Re: [asterisk-users] {s} - extension
On Wed, Mar 5, 2008 at 10:12 AM, Daniel Suleyman [EMAIL PROTECTED] wrote: but when I use next construction(As I understand it is used to allow to process any extension dialed by user) exten = s,1,Answer; exten = s,2,Playback(hello-world,skip); exten = s,3,Hangup; AFAIK, s extension is used in analogue PSTN incoming calls, as the call itself doesn't contain the extension (public telephone number) it tries to reach. If want to catch any extension dialed by the user you should use something like this: exten = _.,1,Answer; exten = _.,2,Playback(hello-world,skip); exten = _.,3,Hangup; In any case, I cannot understand why you would like to use it. You should define your extensions and use the invalid extension (i) to catch calls sent to any number not detailed in the dialplan. -- Andres Jimenez GPG : http://www.andresin.com/gpg/[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] {s} - extension
On Wed, Mar 5, 2008 at 12:04 PM, Daniel Suleyman [EMAIL PROTECTED] wrote:
The idea is that the person connecting and dial anything he want and
the script is deciding to proceed the call or to terminate it(I think
it will be easy to manage extensions.conf - no need to create
extensions).
It is easy. I meant you have to configure extensions (what to do
when a number is dialed) and handle unknown extensions with the i
extensions.
You know {i} doesent work
exten = i,1,Answer;
exten = i,2,Playback(welcome,skip);
exten = i,3,Hangup;
as I thought when i will dial wrong number it will play welcome message but
asterisk promt - Call from 7007 to extension 700786 rejected because
extension not found.
Have you reloaded your extensions.conf file?
in asterisk CLI extensions reload
--
Andres Jimenez
GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED]
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Re: [asterisk-users] {s} - extension
On Wed, Mar 5, 2008 at 1:36 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: This is not needed. If the extension is not found, there is a fallthrough to 's' (Right? Or is it chan_zap-specific)? I would say it's chan_zap-specific. From http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf For some kinds of connections — such incoming calls from an outside telephone line — the user has not dialed an extension. In that case, Asterisk behaves as if the user had dialed a special extension named s (for Start). Asterisk will look for an extension number s in the definition of the context for that channel for instructions about what it should do to handle the call. The key factor is that s is used when NO EXTENSION has been specified (when the call is not clearly directed to an specific number). As far as I know, that's the way analog lines behave. The line just receives the call, but no information says to which number the call was sent. -- Andres Jimenez GPG : http://www.andresin.com/gpg/[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
[asterisk-users] Problems configuring Astribank
Hi, all My Asterisk uses a Digium TE120Pand I would like to add an Astribank zaptel_hardware sees is, but I cannot get it working pbx:~# zaptel_hardware Argument IRQ isn't numeric in numeric comparison (=) at /usr/local/share/perl/5.8.8/Zaptel/Span.pm line 114. usb:005/002 xpp_usb- e4e4:1131 Astribank-8/16 USB-firmware pci::04:00.0 wcte12xp+d161:0120 Wildcard TE12xP In the Astribank line (module part) there is a - sign after the module name (while the TE120P has a + sign). Does that mean anything? genzaptelcong doesn't generate any config for the Astribank. Can anyone send a copy of their working Astribank settings? I am concerned about the span used by the Astribank. I just cannot find anything that gives me a clue... Thanks in advance. Regards, -- Andres Jimenez GPG : http://www.andresin.com/gpg/[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] Problems configuring Astribank
I just upgraded zaptel to 1.4.9.2 (and rebuild everything, of course) but no improvements. zaptel_hardware's output is the same and genzaptelconf I forgot to mention that zapconf fails too... pbx:~# zapconf Argument IRQ isn't numeric in numeric comparison (=) at /usr/local/share/perl/5.8.8/Zaptel/Span.pm line 114. Failed probing type for channel IRQ at /usr/sbin/zapconf line 230. Cheers, Andres On Tue, Mar 4, 2008 at 7:04 PM, Andres Jimenez [EMAIL PROTECTED] wrote: Hi, all My Asterisk uses a Digium TE120Pand I would like to add an Astribank zaptel_hardware sees is, but I cannot get it working pbx:~# zaptel_hardware Argument IRQ isn't numeric in numeric comparison (=) at /usr/local/share/perl/5.8.8/Zaptel/Span.pm line 114. usb:005/002 xpp_usb- e4e4:1131 Astribank-8/16 USB-firmware pci::04:00.0 wcte12xp+d161:0120 Wildcard TE12xP In the Astribank line (module part) there is a - sign after the module name (while the TE120P has a + sign). Does that mean anything? genzaptelcong doesn't generate any config for the Astribank. Can anyone send a copy of their working Astribank settings? I am concerned about the span used by the Astribank. I just cannot find anything that gives me a clue... Thanks in advance. Regards, -- Andres Jimenez GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED] -- Andres Jimenez GPG : http://www.andresin.com/gpg/[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] Cisco 79xx users/consultants, 7970G color in particular share information
On Sat, Mar 1, 2008 at 6:00 PM, Lacy Moore [EMAIL PROTECTED] wrote: Not sure on #1, but #2 is not possible on SIP. Busy Lamp Field IS available on SIP. If it is not, I cannot imagine how my GXP-2000 does it. -- Andres Jimenez GPG : http://www.andresin.com/gpg/[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] [URGENT] Zap channels fail to load
On Wed, Feb 27, 2008 at 10:25 AM, linuxian iandsd [EMAIL PROTECTED] wrote: i believe your problem is at the hardware/driver/provider level so you will be looking at the : zaptel.conf zapata.conf files Thanks for the advice. I will give it a try this evening. my second advice is : clonezilla ! it will clone any systeme from ide to sata to raid 5 make you feel calm confidente when doing changes to critical system components knowing you have your files safe on your other RAID5 server. believe me that would make you more productive because you are concentrating on the programing you are doing rather than being overtaken by the fear that you will break something that works. well, some of you will know the feeling. My friends here are rsync (with backup option) and svn. Before start and after changing any configuration I do an rsync copy to my backup server. It takes only seconds and it keeps in a separated folder any file changed in the process, so I can recover the previous settings in just a few seconds using a single command. Additionally a cron job takes my machines' /etc folders and commit them an svn server so I can even recover an old version of a config file easily. -- Andres Jimenez GPG : http://www.andresin.com/gpg/[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] [URGENT] Zap channels fail to load
On Wed, Feb 27, 2008 at 12:01 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: I had a theory on how this had happened in the specific case. But so far the OP has not confirmed or denied it. I did deny it. Please see it here: http://lists.digium.com/pipermail/asterisk-users/2008-February/206516.html You were right about the problem, but not about the cause. -- Andres Jimenez GPG : http://www.andresin.com/gpg/[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] [URGENT] Zap channels fail to load
On Wed, Feb 27, 2008 at 12:45 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: Are the new zaptel drivers loaded? cat /sys/module/zaptel/version Yes. I did fix it upgrading it back (and rebuilding asterisk and libpri) to zaptel 1.4.8 -- Andres Jimenez GPG : http://www.andresin.com/gpg/[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
[asterisk-users] About faxes recived through a PRI and passed to a fax machine connected to a FXS port
Hi, all I want to configure a few FXS ports in an Antribank-16 to be able to receive faxes sent throught a PRI: E1 ==Zap * ==FXS * ==Fax machine My asterisk box has a Digium TE120P (for the PRI). Versions are *= 1.4.17 | Zaptel=1.4.8 | libpri=1.4.5 The Astribank is not configured yet, because I am a little bit confused about how to do it. Let's say I configure the FXS ports in the Astribank as channels 41 42 (only 2 for the moment). I dedicated DIDs 11 22 for internal faxes. Can I just set the dial plan for : (not actual coonfig) exten= 11,1,Dial(Zap/41) exten= 22,1,Dial(Zap/42) Do I need any other piece of software? I know 1.6-beta is capable of managing faxes properly, but I won't upgrade my * if any other option is available. Thank you. -- Andres Jimenez GPG : http://www.andresin.com/gpg/[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] About faxes recived through a PRI and passed to a fax machine connected to a FXS port
On Wed, Feb 27, 2008 at 6:58 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: Let's say I configure the FXS ports in the Astribank as channels 41 42 (only 2 for the moment). Hmmm... One full E1 span will get you channels 1-31 (even if you use only up to channel 24, the span will register 31 channels). So the FXS ports of the Astribank will occupy ports 32-39 and 46-53 (40-45 are the I/O ports). But then again, normally you just use zapconf / genzaptelconf and get a working configuration. So I suspect you got the channel numbers wrong there. I used those numbers as an example, but I though I could relocate the Astribank channels to a higher number and have a cleaner setup. I don't like using automatic tools, because they used the american defaults. Should I just use zapconf/genzaptelconf and change the zones? Thanks -- Andres Jimenez GPG : http://www.andresin.com/gpg/[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
[asterisk-users] [URGENT] Zap channels fail to load
I have spent some time this morning trying to add an Astribank to our current Asterisk, but it failed, so I just removed the hardware, restore the config files to the original setup and started asterisk.; I could see that no Zap channels are started so I did load chan_zap.so: pbx*CLI module load chan_zap.so [Feb 26 10:32:18] WARNING[3809]: pbx.c:2968 ast_register_application: Already have an application 'ZapSendKeypadFacility' == Parsing '/etc/asterisk/zapata.conf': Found [Feb 26 10:32:18] WARNING[3809]: chan_zap.c:904 zt_open: Unable to specify channel 1: Device or resource busy [Feb 26 10:32:18] ERROR[3809]: chan_zap.c:7186 mkintf: Unable to open channel 1: Device or resource busy here = 0, tmp-channel = 1, channel = 1 [Feb 26 10:32:18] ERROR[3809]: chan_zap.c:10527 build_channels: Unable to register channel '1-15' Fair enough. I did unloaded chan_zap.so (because of the first error) and tried again: pbx*CLI module load chan_zap.so [Feb 26 10:32:18] WARNING[3809]: pbx.c:2968 ast_register_application: Already have an application 'ZapSendKeypadFacility' == Parsing '/etc/asterisk/zapata.conf': Found [Feb 26 10:32:18] WARNING[3809]: chan_zap.c:904 zt_open: Unable to specify channel 1: Device or resource busy [Feb 26 10:32:18] ERROR[3809]: chan_zap.c:7186 mkintf: Unable to open channel 1: Device or resource busy here = 0, tmp-channel = 1, channel = 1 [Feb 26 10:32:18] ERROR[3809]: chan_zap.c:10527 build_channels: Unable to register channel '1-15' It looks like the problem is in the zap card's first channel: pbx:~# cat /proc/zaptel/1 Span 1: WCT1/0 Wildcard TE12xP Card 0 HDB3/CCS IRQ misses: 36 1 WCT1/0/1 Clear (In use) 2 WCT1/0/2 Clear 3 WCT1/0/3 Clear 4 WCT1/0/4 Clear 5 WCT1/0/5 Clear 6 WCT1/0/6 Clear 7 WCT1/0/7 Clear 8 WCT1/0/8 Clear 9 WCT1/0/9 Clear 10 WCT1/0/10 Clear 11 WCT1/0/11 Clear 12 WCT1/0/12 Clear 13 WCT1/0/13 Clear 14 WCT1/0/14 Clear 15 WCT1/0/15 Clear 16 WCT1/0/16 HDLCFCS 17 WCT1/0/17 Clear 18 WCT1/0/18 Clear 19 WCT1/0/19 Clear 20 WCT1/0/20 Clear 21 WCT1/0/21 Clear 22 WCT1/0/22 Clear 23 WCT1/0/23 Clear 24 WCT1/0/24 Clear 25 WCT1/0/25 26 WCT1/0/26 27 WCT1/0/27 28 WCT1/0/28 29 WCT1/0/29 30 WCT1/0/30 31 WCT1/0/31 Is there any change any Astribank related stuff can be causing this? I have ensured that no Astribanks modules are loaded and even rebooted the box, but no success. pbx:~# cat /etc/zaptel.conf # CRC off # loadzone = uk defaultzone = uk span=1,1,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-24 #or this with crc on # #loadzone = uk #defaultzone = uk #span=1,1,0,ccs,hdb3,crc4 #bchan=1-15 #dchan=16 #bchan=17-24 pbx:~# cat /etc/asterisk/zapata.conf language=en internationalprefix = 00 nationalprefix = 0 switchtype = euroisdn pridialplan = local priindication = outofband usecallerid = yes hidecallerid = no callwaiting = yes usecallingpres = yes callwaitingcallerid = yes threewaycalling = yes transfer = yes cancallforward = yes callreturn = yes group = 1 callgroup = 0 pickupgroup = 0 immediate = no echotraining = yes echocancel = yes echocancelwhenbridged = no facilityenable = yes musiconhold = default ;overlapdial = yes overlapdial = no immediate = no txgain = -4.0 rxgain = -4.0 signalling = pri_cpe channel = 1-15 ;channel = 17-32 channel = 17-24 ;toneduration=100 toneduration=300 ;relaxdtmf=yes Thanks, -- Andres Jimenez GPG : http://www.andresin.com/gpg/[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] [URGENT] Zap channels fail to load
I forgot to mentio asterisk log this 2 errors: [Feb 26 08:38:01] ERROR[30245] chan_zap.c: Unable to get parameters [Feb 26 08:38:01] ERROR[30245] chan_zap.c: Unable to register channel '1-15' Any hint? Thanks in advance. Andres On Tue, Feb 26, 2008 at 10:44 AM, Andres Jimenez [EMAIL PROTECTED] wrote: I have spent some time this morning trying to add an Astribank to our current Asterisk, but it failed, so I just removed the hardware, restore the config files to the original setup and started asterisk.; I could see that no Zap channels are started so I did load chan_zap.so: pbx*CLI module load chan_zap.so [Feb 26 10:32:18] WARNING[3809]: pbx.c:2968 ast_register_application: Already have an application 'ZapSendKeypadFacility' == Parsing '/etc/asterisk/zapata.conf': Found [Feb 26 10:32:18] WARNING[3809]: chan_zap.c:904 zt_open: Unable to specify channel 1: Device or resource busy [Feb 26 10:32:18] ERROR[3809]: chan_zap.c:7186 mkintf: Unable to open channel 1: Device or resource busy here = 0, tmp-channel = 1, channel = 1 [Feb 26 10:32:18] ERROR[3809]: chan_zap.c:10527 build_channels: Unable to register channel '1-15' Fair enough. I did unloaded chan_zap.so (because of the first error) and tried again: pbx*CLI module load chan_zap.so [Feb 26 10:32:18] WARNING[3809]: pbx.c:2968 ast_register_application: Already have an application 'ZapSendKeypadFacility' == Parsing '/etc/asterisk/zapata.conf': Found [Feb 26 10:32:18] WARNING[3809]: chan_zap.c:904 zt_open: Unable to specify channel 1: Device or resource busy [Feb 26 10:32:18] ERROR[3809]: chan_zap.c:7186 mkintf: Unable to open channel 1: Device or resource busy here = 0, tmp-channel = 1, channel = 1 [Feb 26 10:32:18] ERROR[3809]: chan_zap.c:10527 build_channels: Unable to register channel '1-15' It looks like the problem is in the zap card's first channel: pbx:~# cat /proc/zaptel/1 Span 1: WCT1/0 Wildcard TE12xP Card 0 HDB3/CCS IRQ misses: 36 1 WCT1/0/1 Clear (In use) 2 WCT1/0/2 Clear 3 WCT1/0/3 Clear 4 WCT1/0/4 Clear 5 WCT1/0/5 Clear 6 WCT1/0/6 Clear 7 WCT1/0/7 Clear 8 WCT1/0/8 Clear 9 WCT1/0/9 Clear 10 WCT1/0/10 Clear 11 WCT1/0/11 Clear 12 WCT1/0/12 Clear 13 WCT1/0/13 Clear 14 WCT1/0/14 Clear 15 WCT1/0/15 Clear 16 WCT1/0/16 HDLCFCS 17 WCT1/0/17 Clear 18 WCT1/0/18 Clear 19 WCT1/0/19 Clear 20 WCT1/0/20 Clear 21 WCT1/0/21 Clear 22 WCT1/0/22 Clear 23 WCT1/0/23 Clear 24 WCT1/0/24 Clear 25 WCT1/0/25 26 WCT1/0/26 27 WCT1/0/27 28 WCT1/0/28 29 WCT1/0/29 30 WCT1/0/30 31 WCT1/0/31 Is there any change any Astribank related stuff can be causing this? I have ensured that no Astribanks modules are loaded and even rebooted the box, but no success. pbx:~# cat /etc/zaptel.conf # CRC off # loadzone = uk defaultzone = uk span=1,1,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-24 #or this with crc on # #loadzone = uk #defaultzone = uk #span=1,1,0,ccs,hdb3,crc4 #bchan=1-15 #dchan=16 #bchan=17-24 pbx:~# cat /etc/asterisk/zapata.conf language=en internationalprefix = 00 nationalprefix = 0 switchtype = euroisdn pridialplan = local priindication = outofband usecallerid = yes hidecallerid = no callwaiting = yes usecallingpres = yes callwaitingcallerid = yes threewaycalling = yes transfer = yes cancallforward = yes callreturn = yes group = 1 callgroup = 0 pickupgroup = 0 immediate = no echotraining = yes echocancel = yes echocancelwhenbridged = no facilityenable = yes musiconhold = default ;overlapdial = yes overlapdial = no immediate = no txgain = -4.0 rxgain = -4.0 signalling = pri_cpe channel = 1-15 ;channel = 17-32 channel = 17-24 ;toneduration=100 toneduration=300 ;relaxdtmf=yes Thanks, -- Andres Jimenez GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED] -- Andres Jimenez GPG : http://www.andresin.com/gpg/[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] [URGENT] Zap channels fail to load
On Tue, Feb 26, 2008 at 12:21 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: Note: [Urgent] is generally not a good way to escalate the issue on a public mailing list. We're all here for the fun of it and demanding prompt reply may actually serve the other way. I am sorry about the scalating, but I was panicking a little bit after a couple of hours trying to fix the issue. If you have paid to get support (e.g: by buying hardware), this may be a good time to use it. I am too used to get/give free advice that I forget I can use it. Contact me privately :-) Nice to meet an Astribank guru . I will contact you soon :-) The symptom is expelained in the following report that I filed earlier today (unrelated to this one) http://bugs.digium.com/12071 Channel 1 was left open from a failed configuration attempt So the real error hides earlier in your logs. Look for 'chan_zap' in the logs from the startup of Asterisk. And sadly you must restart Asterisk to fix the error. I din't fix it for me. Is this a fractional E1, indeed? Not sure about how to call it. The provider (Eircom, Ireland) calls it PRA. This messages follows in the other message reply. -- Andres Jimenez GPG : http://www.andresin.com/gpg/[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] [URGENT] Zap channels fail to load
Comes from a previous message. On Tue, Feb 26, 2008 at 12:25 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: Here's my guess: You built Asterisk vs. a newer Zaptel (that happened to have the Astribank drivers). Now you reverted to the old Zaptel drivers. And those are of a version before 1.4.8 . Hence the new ZT_GET_PARAMS of 1.4.8 does not exist there. The ZT_GET_PARAMS ioctl Asterisk sends is thus not understood by Zaptel and fails. Unrevert to the new Zaptel version (of the modules. Stick with the original zaptel.conf). Does this help? I did build asterisk using zaptel 1.4.8 about 10 days ago. We were having issues with DTMF , so I downgraded zaptel (rebuilding Asterisk and libpri, of course) to 1.4.7, but the problems remained. Anything else worked just fine, so I kept researching and no other changes were done till today. Zaptel 1.4.7 was working perfectly for almost 2 weeks and has the Astribanks drivers too, so I just tried to add the Astribank to my configuration today. I wasn't being able to make it work, so I reverted the changes made earlier and disconnected the Astribank trying to go back to the previous known working config. I have rebuild everything against Zaptel 1.4.8 and it works now, but I am a little bit concerned about why asterisk tried to use the functions belonging to zaptel 1.4.8. -- Andres Jimenez GPG : http://www.andresin.com/gpg/[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] [URGENT] Zap channels fail to load
On Tue, Feb 26, 2008 at 12:05 PM, Louwrens Benadé [EMAIL PROTECTED] wrote: What's your output from 'ztcfg -vv'? pbx:~# ztcfg -vv Zaptel Version: 1.4.8 Echo Canceller: MG2 Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: Clear channel (Default) (Slaves: 03) Channel 04: Clear channel (Default) (Slaves: 04) Channel 05: Clear channel (Default) (Slaves: 05) Channel 06: Clear channel (Default) (Slaves: 06) Channel 07: Clear channel (Default) (Slaves: 07) Channel 08: Clear channel (Default) (Slaves: 08) Channel 09: Clear channel (Default) (Slaves: 09) Channel 10: Clear channel (Default) (Slaves: 10) Channel 11: Clear channel (Default) (Slaves: 11) Channel 12: Clear channel (Default) (Slaves: 12) Channel 13: Clear channel (Default) (Slaves: 13) Channel 14: Clear channel (Default) (Slaves: 14) Channel 15: Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Clear channel (Default) (Slaves: 17) Channel 18: Clear channel (Default) (Slaves: 18) Channel 19: Clear channel (Default) (Slaves: 19) Channel 20: Clear channel (Default) (Slaves: 20) Channel 21: Clear channel (Default) (Slaves: 21) Channel 22: Clear channel (Default) (Slaves: 22) Channel 23: Clear channel (Default) (Slaves: 23) Channel 24: Clear channel (Default) (Slaves: 24) 24 channels to configure. -- Andres Jimenez GPG : http://www.andresin.com/gpg/[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] [URGENT] Zap channels fail to load
On Tue, Feb 26, 2008 at 1:35 PM, Louwrens Benadé [EMAIL PROTECTED] wrote: Why does this look suspiciously like a T1 line? Are you sure this is a fractional E1? My provider names the line a PRA, but this is understood anywhere as a PRI (no fractional). From the Asterisk configuration point of view, is there any other difference between the fractional and full PRI apart from the number of channels? Do you guys know of any particular setting our provider (Eircom, Ireland) could require? I have a problem with DTMF when the PRI is the one carrying the call: 1s and 2s are not transmitted. If the call is internal or carryed by and IAX trunk + SIP it works nicely. Regards, -- Andres Jimenez GPG : http://www.andresin.com/gpg/[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] load balancing SIP extensions
On Fri, Feb 22, 2008 at 11:42 AM, Vieri [EMAIL PROTECTED] wrote: However, say ext. 4001 is registered on *1 and 4002 is registered on *2, if 4001 tries to call 4002 then I would like to do something like: - lookup 4002 on *1, try to establish a call if it's REGISTERED here - if it's not registered here then try to look it up on *2 and establish the call there You can do that using the dial plan. - Create an IAX link between both servers - DIal plan in both servers: First priority Dial using SIP/EXTEN Second priority IAX/EXTEN Dial IAX/EXTEN -- Andres Jimenez GPG : http://www.andresin.com/gpg/[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] load balancing SIP extensions
On Fri, Feb 22, 2008 at 5:49 PM, Vieri [EMAIL PROTECTED] wrote: --- Andres Jimenez [EMAIL PROTECTED] wrote: On Fri, Feb 22, 2008 at 11:42 AM, Vieri [EMAIL PROTECTED] wrote: However, say ext. 4001 is registered on *1 and 4002 is registered on *2, if 4001 tries to call 4002 then I would like to do something like: - lookup 4002 on *1, try to establish a call if it's REGISTERED here - if it's not registered here then try to look it up on *2 and establish the call there You can do that using the dial plan. - Create an IAX link between both servers - DIal plan in both servers: First priority Dial using SIP/EXTEN Second priority IAX/EXTEN Dial IAX/EXTEN Thanks. I'll try that although I hope it won't go into an infinite loop between the 2 servers. Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- 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 -- Andres Jimenez GPG : http://www.andresin.com/gpg/[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] load balancing SIP extensions
On Fri, Feb 22, 2008 at 5:49 PM, Vieri [EMAIL PROTECTED] wrote: Thanks. I'll try that although I hope it won't go into an infinite loop between the 2 servers. You are right. That could happen if the phone is not registered anywhere You can put some security in the dialplan. if calls comes from IAX it means that PHONE is not registered in the other server. Just create special extensions to take the IAX calls (instead of GoTo): PHONE is 101 SERVER 1 exten = 101,1, Dial SIP/101 exten = 101,1, Dial IAX-SERVER2/55101 exten = 55101,1, Dial SIP/101 exten = 55101,1, Hangup SERVER 2 exten = 101,1, Dial SIP/101 exten = 101,1, Dial IAX-SERVER1/55101 exten = 55101,1, Dial SIP/101 exten = 55101,1, Hangup I hope it helps, -- Andres Jimenez GPG : http://www.andresin.com/gpg/[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] GXP-2020 Transfer Key
Sorry s/r/t/ :-) Are you allowing calls to be transfered? (t option in Dial command) On Wed, Feb 20, 2008 at 1:20 PM, Andres Jimenez [EMAIL PROTECTED] wrote: Are you allowing calls to be transfered? (r option in Dial command) On Wed, Feb 20, 2008 at 1:50 PM, Gustavo Gonzalez [EMAIL PROTECTED] wrote: Hello! is there a way to get keep working the TRANSFER key of GXP-2020 with asterisk?. Attended and blind transfer does not work wiith this IP Phone Alejandro González Grupo Gestión 4384-0660 www.grupo-gestion.com.ar [EMAIL PROTECTED] --- --- RI 9000-1069 Sistema de Gestión de Calidad Certificado por IRAM Norma ISO: 9001-2000 ___ -- 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 -- Andres Jimenez GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED] -- Andres Jimenez GPG : http://www.andresin.com/gpg/[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] GXP-2020 Transfer Key
Are you allowing calls to be transfered? (r option in Dial command) On Wed, Feb 20, 2008 at 1:50 PM, Gustavo Gonzalez [EMAIL PROTECTED] wrote: Hello! is there a way to get keep working the TRANSFER key of GXP-2020 with asterisk?. Attended and blind transfer does not work wiith this IP Phone Alejandro González Grupo Gestión 4384-0660 www.grupo-gestion.com.ar [EMAIL PROTECTED] --- --- RI 9000-1069 Sistema de Gestión de Calidad Certificado por IRAM Norma ISO: 9001-2000 ___ -- 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 -- Andres Jimenez GPG : http://www.andresin.com/gpg/[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] Problem with DTMF dialing
On Fri, Feb 15, 2008 at 9:05 AM, Andres Jimenez [EMAIL PROTECTED] wrote: On Wed, Feb 13, 2008 at 10:48 AM, Andres Jimenez [EMAIL PROTECTED] wrote: On Tue, Feb 12, 2008 at 10:03 PM, Andrew Joakimsen [EMAIL PROTECTED] wrote: Maybe it is related but with PRI Asterisk does not generate any tone it sends a signal regarding your keypress. If you are using SIP phones make sure the dtmfmode in use is RFC2833. I have just double check and my phones use DTMF in RFC2833 mode. I wil try to downgrade my zaptel later today CONFIRMED. The problem disappears after downgrading zaptel from 1.4.8 to 1.4.7 Please forgive me because I was wrong. After downgrading zaptel DMTF works much better, but for some reason numbers 1 2 are not send through DTMF. Every other key (including * #) work like a charm. DTMF works nicely in the LAN side (i. e. voicemail login) , but if I try to reach our voicemail from the outside I see any key pressed except 1 2. Telephone is Grandstream GXP-2000, but I think I should blame my * . I know we are having this problem when dialing through Zap channels (Digium TE120P card) Any hint? Cheers, -- Andres Jimenez GPG : http://www.andresin.com/gpg/[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] Problem with DTMF dialing
On Wed, Feb 13, 2008 at 10:48 AM, Andres Jimenez [EMAIL PROTECTED] wrote: On Tue, Feb 12, 2008 at 10:03 PM, Andrew Joakimsen [EMAIL PROTECTED] wrote: Maybe it is related but with PRI Asterisk does not generate any tone it sends a signal regarding your keypress. If you are using SIP phones make sure the dtmfmode in use is RFC2833. I have just double check and my phones use DTMF in RFC2833 mode. I wil try to downgrade my zaptel later today CONFIRMED. The problem disappears after downgrading zaptel from 1.4.8 to 1.4.7 -- Andres Jimenez GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED] -- Andres Jimenez GPG : http://www.andresin.com/gpg/[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] Problem with DTMF dialing
On Tue, Feb 12, 2008 at 10:03 PM, Andrew Joakimsen [EMAIL PROTECTED] wrote: Maybe it is related but with PRI Asterisk does not generate any tone it sends a signal regarding your keypress. If you are using SIP phones make sure the dtmfmode in use is RFC2833. I have just double check and my phones use DTMF in RFC2833 mode. I wil try to downgrade my zaptel later today -- Andres Jimenez GPG : http://www.andresin.com/gpg/[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] Problem with DTMF dialing
I am having similar problems running the same versions of Asterisk, libpri zaptel. The Asterisk bug (http://bugs.digium.com/view.php?id=11855) was supossed to be related to FXO only, but I am having issues with a PRI line and Digium's TE120P. Do you guys think it can be the same issue? -- Andres Jimenez GPG : http://www.andresin.com/gpg/[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] Load Balancing over 2 E1 Lines
On Dec 12, 2007 8:08 AM, Eric Delaporte [EMAIL PROTECTED] wrote: I read something about DIAL(Zap/r1/…) for using round robin, and it seems to work. That will give you the same number of calls routed to each line Is there any other possible way to make sure that all lines are used in the same amount of minutes? You are going to need an AGI app or something storing how many minutes have been routed through each line and, on every call, choosing the less used one as the line to go out. -- Andres Jimenez GPG : http://www.andresin.com/gpg/[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
[asterisk-users] Installing/configuring TE120P debian way
Hi all I use asterisk (1.2 brach) from debian official packages and it works fine. Now I need to install and configure a Digium TE120P card, but I cannot find any guide to install it using debian packages. I would like to know if anyone of you knows about packages that would include the necessary kernel modules or any other method that won't be broken when the asterisk packages are updated. Would anyone consider just install everything from source (branch 1.4) as the best option? I would like to keep an easy upgradeable system like Debian packages, but could use source code if necessary. Cheers, -- Andres Jimenez GPG : http://www.andresin.com/gpg/[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] Best way to detect unknown and/or private incoming caller-id?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 2007/8/18, voiplist : Is there a better way to catch calls which are purposely blocked by the calling party? Sometimes they come through as 000-000- and as I recall sometimes just blank or unknown. The problem here is How can you be sure the calling PERSON is purposely blocking its own CALLERID? I don think you shouldn't be punishing, for example, SkypeOut users or people using dodgy carriers. If your Playback says you don't accept any anonymous call at least they would be able to change it. - -- Andres Jimenez GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED] -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: http://firegpg.tuxfamily.org iD8DBQFGx0me8SZxpGYWwpYRAoR7AKCdZGX8//GfdPCZovRuQN87hQh90QCdHgBl 5tTHq8WRiTjum3GIEwgkeAs= =tFmP -END PGP SIGNATURE- ___ --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] Lock extension from asterisk
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi all I am working in a new set up with Grandstream GXP-2000 handsets. I like those phone, but they lack a feature I need: the phone cannot be locked by the user. What I actually want is a user to be able to avoid someone else making calls from his phone without giving him access to SIP configuration access to the phone. i.e. let say I want user 132 (that uses extension 132 in our system) to be able to lock his phone (located in a publicly accessible office). Could he dial an special extension (i.e. ) and Asterisk will drop any call until another special extension (i.e. ) is dialed? Suggestions? - -- Andres Jimenez -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.3 (GNU/Linux) Comment: http://firegpg.tuxfamily.org iD8DBQFGxX6A8SZxpGYWwpYRAtojAJ4yKE77nv9rpkoXXr1i4SOiLPb7JACgug+7 64yg8fCDRdnmeZFmmpGynwQ= =eCWP -END PGP SIGNATURE- ___ --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] Lock extension from asterisk
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Hash: SHA1
2007/8/17, Gordon Henderson :
S (all untested!)
exten = ,1,Answer()
exten = ,n,Set(me=${CALLERID(num)})
exten = ,n,Set(DB(${me}/locked)=1)
exten = ,1,Answer()
exten = ,n,Set(me=${CALLERID(num)})
exten = ,n,VMAuthenticate(${me})
exten = ,n,Set(DB(${me}/locked)=)
(I think I swapped the and here, but I'm sure you can see that!)
and in the dial-plan where call processing takes place:
exten = s,1,Set(me=${CALLERID(num)})
exten = s,n,Set(locked=${DB(${me}/locked)})
exten = s,n,GotoIf(${locked}?:doneLockCheck)
exten = s,n,Playback(sorry-cant-let-you-do-that)
exten = s,n,Hangup()
exten = s,n(doneLockCheck),Noop(We're not locked)
Works like a charm. Thanks very much.
- --
Andres Jimenez
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W3QsbPV53glyOdxaFVNnFrw=
=U7Ab
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Re: [asterisk-users] Lock extension from asterisk
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Thanks to Gordon Doug I have now a very good locking system using
one only extension.
Extension informs you about the current lock situation and, if
authentacated, it changes it and explain the change done.
;Locking system
;LOCK
exten = ,1,Answer()
exten = ,n,Set(me=${CALLERID(num)})
exten = ,n,GotoIf(${DB(${me}/locked)}?,101:,201)
exten = ,101,Playback(security)
exten = ,n,Playback(activated)
exten = ,n,VMAuthenticate(${me})
exten = ,n,Set(DB(${me}/locked)=)
exten = ,n,Playback(security)
exten = ,n,Playback(now)
exten = ,n,Playback(de-activated)
exten = ,n,Hangup()
exten = ,201,Playback(security)
exten = ,n,Playback(de-activated)
exten = ,n,VMAuthenticate(${me})
exten = ,n,Set(DB(${me}/locked)=1)
exten = ,n,Playback(security)
exten = ,n,Playback(now)
exten = ,n,Playback(activated)
exten = ,n,Hangup()
Thanks again, guys
- --
Andres Jimenez
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Re: [asterisk-users] Lock extension from asterisk
2007/8/17, Andres Paglayan [EMAIL PROTECTED]:
Guys, very nice dialplan programming,
as a user's opinion, the two extension approach might be better.
so the user doesn't need to remember whether the phone is locked or not,
and accidentally lock it when the contrary was meant,
(unless you send some sip text to the phone display)
In the latest version (see below) I added some playback that will say
if the phone is lock or unlock, before and after locking/unlocking it.
;Locking system
;LOCK
exten = ,1,Answer()
exten = ,n,Set(me=${CALLERID(num)})
exten = ,n,GotoIf(${DB(${me}/locked)}?,101:,201)
exten = ,101,Playback(security)
exten = ,n,Playback(activated)
exten = ,n,VMAuthenticate(${me})
exten = ,n,Set(DB(${me}/locked)=)
exten = ,n,Playback(security)
exten = ,n,Playback(now)
exten = ,n,Playback(de-activated)
exten = ,n,Hangup()
exten = ,201,Playback(security)
exten = ,n,Playback(de-activated)
exten = ,n,VMAuthenticate(${me})
exten = ,n,Set(DB(${me}/locked)=1)
exten = ,n,Playback(security)
exten = ,n,Playback(now)
exten = ,n,Playback(activated)
exten = ,n,Hangup()
--
Andres Jimenez
GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED]
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Re: [asterisk-users] Lock extension from asterisk
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 2007/8/17, Doug Lytle : Just a note, You will want to make sure that (911/999) calls are handled properly when the phone is locked down. Good point. - -- Andres Jimenez GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED] -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: http://firegpg.tuxfamily.org iD8DBQFGxgMD8SZxpGYWwpYRAvXHAJ9ZsPF6wzEaEn6y/VDfgxvuJdmXkgCfYxrz ZEEXnAqeELULlSqJxqmdaJw= =NvEA -END PGP SIGNATURE- ___ --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] Introducing myself
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, all First post to a new (for me!)list. Netiquette as a must. My name is Andres Jimenez and I am an spaniard working as System Administrator in Dublin (Ireland). I just started working with Asterisk, but thanks to all the available documentation the community has created I an being able to get over any problem in my VoIP setup. I want to thank you all for your previous and future help. - -- Andres Jimenez -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.3 (GNU/Linux) Comment: http://firegpg.tuxfamily.org iD8DBQFGxCoP8SZxpGYWwpYRAjaVAKDdJ/8H0O7Cx/hLmwDI/7XQARag+gCg1uGN f1bzk8UGM97F+4Elciip7og= =j6ET -END PGP SIGNATURE- ___ --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] About cards for ISDN-PRI in Ireland
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi all I have already a fully operational Asterisk PBX connected only to the IP world. As a necessary step before smashing our current PBX, I need to install and configure an ISDN card able to take an incoming E1 line provided by Eircom. That's the one and only ISDN line we will ever have, so I am planning to get a single port card similar to Digium's Wildcard TE120P [1] . That card seems to be pretty new (no page yet in Voip-info.org and just 2 results in a search), but looks as just a new version of TE110P [2] Have anyone had problems with newest card? Should I go straight to the well known old version? Can you suggest any alternatives? I meant something cheaper or better with a similar price. That one is for the Irish fellows. Which card are you using for similar setups in Ireland (Dublin4)? Is there any card that won't play nice with Eircom? Cheers, [1] http://www.digium.com/en/products/hardware/te120p.php [2] http://www.voip-info.org/wiki/index.php?page=Digium%20Wildcard%20TE110P - -- Andres Jimenez -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.3 (GNU/Linux) Comment: http://firegpg.tuxfamily.org iD8DBQFGxC6E8SZxpGYWwpYRAsv7AJ9Ojo0hY+ZoHzCTpcLHVc4CoaS8ngCfR+uh ovBJoFi262l/Xqfc+mVuBgw= =KVu0 -END PGP SIGNATURE- ___ --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
