RE : [asterisk-users] Linux distro + Asterisk or Trixbox?
Hi men, Have a look at : www.asterisknow.org This will be THE standard ! Best Regards, Francois BERGERET, France. ___ --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] Learning Asterisk Internals
Hi everyone, I would like to start learning Asterisk internals. Where should I start before getting directly to the source code? Is there any sort of Asterisk internal for newbies? I know there is a project similar to this at asteriskdocs.org, but it's still in development. I would appreciate any advices. Best regards, Mike. ___ --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] call forward
Dear Please I have a problem since an extension has a call forward service assigned to international number ,if any party is calling the forwarded extension ,the party calling will charge the call . How can I let the forwarded extension party be charged Thanks * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. *___ --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] call forward
Dear Please I have a problem since an extension has a call forward service assigned to international number ,if any party is calling the forwarded extension ,the party calling will charge the call . How can I let the forwarded extension party be charged ,in other words how can I let be the calling source be the extension that have the forward service. Thanks * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. *___ --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] AGI and php simple example
Hi can you give a simple example of php and AGI script? I've read http://www.voip-info.org/wiki-Asterisk+AGI+php but i can't understand how to play sounds and read DTMF digits... For example, if i will write the following script directly from php: - play a welcome message - play a recorded message (For example: Digit a number:) - read the DTMF digits - if the number digited is greater than 5 - play a recorded message (You have digit a number greater than 5) else - play a recorded message (You have digit a number smaller than 5) - hangup Is it possible to write a php script that do these things? How? Thanks ___ --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] call forward
Looking at my cdr each leg of the call has its own CDR record and they both have correct accountcodes. Perhaps you are trying to bill by caller id, in which case what you are asking is not possible. On 12/17/06, Khaled [EMAIL PROTECTED] wrote: Dear Please I have a problem since an extension has a call forward service assigned to international number ,if any party is calling the forwarded extension ,the party calling will charge the call . How can I let the forwarded extension party be charged ,in other words how can I let be the calling source be the extension that have the forward service. Thanks -- * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is it possible to use Asterisk voicemail as anouncement system only?
On 12/15/06, Michael Hamann [EMAIL PROTECTED] wrote: Hello, we are using asterisk in combination with the voicemail system. I´m just wondering if it is possible to switch the voicemail to an I am on holiday mode. This means that the unavailable message is played to the caller but no possability to record a message. So far I did not find an option in the voicemail.conf for this. Any ideas except creating my own ivr menu ? best regards Michael What happens if you use the maxmsg variable in voicemail.conf and set it to 0 or 1? Don't know if there's a minimum limit, the max is I think and default is 100. Maybe there's a built-in vacation mode feature but I don't know about it. HTH \R ___ --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] call forward=agi
Please Can I have the value of $k which is a variable at diaparties.agi in extensipns.conf . Thanks _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Joakimsen Sent: Sunday, December 17, 2006 3:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] call forward Looking at my cdr each leg of the call has its own CDR record and they both have correct accountcodes. Perhaps you are trying to bill by caller id, in which case what you are asking is not possible. On 12/17/06, Khaled [EMAIL PROTECTED] wrote: Dear Please I have a problem since an extension has a call forward service assigned to international number ,if any party is calling the forwarded extension ,the party calling will charge the call . How can I let the forwarded extension party be charged ,in other words how can I let be the calling source be the extension that have the forward service. Thanks _ * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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 http://lists.digium.com/mailman/listinfo/asterisk-users * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. *___ --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] Day/night service and indications on the phone
Hello everybody. I have created an extension that enables/disables the night service mode on asterisk (i.e. voicemail is started on incoming calls, instead of entering a queue). The night mode is activated/deactivated by the front desk operator when the office is about to close, by pressing a line key on the phone, which dials the related extension. Does anyone know a way to have an indication on the operator's phone using BLF-style hints or similar? I'd like to have a LED turned on when night mode is active (most traditional PBXes offer this feature). Is there a way, for instance, to force the device state for a dialplan hint, e.g. on a fake or local channel, so that I can map a BLF key on the phone to that hint? I have not find anything suitable so far (except for a PC system tray's icon, which is not applicable to my situation). Thanks, Alberto. ___ --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] AGI and php simple example
I've read http://www.voip-info.org/wiki-Asterisk+AGI+php but i can't understand how to play sounds and read DTMF digits... Have a look at this : http://phpagi.sourceforge.net/ hth ___ --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] Day/night service and indications on the phone
[EMAIL PROTECTED] wrote: Is there a way, for instance, to force the device state for a dialplan hint, e.g. on a fake or local channel, so that I can map a BLF key on the phone to that hint? [turn on mwi] touch /var/spool/asterisk/voicemail/context/device/msg0001.txt [turn off mwi] rm /var/spool/asterisk/voicemail/context/device/msg0001.txt -f Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Day/night service and indications on the phone
On Dec 17, 2006, at 3:26 PM, Doug Lytle wrote: [EMAIL PROTECTED] wrote: Is there a way, for instance, to force the device state for a dialplan hint, e.g. on a fake or local channel, so that I can map a BLF key on the phone to that hint? [turn on mwi] touch /var/spool/asterisk/voicemail/context/device/msg0001.txt [turn off mwi] rm /var/spool/asterisk/voicemail/context/device/msg0001.txt -f You can also use the devicestate commands in BRIstuffed asterisk. Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? PGP.sig Description: This is a digitally signed message part ___ --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] Day/night service and indications on the phone
On Dom, Dicembre 17, 2006 15:26, Doug Lytle wrote: [EMAIL PROTECTED] wrote: Is there a way, for instance, to force the device state for a dialplan hint, e.g. on a fake or local channel, so that I can map a BLF key on the phone to that hint? [turn on mwi] touch /var/spool/asterisk/voicemail/context/device/msg0001.txt [turn off mwi] rm /var/spool/asterisk/voicemail/context/device/msg0001.txt -f Doug Thanks for the tip. But doesn't that conflict with the real message waiting indication? (The phone extension has its own voice mailbox). ___ --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] Day/night service and indications on the phone
On 12/17/06, Michiel van Baak [EMAIL PROTECTED] wrote: You can also use the devicestate commands in BRIstuffed asterisk. That's what I use. ___ --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] Day/night service and indications on the phone
On Dom, Dicembre 17, 2006 15:56, Michiel van Baak wrote: You can also use the devicestate commands in BRIstuffed asterisk. Michiel van Baak Thanks, this looks like what I need, although I'd better not to bristuff any of my asterisk boxes. I'll try to play with app_devstate.c alone (maybe it'll compile outside bristuff, without the need to patch the whole source). Alberto. ___ --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] Small form factor system w/PCI slot
I've been using a Compaq Deskpro EN SFF. They're small, have 3 pci slots, and you can get them up to 1Ghz PIII w/ 512MB of ram on ebay for under $100. Great for testing. When you're done with it throw in a PCI-PCMCIA adapter and turn it into a wireless AP or throw in a network card. They make a great router. You can't beat the price. Corporations bought them by the hundreds and they're all coming off of corporate lease. A TDM400 fits just right. If you want an even smaller package the Compaq Deskpro EN ultra SFF has 2 full size pci slots. --Zach On Fri, 2006-06-09 at 09:34 +0200, Jens Vagelpohl wrote: On 9 Jun 2006, at 02:04, Leo Ann Boon wrote: Jens Vagelpohl wrote: Hi everyone, I'm trying to buy a small form-factor PC system for use with Asterisk and Hylafax in conjunction with a Eicon DIVA Server single-port ISDN card (needs full-size 5V PCI 2.2 slot, but PCI-X compatible). Use is very light - at most a single call at any one time. If the Mac Mini had a PCI slot I'd try to use that one, but oh well ;) You mean PCI-E? If you really need PCI-X, then you're out of luck. PCI-X is only available on server boards. For a single port ISDN, one of those Mini-ITX boxes should work. I built something similar using a Mini-ITX (1GHz CPU) with an AVM Fritz! PCI ISDN card using chan_capi. IIRC, Xorcom has a TS-1 which is a SFF Asterisk server for $500. BTW, I don't think the Mini-ITX mobos can support PCI-E. It's a normal 5.5 V PCI slot, the card can also deal with PCI-X slots as the documentation claims. I'll take a look at Xorcom's offerings, thanks. jens ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Day/night service and indications on the phone
[EMAIL PROTECTED] wrote: But doesn't that conflict with the real message waiting indication? (The phone extension has its own voice mailbox). Yes it would. Our operators don't have voicemail on their phones. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What web interfaces are available today for debian based Asterisk installation?
Hi list, It's been a while since I've done asterisk stuff, and I'm wondering if there any news in the field. What do you people use today for http management of debian based Asterisk setup? Preferably something with the proven .deb extension. Any recommendations are welcome. Thank you, Maxim. -- Cheers, Maxim Vexler Free as in Freedom - Do u GNU ? ___ --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] Day/night service and indications on the phone
On Dom, Dicembre 17, 2006 16:10, [EMAIL PROTECTED] wrote: On Dom, Dicembre 17, 2006 15:56, Michiel van Baak wrote: You can also use the devicestate commands in BRIstuffed asterisk. Michiel van Baak Thanks, this looks like what I need, although I'd better not to bristuff any of my asterisk boxes. I'll try to play with app_devstate.c alone (maybe it'll compile outside bristuff, without the need to patch the whole source). Alberto. I'm happy to report that with a very litte change to app_devstate.c (just in the way ast_device_state_changed_literal() is called) that module just compiles and works fine even without bristuffing anything. BTW I'm using a Thomson ST2030S phone with a status key subscribed to a DS/xxx hint. Thanks again for your precious help! ___ --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] What web interfaces are available today for debian based Asterisk installation?
On Sun, Dec 17, 2006 at 07:07:07PM +0200, Maxim Vexler wrote: Hi list, It's been a while since I've done asterisk stuff, and I'm wondering if there any news in the field. What do you people use today for http management of debian based Asterisk setup? Preferably something with the proven .deb extension. destar has a tar extension, but a prefix quite similar to Debian. Availble in Etch. FreePBX is not yet availble, though a Sarge-based CD that includes it and generally works could be downloaded from http://updates.xorcom.com/iso/rapid-current.iso and the package is availble from deb http://updates.xorcom.com/rapid future main -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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 1.4.0 B4 Sounds Directory
Samy Antoun wrote: I noticed that the sound directory is missing from asterisk-1.4.0-beta4.tar.gz. This is incorrect; the sounds directory is present and contains two files. This directory (7 M) witch existed in asterisk-1.4.0-beta3.tar.gz has GSM Core Sounds and some MOH. There was a packaging error when this tarball was created, and the sound/MOH file tarballs were not included. However, the 'make install' process will automatically download the sounds during installation anyway; adding them to the tarball just makes it slightly quicker to do the installation. ___ --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 1.4.0 B4 Sounds Directory
On Sun, Dec 17, 2006 at 12:35:41PM -0600, Kevin P. Fleming wrote: Samy Antoun wrote: I noticed that the sound directory is missing from asterisk-1.4.0-beta4.tar.gz. This is incorrect; the sounds directory is present and contains two files. This directory (7 M) witch existed in asterisk-1.4.0-beta3.tar.gz has GSM Core Sounds and some MOH. There was a packaging error when this tarball was created, and the sound/MOH file tarballs were not included. However, the 'make install' process will automatically download the sounds during installation anyway; adding them to the tarball just makes it slightly quicker to do the installation. Note that this breaks Debian package building. Can we assume that the released tarall will include the gsm sounds? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] GXP2000 and BLF
I am trying to set up the BLF on a GXP2000.
Currently what I have is
extesions.conf:
[globals]
polycom430=SIP/101
[internal]
;exten = 101,1,Dial(SIP/101,10,)
;exten = 101,2,VoiceMail([EMAIL PROTECTED])
;exten = 101,102,VoiceMail([EMAIL PROTECTED])
exten = 101,1,Macro(voicemail,${polycom430})
[macro-voicemail]
exten = s,1,Dial(${ARG1},10,tT)
exten = s,2,VoiceMail([EMAIL PROTECTED])
exten = s,102,VoiceMail([EMAIL PROTECTED])
[ext-local-custom]
exten = 101,hint,${polycom430}
sip.conf:
[general]
subscribecontext=ext-local-custom
And have set up the key to Asterisk BLF with UserID101
When I reload the phone, I get the following error:
[Dec 17 13:34:33] ERROR[2551]: chan_sip.c:14064 handle_request_subscribe:
Got SUBSCRIBE for extension [EMAIL PROTECTED] from 192.168.1.248, but there is
no
hint for that extension
Any help is greatly appreciated.
Chris
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Re: [asterisk-users] TDM2400
I am having 2 more issues, when starting asterisk I got the below message:
Dec 17 22:27:54 NOTICE[4554]: indications.c:505
ast_unregister_indication_country: Removed default indication country 'us'
Dec 17 22:27:54 WARNING[4554]: chan_zap.c:10874 setup_zap: Ignoring
signalling
Dec 17 22:27:54 WARNING[4554]: chan_zap.c:10874 setup_zap: Ignoring
answeronpolarityswitch
Dec 17 22:27:54 WARNING[4554]: chan_zap.c:10874 setup_zap: Ignoring
hanguponpolarityswitch
my setup is : softphone---softswitch(asterisk)Termination GW(asterisk
with TDM card)
when dialing from my softphone I got :
-- Executing Dial(SIP/10.8.0.6-0947d008, (Zap/g1/6159587)) in new
stack
Dec 17 23:04:16 WARNING[5052]: channel.c:2597 ast_request: No channel type
registered for '(Zap'
Dec 17 23:04:16 NOTICE[5052]: app_dial.c:1056 dial_exec_full: Unable to
create channel of type '(Zap' (cause 66 - Channel not implemented)
my extensions.conf file has:
[globals]
TRUNK=Zap/g1
[topstn]
exten=_2.,1,Dial,(${TRUNK}/${EXTEN:3})
Please help ...
Thanks,
On 12/12/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Mon, Dec 11, 2006 at 08:08:18PM -0500, Time Bandit wrote:
[channels]
context=default
signalling=fxs_ls
;channel=1-16
usecallerid=yes
hidecallerid=no
callwaiting=yes
restrictcid=no
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
;accountcode=lss0101
answeronpolarityswitch=yes
hanguponpolarityswitch=yes
To the best of my knowledge, all the settings you put after defining
the channles (channel= line) are useless. You have to set all the
settings BEFORE you define the channels.
Should be. However in practice after the first reload all of them will
be applied (in this specific case).
/me points again to genzaptelconf that should have made this thread
unnecessary.
--
Tzafrir Cohen
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] rxfax detection problems with multiple contexts
Thanks Andrew! I had it in separate contexts originally, but for some
reason it wasn't working with fax detection. I didn't realize I could
drop rxfax onto the s extension, for some reason I kept thinking it
was somehow connected with the fax extension only. Works beautifully
now, thanks a lot.
regards,
mayo
On Dec 16, 06, at 19:06 , Andrew Joakimsen wrote:
If you are using one line for fax only then you do not need to do
fax detect. Put it in its own context and make ths s extension be
Rxfax.
Everyone always tries to over-compliate stuff, and it seems to me
the less you know about asterisk the more elaborate and
overcomplicated schemes people can come up with.
On 12/16/06, Mayo Jordanov [EMAIL PROTECTED] wrote: Hello,
I have a rather odd problem with Asterisk detecting faxes. I have two
POTS lines coming into the box (TDM400P). Line 1 is for voice, Line 2
is fof fax. When I set them up with channel = 1-2 in zapata.conf ,
all is fine, but as soon as I have two channel = definitions,
Asterisk is unable to detect faxes. The fax line is not supposed to
ring local phones, so the most obvious solution was to try and split
the contexts. The configuration below is my current setup that works
almost flawlessly. The bits that aren't working are pretty annoyances
that result from using single context for both lines.
With the setup the way I want it, last two lines of channel 1
configuration and whole channel 2 configuration in zapata.conf would
be uncommented and there would be no fax detection in from-analog-zap
context - as per the comments in the config. As it is now, from-
analog-zap2 may not be giving enough time for fax detection, but I've
tried variations. Generally, all it will do is keep on ringing and
ringing and not detecting the fax tone. I've tried turning off echo
cancelation and few other things with no luck.
Is this possible bug in chan_zap or something related? Is there any
way I can debug more? From just looking at the console it looks like
a regular incoming call that keeps ringing and falling though. Any
ideas or recommendations?
Thanks,
mayo
The setup is as follows:
zapata.conf:
[channels]
language=en
switchtype=national
signalling=fxs_ks
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancelwhenbridged=no
echocancel=64
echotraining=800
callgroup=1 ; i do not use call groups, left it in as it's default
pickupgroup=1
rxgain=0
txgain=0
group=1
immediate=no
context=from-analog-zap1
faxdetect=both ; normally would be: none
channel = 1-2 ; normaly would be: 1
;channel 2 ; normally would be uncommented
;echocancelwhenbridged=no
;echocancel=64
;echotraining=800
;rxgain=0
;txgain=0
;context=from-analog-zap2
;faxdetect=both
;immediate=no
;group=1
;usecallerid=no
;signalling=fxs_ks
;channel = 2
Both channels are in group 1 on purpose, as both lines may be used
for outgoing calls.
extensions.conf:
[from-analog-zap1]
include = incoming
; normally this wouldn't be here if zapata.conf worked as intended.
this bit would get handled by from-analog-zap2
exten = fax,1,GotoIf($[${CHANNEL} != Zap/2-1]?4)
exten = fax,2,Set(FAXFILE=/var/spool/asterisk-fax/${UNIQUEID}.tif)
exten = fax,3,rxfax(${FAXFILE})
exten = fax,4,Hangup
[from-analog-zap2]
exten = s,1,Answer
exten = s,n,LookupCIDName
exten = s,n,NoOp(CallerID: ${CALLERID})
exten = s,n,Hangup
exten = fax,1,Set(FAXFILE=/var/spool/asterisk-fax/${UNIQUEID}.tif)
exten = fax,n,rxfax(${FAXFILE})
exten = fax,n,Hangup
exten = i,1,Hangup
exten = h,1,Hangup
[incoming]
include = parkedcalls
exten = s,1,Answer
exten = s,n(zapateller),Zapateller(nocallerid)
exten = s,n,LookupCIDName
exten = s,n,NoOp(CallerID: ${CALLERID})
exten = s,n(ring),Dial(SIP/2000SIP/2001SIP/2002,30)
exten = s,n,Voicemail(u2999)
exten = s,n,Hangup
exten = s,n(ring)+101,Voicemail(b2999)
exten = s,n,Hangup
exten = i,1,Hangup
exten = h,1,Hangup
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[asterisk-users] 1.4 sounds long space before and after prompt
Is anyone else finding in the new audio files that the longer space at the beginning and end of the files tends to be extremely irritating? An excellent example is when going into voicemail and Allison says how many messages you have, the space between the files is annoyingly long: you have .. four .. old .. messages ..and.. first .. message .. received . July . twenty .. second Under the old sound files, this continuity was still a little long, but workable. The new sound files make these positively sound like a computer playing individual files rather than a continuous sentence. If I release these sound files as they are to my users, they are going to revolt. They already complain about the old Octel VM system prompts being played back too slowly and these are much slower than that. I mentioned this a while back when the new sounds were in beta, but haven't seen anything more about it. So either this says something about my and my users' level of patience, I'm missing something that changed between 1.2 and 1.4 that could fix this, or the focus has been on lower-level issues with 1.4 than on the sound files. With the new higher-quality sound files, I could manually edit all the offending files (there are lots of them) and correct what I perceive to be a problem. However, if this is a common enough complaint, maybe others would want to help as well, and we could get the fixed files put back into core Asterisk. Note that this doesn't appear to be a problem with the speed of the sound files as some others have experienced. The tempo is probably okay, and the pitch is fine. It's the spacing between files that's the issue I'm talking about. Thanks in advance for any feedback. --- Gil Kloepfer [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] TDM2400
After restarting the machine I am getting the below messages when dialing:
Dec 18 00:09:35 WARNING[2897]: channel.c:2571 ast_request: No translator
path exists for channel type Zap (native 68) to 256
Dec 18 00:09:35 NOTICE[2897]: app_dial.c:1056 dial_exec_full: Unable to
create channel of type 'Zap' (cause 0 - Unknown)
On 12/17/06, O. Kamal [EMAIL PROTECTED] wrote:
I am having 2 more issues, when starting asterisk I got the below message:
Dec 17 22:27:54 NOTICE[4554]: indications.c:505
ast_unregister_indication_country: Removed default indication country 'us'
Dec 17 22:27:54 WARNING[4554]: chan_zap.c:10874 setup_zap: Ignoring
signalling
Dec 17 22:27:54 WARNING[4554]: chan_zap.c:10874 setup_zap: Ignoring
answeronpolarityswitch
Dec 17 22:27:54 WARNING[4554]: chan_zap.c:10874 setup_zap: Ignoring
hanguponpolarityswitch
my setup is : softphone---softswitch(asterisk)Termination GW(asterisk
with TDM card)
when dialing from my softphone I got :
-- Executing Dial(SIP/10.8.0.6-0947d008, (Zap/g1/6159587)) in new
stack
Dec 17 23:04:16 WARNING[5052]: channel.c:2597 ast_request: No channel type
registered for '(Zap'
Dec 17 23:04:16 NOTICE[5052]: app_dial.c:1056 dial_exec_full: Unable to
create channel of type '(Zap' (cause 66 - Channel not implemented)
my extensions.conf file has:
[globals]
TRUNK=Zap/g1
[topstn]
exten=_2.,1,Dial,(${TRUNK}/${EXTEN:3})
Please help ...
Thanks,
On 12/12/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Mon, Dec 11, 2006 at 08:08:18PM -0500, Time Bandit wrote:
[channels]
context=default
signalling=fxs_ls
;channel=1-16
usecallerid=yes
hidecallerid=no
callwaiting=yes
restrictcid=no
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
;accountcode=lss0101
answeronpolarityswitch=yes
hanguponpolarityswitch=yes
To the best of my knowledge, all the settings you put after defining
the channles (channel= line) are useless. You have to set all the
settings BEFORE you define the channels.
Should be. However in practice after the first reload all of them will
be applied (in this specific case).
/me points again to genzaptelconf that should have made this thread
unnecessary.
--
Tzafrir Cohen
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto: [EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Re: how to define a secure trunk
no, ipsec headers add much more traffic overhead even than small voice rtp packets bears (using low bitrate codec). this ipsec overhead is not too crucial when ecapsulating relatively big data packet Benny Amorsen wrote: PJ == Pavel Jezek [EMAIL PROTECTED] writes: PJ tunneling small rtp packets through vpn has big overhead, better PJ to use application level encryption - encrypted iax or srtp. IPSEC in transport mode without NAT has a very low overhead. /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] VoipTalk unable to accept calls at present?
I have managed to resolve this. If anybody is interested my machine is multihomed. I set IAX.conf and SIP.conf just to listen on one ip address, this seemed to solve the problem. Regards From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Charlie Grosvenor Sent: 14 December 2006 22:38 To: asterisk-users@lists.digium.com Subject: [asterisk-users] VoipTalk unable to accept calls at present? I am trying to get asterisks to work with http://www.voiptalk.org 's IAX service. I have configured asterisks as per their instructions and am using the x-lite soft phone. When I get an incoming call the softphone rings but the caller (from pstn) gets a recorded message saying the number is unable to accept calls at present. Does anybody know what might be causing this? Thanks ___ --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] Dial 9 For Outside Line?
[default]
Some extensions defined
exten = _0[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1})
exten = _00.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:2})
exten = _09XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
I have the above defined in extensions.conf. This enables me to make
outgoing calls but would like to make it so you have to dial 9 to do
this. Could somebody let me know what I need to change for it to do
this?
Thanks
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Re: [asterisk-users] MOH Between Asterisk Servers
Set the musiconhold class on the original server if you want the MOH to match up correctly. Both servers notice that the call is on hold... makes sense to me. On Fri, 2006-12-15 at 14:56 -0700, Douglas Garstang wrote: Scenario: A call is sent from one Asterisk system to another with IAX. The remote Asterisk system runs the Queue application, which then starts to play a different music on hold class then the standard 'default'. The console on this system displays: -- Executing Queue(IAX2/xxx.yyy.142.203:4569-4, demo_QMain|t|||60) in new stack -- Started music on hold, class 'demo_MainOffice', on IAX2/xxx.yyy.142.203:4569-4 -- Called SIP/2943367 -- Called SIP/2943368 -- SIP/2943367-1bb8 is ringing -- SIP/2943368-537f is ringing However, on the first Asterisk system, we see this on the console: -- Called dundiapps:[EMAIL PROTECTED]/demo_EMain -- Call accepted by xxx.yyy.142.204 (format g729) -- Format for call is g729 -- Started music on hold, class 'default', on IAX2/xxx.yyy.142.203:4569-5 The music on hold class in use is not being conveyed back to the original Asterisk system. Please don't tell me this is a limitation. That would be very very bad. Doug. ___ --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 -- Aaron Daniel Senior Voice Analyst Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] Dial 9 For Outside Line?
exten = _0[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1})
exten = _00.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:2})
exten = _09XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
Just add a 9 in front, like this :
exten = _90[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1})
exten = _900.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:2})
exten = _909XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
hth
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Re: [asterisk-users] Dial 9 For Outside Line?
Just add a 9 in front, like this :
exten = _90[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1})
exten = _900.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:2})
exten = _909XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
Oups, pressed Send too fast, here is take 2
exten = _90[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:2})
exten = _900.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:3})
exten = _909XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:1})
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Re: [asterisk-users] Dial 9 For Outside Line?
Am Sonntag, den 17.12.2006, 18:11 -0500 schrieb Time Bandit:
exten = _0[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1})
exten = _00.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:2})
exten = _09XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
Just add a 9 in front, like this :
exten = _90[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1})
exten = _900.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:2})
exten = _909XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
And don't forget to adapt the EXTEN in the Dial command, else it will
send one digit too much to voiptalk:
exten = _90[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:2})
exten = _900.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:3})
exten = _909XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:1})
(notice the ${EXTEN:x} syntax where x is the number of digits to cut
off from the beginning)
BR
Anselm
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[asterisk-users] BLF on GXP2000
I am trying to set up the BLF on a GXP2000.
Currently what I have is
extensions.conf:
[globals]
polycom430=SIP/101
[internal]
exten = 101,1,Macro(voicemail,${polycom430})
[macro-voicemail]
exten = s,1,Dial(${ARG1},10,tT)
exten = s,2,VoiceMail([EMAIL PROTECTED] )
exten = s,102,VoiceMail([EMAIL PROTECTED])
[ext-local-custom]
exten = 101,hint,${polycom430}
sip.conf:
[general]
subscribecontext=ext-local-custom
And have set up the key to Asterisk BLF with UserID101
When I reload the phone, I get the following error:
[Dec 17 13:34:33] ERROR[2551]: chan_sip.c:14064 handle_request_subscribe:
Got SUBSCRIBE for extension [EMAIL PROTECTED] from 192.168.1.248 , but there is
no hint for that extension
Any help is greatly appreciated.
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[asterisk-users] bridging isdn calls to free up channels
I was incorrect in a previous email... The situation in question is this: Asterisk ---BRI--- PBX ---BRI--- PSTN There are Samsung extensions on the PBX and SIP extensions on Asterisk. I want to be able to use TAPI to initiate dialling, and the PBX has no such feature so Asterisk must initiate it. For an Asterisk initiated call from a PBX extension to a PSTN number, this works as follows: 1. TAPI (eg Outlook) sends the instruction to Asterisk 2. Asterisk calls the extension. 3. The extension answers 4. Asterisk dials the PSTN number 5. Asterisk joins the ends together This works great except it ties up two BRI channels between asterisk and the PBX for the duration of the call. Is there a trick I can do to tell the PBX to join the channels together internally? A transfer or something? I'm using mISDN at the moment, but I guess I could use CAPI if the required features were missing from mISDN... Thanks James ___ --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] sip peer name channel variable?
Started out looking for what I thought was going to be a simple variable name,
have not found it.
Does anyone know of a variable that would contain only the SIP peer name of the
originating channel?
${CHANNEL} contains it, but it needs to be parsed and our dial plan sometimes
uses local channels, in one case it may be SIP/peer-id and in another case
local/peer-id
The peer is defined as type=friend
v1.2.13
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Re: [asterisk-users] FYI Panasonic Wireless Phone MWI
Yes I was very specific. Go back to my original post - search Panasonic MWI - I described what I said and gave a link to the Panasonic specs for this phone which clearly states that the MWI light blinks with new messages and that phone company subscription to VM is required. I did not mention Asterisk because if it works with phone company VM it would work with Asterisk, assuming the FXO you were using was capable and setup correctly. Doug On Sat, 16 Dec 2006, Steve Prior wrote: Noah Miller wrote: Last week I asked about MWI indicators on wireless phones that would work with Asterisk. I sent a message off to Panasonic asking them about it because in their ads they specifically stated that the indicator works with and requires phone company voicemail subscription. That indicator will not work for your voicemail. We do not have any phone system that has a message alert indicator that will work both for your voicemail and your answering machine. How exactly did you phrase the question to their tech support? If you described Asterisk as an answering machine then you'd get the wrong answer. If you described Asterisk as a PBX which provides a signal just like a telco voicemail would, then the answer would make sense. Steve ___ --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 Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CLI output to file
Hi all, How can I redirect the CLI output to file without viewing it on screen? Is it possible. ___ --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] CLI output to file
Hi all, How can I redirect the CLI output to file without viewing it on screen? Is it possible. Read and edit /etc/asterisk/logger.conf You should have already that output at /var/log/asterisk/messages. -- Oh, wow! Look at the moon! ___ --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] CLI output to file
Thanks! But the information in /var/log/asterisk/messages is much different from the messages in CLI. I want to log the message in CLI to file for easy debugging. On 12/18/06, Octavio Ruiz (Ta^3) [EMAIL PROTECTED] wrote: Hi all, How can I redirect the CLI output to file without viewing it on screen? Is it possible. Read and edit /etc/asterisk/logger.conf You should have already that output at /var/log/asterisk/messages. -- Oh, wow! Look at the moon! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CLI output to file
Thanks! But the information in /var/log/asterisk/messages is much different from the messages in CLI. I want to log the message in CLI to file for easy debugging. It is the same, see the levels in logger.conf, copy the console config to the messages config: (just for example) console = notice,warning,error,verbose,dtmf messages = notice,warning,error,verbose,dtmf And you will have the same output. :) On 12/18/06, Octavio Ruiz (Ta^3) [EMAIL PROTECTED] wrote: Hi all, How can I redirect the CLI output to file without viewing it on screen? Is it possible. Read and edit /etc/asterisk/logger.conf You should have already that output at /var/log/asterisk/messages. -- Octavio Ruiz Cervera Neocenter, SA. de CV. http://www.neocenter.com/ Soluciones para Centros de Contacto y Telefonía IP Tel.: (+52 55) 8590-9000 Ext. 9016 Cel.: (+55 55) 5514-087790 ___ --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] Davox
Does anyone know how to connect the Davox dialler to Asterisk? It has a few mentions (such as on the Asterisk business edition page) but no real detail. later, PaulH ___ --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] Bandwidth requirements for 1, 000, 000 minutes a month
I second that Luki. We at www.cyberdyne-ip.com (yes shameless plug) only use ulaw for termination. Of course we have to offer g729, GSM, etc. to our customers... but for best quality, we transcode to ulaw if we send the call to another carrier for termination. 729 may use less bandwidth and in turn cost less, but what is more important... cost or call quality? My 2 cents, bp On 12/15/06, Luki [EMAIL PROTECTED] wrote: But who in there right state if mind would use ulaw? Just take them away to the funny farm ha ha ho ho!! :-P I do. Exclusively. I personally don't like the g729 compression (audio quality and license issues) any my customers definitely notice the difference right away and wonder why the quality degraded. I guess I spoiled them with ulaw. So no g729 here. g726-32 on the other hand was acceptable, although the difference is still noticeable. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good Commercial Grade Service Provider?
Check out www.cyberdyne-ip.com. Great rates, great quality, unlimited channels, and an easy to use GUI to manage your account. FYI, You may have more responses if you ask the -biz list. bp On 12/15/06, Paul Connolly [EMAIL PROTECTED] wrote: We currently have an Asterisk system with a PRI and 6 POTs lines for backup. We are looking to add service such as Voicepulse Connect as an extra level of redundancy and a cost saving alternative to PRI calls. VP Connect only allows 4 simultaneous calls; we are looking for 4 to 5 times that to support our call center. Also, in looking through the archives, it seems like VP has had their share of outages and problems. Can anyone suggest a good commercial grade package/provider? ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good Commercial Grade Service Provider?
I tried to setup an account with Cyberdyne-ip.com after filling out the form all I get when I try to log in is Invalid User name and password please go back javascript:window.history.back(); and try again If the login don't what about there service? :-\ Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-866-638-1254 For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBXt=email Need A Toll Free Number? http://www.bochterservices.com/?t=TFdidt=email For new and used security items http://www.bochterservices.com/?j=storet=email BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email William Piper wrote: Check out www.cyberdyne-ip.com http://www.cyberdyne-ip.com. Great rates, great quality, unlimited channels, and an easy to use GUI to manage your account. FYI, You may have more responses if you ask the -biz list. bp On 12/15/06, *Paul Connolly* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: We currently have an Asterisk system with a PRI and 6 POTs lines for backup. We are looking to add service such as Voicepulse Connect as an extra level of redundancy and a cost saving alternative to PRI calls. VP Connect only allows 4 simultaneous calls; we are looking for 4 to 5 times that to support our call center. Also, in looking through the archives, it seems like VP has had their share of outages and problems. Can anyone suggest a good commercial grade package/provider? ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound (clean). Database: 0659-1, 12/16/2006 - 12/17/2006 11:54:14 PM ___ --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] sip peer name channel variable?
Check out this page:
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sipchaninfo
bp
On 12/17/06, Damon Estep [EMAIL PROTECTED] wrote:
Started out looking for what I thought was going to be a simple variable
name, have not found it.
Does anyone know of a variable that would contain only the SIP peer name
of the originating channel?
${CHANNEL} contains it, but it needs to be parsed and our dial plan
sometimes uses local channels, in one case it may be SIP/peer-id and in
another case local/peer-id
The peer is defined as type=friend
v1.2.13
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Re: [asterisk-users] Good Commercial Grade Service Provider?
Al, I just logged in with _your_ username password and it worked fine for me. I used Internet Explorer and Firefox... both worked fine. I'm guessing that you may have typed in your password wrong. Please contact [EMAIL PROTECTED] from the email that you signed up from and we will forward your login info to you. Thanks, bp On 12/17/06, Al Bochter [EMAIL PROTECTED] wrote: I tried to setup an account with Cyberdyne-ip.comhttp://cyberdyne-ip.com/after filling out the form all I get when I try to log in is Invalid User name and password please go back and try again If the login don't what about there service? :-\ Best regards, Al Bochter Bochter Serviceshttp://www.BochterServices.com/?t=Email http://www.bochterservices.com/?t=Email (VoIP PBX) 1-866-638-1254 For Information on PBX Systems for SOHOhttp://www.bochterservices.com/?j=PBXt=email Need A Toll Free Number?http://www.bochterservices.com/?t=TFdidt=email For new and used security itemshttp://www.bochterservices.com/?j=storet=email BUY Coins, Silver and Goldhttp://www.bochterservices.com/?j=goldt=email William Piper wrote: Check out www.cyberdyne-ip.com. Great rates, great quality, unlimited channels, and an easy to use GUI to manage your account. FYI, You may have more responses if you ask the -biz list. bp On 12/15/06, Paul Connolly [EMAIL PROTECTED] wrote: We currently have an Asterisk system with a PRI and 6 POTs lines for backup. We are looking to add service such as Voicepulse Connect as an extra level of redundancy and a cost saving alternative to PRI calls. VP Connect only allows 4 simultaneous calls; we are looking for 4 to 5 times that to support our call center. Also, in looking through the archives, it seems like VP has had their share of outages and problems. Can anyone suggest a good commercial grade package/provider? ___ --Bandwidth and Colocation provided by Easynews.comhttp://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Inbound (clean). Database: 0659-1, 12/16/2006 - 12/17/2006 11:54:14 PM ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good Commercial Grade Service Provider?
Ok I retyped the same information in same user name them tried to log in and it worked that time. But anyways am in.. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-866-638-1254 For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBXt=email Need A Toll Free Number? http://www.bochterservices.com/?t=TFdidt=email For new and used security items http://www.bochterservices.com/?j=storet=email BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email William Piper wrote: Al, I just logged in with _your_ username password and it worked fine for me. I used Internet Explorer and Firefox... both worked fine. I'm guessing that you may have typed in your password wrong. Please contact [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] from the email that you signed up from and we will forward your login info to you. Thanks, bp On 12/17/06, *Al Bochter* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I tried to setup an account with Cyberdyne-ip.com http://cyberdyne-ip.com/ after filling out the form all I get when I try to log in is Invalid User name and password please go back and try again If the login don't what about there service? :-\ Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email http://www.bochterservices.com/?t=Email (VoIP PBX) 1-866-638-1254 For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBXt=email http://www.bochterservices.com/?j=PBXt=email Need A Toll Free Number? http://www.bochterservices.com/?t=TFdidt=email http://www.bochterservices.com/?t=TFdidt=email For new and used security items http://www.bochterservices.com/?j=storet=email http://www.bochterservices.com/?j=storet=email BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email http://www.bochterservices.com/?j=goldt=email William Piper wrote: Check out www.cyberdyne-ip.com http://www.cyberdyne-ip.com/. Great rates, great quality, unlimited channels, and an easy to use GUI to manage your account. FYI, You may have more responses if you ask the -biz list. bp On 12/15/06, *Paul Connolly* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: We currently have an Asterisk system with a PRI and 6 POTs lines for backup. We are looking to add service such as Voicepulse Connect as an extra level of redundancy and a cost saving alternative to PRI calls. VP Connect only allows 4 simultaneous calls; we are looking for 4 to 5 times that to support our call center. Also, in looking through the archives, it seems like VP has had their share of outages and problems. Can anyone suggest a good commercial grade package/provider? ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound (clean). Database: 0659-1, 12/16/2006 - 12/17/2006 11:54:14 PM ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound (clean). Database: 0659-1, 12/16/2006 - 12/18/2006 12:11:15 AM ___ --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] BLF on GXP2000
what does show hints in the cli display? I think you need to put the
subscribecontext PER SIP DEVICE not global
On 12/17/06, Chris Johnson [EMAIL PROTECTED] wrote:
I am trying to set up the BLF on a GXP2000.
Currently what I have is
extensions.conf:
[globals]
polycom430=SIP/101
[internal]
exten = 101,1,Macro(voicemail,${polycom430})
[macro-voicemail]
exten = s,1,Dial(${ARG1},10,tT)
exten = s,2,VoiceMail([EMAIL PROTECTED] )
exten = s,102,VoiceMail([EMAIL PROTECTED])
[ext-local-custom]
exten = 101,hint,${polycom430}
sip.conf:
[general]
subscribecontext=ext-local-custom
And have set up the key to Asterisk BLF with UserID101
When I reload the phone, I get the following error:
[Dec 17 13:34:33] ERROR[2551]: chan_sip.c:14064 handle_request_subscribe:
Got SUBSCRIBE for extension [EMAIL PROTECTED] from 192.168.1.248 , but there is
no hint for that extension
Any help is greatly appreciated.
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Re: [asterisk-users] Linux distro + Asterisk or Trixbox?
I've used Asterisk on a bunch of RH 7.3 machines which were then replaced by RHEL 4. It is very stable, my biggest compliant is that RHEL(or CentOS, which is a direct rip-off) uses outdated packages (Linux 2.4.x, Apache 1, Mysql 4, php 4, etc) and Linux 2.4.x requires certain USB hardware to use zaptel timing without a hardware card, so we have a bunch of these dual xeon machines with the wrong USB hardware and can only run MeetMe on the one with the t1 cards. So we're moving everything over to SuSE Linux, has more up-to-date packages, still very stable and generally runs asterisk very well. On 12/16/06, Phil Finkler [EMAIL PROTECTED] wrote: Hey all, I've been doing a lot of playing, and a lot of reading, and it seems people are split as to whereas if they're running their favorite Linux distro and asterisk or Trixbox. I'm getting closer to really looking at a production environment and I'm just looking for any opinions. I'm really enjoying learning linux and asterisk, so initial ease of use isn't really a huge benefit to me. In the end stability and upgradeability will be my main concerns. Thanks in advance, Phil ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Fwd: [asterisk-users] Linux distro + Asterisk or Trixbox?
Sorry, I need to take that back. It's RHEL 3, RHEL 4 which we don't run uses Linux 2.6 but not much else is updated. -- Forwarded message -- From: Andrew Joakimsen [EMAIL PROTECTED] Date: Dec 18, 2006 1:47 AM Subject: Re: [asterisk-users] Linux distro + Asterisk or Trixbox? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com I've used Asterisk on a bunch of RH 7.3 machines which were then replaced by RHEL 4. It is very stable, my biggest compliant is that RHEL(or CentOS, which is a direct rip-off) uses outdated packages (Linux 2.4.x, Apache 1, Mysql 4, php 4, etc) and Linux 2.4.x requires certain USB hardware to use zaptel timing without a hardware card, so we have a bunch of these dual xeon machines with the wrong USB hardware and can only run MeetMe on the one with the t1 cards. So we're moving everything over to SuSE Linux, has more up-to-date packages, still very stable and generally runs asterisk very well. On 12/16/06, Phil Finkler [EMAIL PROTECTED] wrote: Hey all, I've been doing a lot of playing, and a lot of reading, and it seems people are split as to whereas if they're running their favorite Linux distro and asterisk or Trixbox. I'm getting closer to really looking at a production environment and I'm just looking for any opinions. I'm really enjoying learning linux and asterisk, so initial ease of use isn't really a huge benefit to me. In the end stability and upgradeability will be my main concerns. Thanks in advance, Phil ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Fwd: [asterisk-users] Linux distro + Asterisk or Trixbox?
On Mon, 18 Dec 2006, Andrew Joakimsen wrote: Sorry, I need to take that back. It's RHEL 3, RHEL 4 which we don't run uses Linux 2.6 but not much else is updated. Keep in mind that the version numbers may be artificially low. Red Hat has an obnoxious policy of back-porting patches and security fixes, so that version X.Y may actually be functionally equivalent to the package released three or four minor revisions later. Neewer versions of CentOS/RHEL do have upgrades where necessary. CentOS 3.5 and later, for example, use Apache 2. -- Steve Sobol, Professional Geek ** Java/VB/VC/PHP/Perl ** Linux/*BSD/Windows Victorville, California PGP:0xE3AE35ED It's all fun and games until someone starts a bonfire in the living room. ___ --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] Multi Operator
Hi I don't see a answer to this question ;=) i am search this solution too .. Thanks bye Jea philippe a écrit : Hi, Actually on my setup all outgoing calls are going trhu a SIP unique account A have a second SIP account with another operator and I would like my setup to use alternatively each of the two accoutns Call 1= Dial SIP/phone1 Call 2= Dial SIP/phone2 Call 3= Dial SIP/phone1 ... If you have an sample please let me know ___ --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] Linux distro + Asterisk or Trixbox?
On Sunday 17 December 2006 10:47 pm, Andrew Joakimsen wrote: I've used Asterisk on a bunch of RH 7.3 machines which were then replaced by RHEL 4. It is very stable, my biggest compliant is that RHEL(or CentOS, which is a direct rip-off) uses outdated packages (Linux 2.4.x, Apache 1, Mysql 4, php 4, etc) and Linux 2.4.x requires certain USB hardware to use zaptel timing without a hardware card, so we have a bunch of these dual xeon machines with the wrong USB hardware and can only run MeetMe on the one with the t1 cards. CentOS 4 was released May 2005 with a 2.6 kernel, Apache 2, and all other similarly current packages. The current kernel is 2.6.9-something. CentOS is a legal re-distribution of RHEL 4 rebuilt from source RPMs. Just like Pie Box, White Box, Tao, Lineox, and all the other Red Hat clones. -- ~ Carla Schroder Linux geek and random computer tamer check out my Linux Cookbook! http://www.oreilly.com/catalog/linuxckbk/ best book for sysadmins and power users ~ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is it possible to use Asterisk voicemail as anouncement system only?
we are using asterisk in combination with the voicemail system. I´m just wondering if it is possible to switch the voicemail to an I am on holiday mode. Just use Play(recordedmsg) instead of voicemail ___ --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] zap sending fax congested
hello all,
i try to send a fax over a zap channel but it is not working i always
get congested but receiving fax over the channel is working.
here are my configs:
zaptel.conf:
# hfc-s pci a span definition
loadzone=nl
defaultzone=nl
span=1,1,3,ccs,ami
bchan=1-2
dchan=3
span=2,2,3,ccs,ami
bchan=4-5
dchan=6
span=3,3,3,ccs,ami
bchan=7-8
dchan=9
span=4,0,3,ccs,ami
bchan=10-11
dchan=12
zapata.conf:
[channels]
language=de
;
signalling=bri_cpe
switchtype=euroisdn
pridialplan=unknown
prilocaldialplan=unknown
nationalprefix=0
internationalprefix=00
usecallingpres=yes
echocancel=yes
echocancelwhenbridged=yes
overlapdial=no
immediate=no
group=1
context=isdn-in
channel = 1-2,4-5,7-8
signalling=bri_net_ptmp
switchtype=euroisdn
pridialplan=local
prilocaldialplan=local
overlapdial=yes
usecallingpres=yes
;echocancel=yes
echocancel=no
echocancelwhenbridged=no
immediate=no
group=2
context=tk-in
channel = 10-11
extensions.conf:
[tk-in]
exten = _X.,1,SetCallerPres(prohib)
exten = _X.,2,CallingPres(32) ; Rufnummer unterdrücken
exten = _X.,3,Dial(ZAP/r1/${EXTEN},120,R) ; rufe die nummer an die
als Parameter ${EXTEN} angeführt wird
; nach einem TImeout von
120s gehe weiter im Schritt 4
exten = _X.,4,Set(PRI_CAUSE=18)
exten = _X.,5,Hangup
exten = _X.,104,Set(PRI_CAUSE=17)
exten = _X.,105,Hangup
Hope somebody can give me a hint
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