08 AM, Csibra wrote: My problem is in subject. As I read in documentations and voip-info.org I can't user ChanIsAvalil because it not detects BUSY information on SIP channel. I've tried to use SIPPEER function, but it gives OK (9 ms) back on BUSY SIP channel. I use Asterisk 1.2.15, SIP extensions are Linksys PAP2. Have any other idea? Well? Is it impossible to detect BUSY on SIP channels? -- Best regards, Gergomailto:[EMAIL PROTECTED] ___ -- 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] [Zaptel] Why no port to Windos?
Friday, December 14, 2007, 5:47:38 AM, Paul wrote: Umm - you could just buy a SPA-3000/3102/3666/etc. What is SPA-3666? -- Best regards, Gergomailto:[EMAIL PROTECTED] ___ --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] 16 ports wanted
Monday, October 22, 2007, 9:47:49 AM, Rilawich wrote: Hi all, I want to have a 16 FXO in a PC. Is it possible to use 4 x TDM404 or 2 TDM808 to get 16 FXO? What is the difference (in performance and control) in using 4 x TDM404 and 2 x TDM808 if possible? ango Well, using more than one TDM card in your PC is not a good idea, because of interrupts. If you have to have 16 FXO you can more options: 1. Using TDM2400P with 4 FXO modules ($1775) 2. Using Xorcom's Astribank (external) ($1170) 3. Using some T1/E1 card with Channel Bank (more expensive) -- Best regards, Gergomailto:[EMAIL PROTECTED] ___ --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] SPA400 and asterisk
Wednesday, June 13, 2007, 9:44:08 AM, Nick wrote: I'd love to get it working... if you could share a sample config or other advice, I'd appreciate it. http://www.voip-info.org/wiki/view/Linksys-Cisco+SPA400 http://voxilla.com/voxilla-tips/voxilla-tips/the-linksys-spa400-and-asterisk-886.html http://www.justfuckinggoogleit.com/ -- Best regards, Gergomailto:[EMAIL PROTECTED] ___ --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] FXO recommendation
Friday, May 4, 2007, 10:42:13 AM, Phil wrote: On Thu, May 03, 2007 at 08:17:25PM +0100, Kyle Gordon wrote: With the gamut of FXO cards out there, I'm looking for a recommendation for home use. I have a nicely working Asterisk 1.4 system that just requires an FXO card to connect my NTL PSTN to it. My previous X101P clone seems to have kicked the bucket. Any suggestions would be greatly appreciated. Well, I've never connected an NTL, er, Virgin PSTN line to Asterisk, but if I were you I'd consider whether you might want further ports in the future - if so, go for at least a TDM400P with just the one module for now. Well this is a digium list, so here will be digium cards recommendation. But You can use a linksys spa3102, that costs about half price of TDM400P. -- Best regards, Gergomailto:[EMAIL PROTECTED] ___ --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] FXO recommendation
Friday, May 4, 2007, 3:06:02 PM, Dave wrote: On Fri, 2007-05-04 at 08:35 -0400, Joe acquisto wrote: Gergo Csibra [EMAIL PROTECTED] Wrote: 5/4/2007 8:12 AM: Yes it is an ATA with an FXS and an FXO port, and you can use as many as you want instead of one TDM400/TDM800/TDM2400. It has two RJ11 ports Yes, one for FXO, one for FXS and an RJ45 Two RJ45, one for local network, one for Internet (if you use this box for voip subscription). , but it cannot replace a TDM card, which is what I thought you were suggesting. Depends on what you mean by replace. Physically no, but functionally yes. You can reach the FXS and FXO port as a simple SIP client, so it works without zaptel. I missed the original posting. Since no one else has spoken up, perhaps I am off base. Please help clear up what I am missing. Original poster wants to have an FXO port. There's many sollution for that, TDM400P is only one (and maybe the most expensive) of them. -- Best regards, Gergomailto:[EMAIL PROTECTED] ___ --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] Asterisk on Debian Etch
Monday, April 23, 2007, 12:44:08 PM, Diego wrote: you need to use apt-get install asterisk. If you MUST HAVE 1.217 or your cats die, there are repositories available. For example, read this: http://www.buildserver.net/ If you still MUST build asterisk yourself, I wish you good luck. Well, it works for me from source, without any issue. Ok, I use mISDN instead of zaptel. -- Best regards, Gergomailto:[EMAIL PROTECTED] ___ --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] misdn and debian
Monday, April 2, 2007, 7:30:57 PM, Giedrius wrote: Has anybody debian and misdn working fine? Maybe you can advices , what kernel and misdn versions to use... I use kernel 2.6.20.1 and misdn 1.1.0 with fritz card, and working fine. The kernel, asterisk (1.2.15) and misdn also compiled from source. -- Best regards, Gergomailto:[EMAIL PROTECTED] ___ --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] Linksys SPA 3102 causing me problems
Friday, March 30, 2007, 5:02:08 AM, Matt wrote: On 3/29/07, Alan Chandler [EMAIL PROTECTED] wrote: I have a linksys SPA 3102 with a DECT phone connected into its Telephone port. It has been working, but something I've done (and I don't know what) means that now everytime asterisk tries to dial it, it says it is busy. I can make calls from it through asterisk I am at a complete loss to know what to try next to fix it. Any ideas? I dont know if you have done this but run a sip show peers and make sure that its registered with asterisk. Sounds like it is not registering with asterisk which would allow you to call out but when it tries to call you it dosent have an ip to contact you at. Wehh... He activated the DND function of Linksys. It can be activate with *78 and deactivate with *79. -- Best regards, Gergomailto:[EMAIL PROTECTED] ___ --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] FAX mISDN
Thursday, March 29, 2007, 7:18:43 PM, LKS wrote: Hi folks! Does anybody know how to receive send faxes throw mISDN? It's almost impossible! Describe your problem, but read this before: http://www.catb.org/~esr/faqs/smart-questions.html It works for me, in 3 places, the analogue fax machines connected to a Linksys PAP2. Everything is the default settings, comes from make samples, only edited the sip.conf and extensions.conf -- Best regards, Gergomailto:[EMAIL PROTECTED] ___ --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] asterisk on debian
On 3/20/07, Josu Lazkano Lete [EMAIL PROTECTED] wrote: I want to install Asterisk on a Debian machine. I need to download the sources or just with apt-get install is enought??? It depends on what version do you want to use. In sarge is only the version 1.0.7. In etch is 1.2.13, but the 1.2 branch is at 1.2.16 and there's the version 1.4.1 is out also. ___ --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] disable client side hangup after dialing 911
On 3/9/07, Wai Wu [EMAIL PROTECTED] wrote: Two things. 1) This is a bug(feature) of standard analog switchs which only clear the talk path when both sides of the call are terminated. Well, not exactly. The call will not terminated until the caller (not both) hangs up. I don't knew the american emergency numbers, but in europe, if the caller hangs up, the call will terminated. If the called party hangs up, the call will not terminated. ___ --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
[asterisk-users] Linksys PAP2 and Caller ID
Hi! Can I use my Linksys PAP2 with asterisk and an analog CLIP phone to show the Caller number on the phone. There's a Caller ID Method: option on Regional settings, but I tested all options, and my CLIP phone never shows the Caller number... :( Any idea? ___ --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
