Re: [Asterisk-Users] E100P and Colt Telecom (Europe)
Here's our Colt snippet from zaptel.conf # loadzone=nl defaultzone=nl # # dit is de COLT E1 # span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 For zapata.conf we use: ; ; all COLT channels go in group 2 group=2 ; its a pri, connected to the network rxgain=0.0 txgain=0.0 signalling = pri_cpe ; ; calls goto default context=colt ; ; override some settings callerid=asreceived transfer=no ;busydetect=yes ; ; channel 16 is used for D channel info channel = 1-15 channel = 17-31 Good luck! Michiel Aaron Clauson wrote: Hi, Has anyone connected * to a Colt E1 line in Europe? If so could you send me the zaptel.conf and zapata.conf. Thanks, Aaron __ Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages! http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Consult transfer on SNOM 105
Does anyone have a consult-transfer working on SNOM? Using 2.04g we can't get it to work, Hold works, Calling the 3rd party works, but the transfer button does nothing. Playing with the REFER setting on the snom gives varying results on the Asterisk console... but no working transfer :( Michiel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Play a file
Dudlik wrote: Hello I use asterisk ver 0.7.2 Can I play any wave file into the client riciever without billing count ? I call from A IAX client to B IAX client. B client is not available and I would like to play some file with the message user_is_unavailable.gsm But when I look into my CDR table, this call is billed. I don't want to bill these messages. Is it possible ? *CLI show application NoCDR [Synopsis]: Make sure asterisk doesn't save CDR for a certain call [Description]: NoCDR(): makes sure there won't be any CDR written for a certain call [Synopsis]: Make sure asterisk doesn't save CDR for a certain call [Description]: NoCDR(): makes sure there won't be any CDR written for a certain call thank you ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ignorepat with capi
massimo wrote:
Hi to all,
I'm trying to make outside call in this way :
ignorepat = 0
exten = _0.,1,Dial(CAPI/xxx:b${exten})
But the first number 0 is not ignored.
I'm doing something wrong ?
Bye
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Try:
exten = _0.,1,Dial(CAPI/xxx:b${exten:1})
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Re: [Asterisk-Users] GUI?
Altus Snyman wrote: Good day all I'm looking for a GUI/Web interface for Asterisk. What I need it for is to see who's line(SIP) is busy work? Something like a switch board? Please give me some info? Thanks http://www.voip-info.org/wiki-Asterisk+GUI ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zapata required?
Steve, No you don't need the zap drivers if you are not using zap-based hardware. You might want to use the ztdummy driver though as a timing source for conferencing. Since the asterisk zaptel support is in a loadable module you can instruct Asterisk *not* to use it by specifiing noload = chan_zap.so in /etc/asterisk/modules.conf Steven Kokinos wrote: Hello- As part of the asterisk build/installation instructions it mentions that the zaptel drivers should be built and configured first. My question is whether they are required at all, in the case of a system with no hardware cards at all (as is the situation in my case). With them loaded I continually get the following message on my console (server not asterisk): Zapata Telephony Interface Registered on major 196 No ISA tormenta card found at d Zapata Telephony Interface Unloaded which seems logical given that I don't have any zap hardware. how would i go about unloading this module and/or does it need to be compiled at all in this case? Regards, -Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zapata required?
Steven Kokinos wrote: Ho do I go about loading the ztdummy driver after unloading zap? $ su - # modprobe ztdummy Thanks, -Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] openline4
Altus Snyman wrote: Good day Does Asterisk work with the Voicetronix Openline4 cards? Yes, see: http://www.voicetronix.com.au/vpb4_v4pci.htm ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Convert ISDN Card in NT Mode
Ignace CARIA wrote: Hello everybody, I try to connect directly to Asterisk an ISDN DECT base station. Here is the scheme: ISDN Line--ISDN CARD(CAPI)--+Asterisk+--ISDN CARD(???)-DECT Base station. My question is: is it possible to convert the second ISDN card into NT-Mode? If yes, which card must I use? How should I configure it? AFAIK The only card compatible with Asterisk to suppoirt NT1 mode currently is the Zaptel BRI card from Junghanns.net. Check http://www.junghanns.net/asterisk/page17.html Michiel Thank you ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Identifying a call with manager interface
Nicolas Bougues wrote: Dear all, I'm trying to play with the manager interface. What I'd like to do is being able to originate a call and trace its status through events. I use the Originate manager command. I then receive several events telling me about the progress of the call, and then the Response message. However, I didn't find a way to be sure that the first Event I receive after the Originate really relates the call I'm making, and not some other random call, since I believe that I may get events for any channel, not just mine. Note that the Channel I'm using is IAX based, and looks like this : IAX2[217.146.224.41:4569]/3 in the events messages. So I have no way to know it's really mine. Event the final Response message doesn't state the UniqueId of the call. Maybe I missed something obvious. Any idea ? You should be able to specify an ActionID with the originate request. Asterisk will the put this ActionID in all replys to your request. Haven't (yet) tried this myself, but check manager.c for the exact implementation. Michiel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk mangling faxes
Hi list,
Faxes come in over an E1 line (on an TE410P) here and then are sent to
an analog fax machine attached to a T1 (also on the TE410P)
channelbank (CAC1).
Problem is that almost all faxes we send out and receive are mangled...
either only halve pages or very stretched text etc.
Setup in extensions.conf is just:
exten = ${NN_FAX},1,Answer
exten = ${NN_FAX},2,Dial(Zap/49,80)
exten = ${NN_FAX},3,Hangup
echocancel is off for Zap/49 since the path is TDM only
Any pointers to where to look??
Thanks, Michiel
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Re: [Asterisk-Users] Asterisk mangling faxes
Eric Wieling wrote:
First of all Asterisk does not support ${VARIABLES} as part of the
extension number. i.e. exten = ${BLAH},1,NoOp is not valid, but exten
= 1234,1,NoOp(${BLAH}) is valid.
Huh !?! Astreisk might not support it but it seems to work fine in
my setup, show dialplan expands the variables correctly, and more
importantly, the correct fax port is selected based on the dialled numer...
Also Asterisk NEEDS the sending fax machine to send standard fax machine
tones (CNG, I think) for it to be detected. When you send a fax the fax
machine starts sending tones as soon as the call is dialed. Asterisk
listens for these tones and if it hears them it will route the call to
exten = fax,1,Blah if such an extension exists.
Yep, but if you don't have a fax extension defined it will only tell you
it detected a fax tone but won't act on it... problem with using the fax
extension is that it makes it harder to support more then one fax machine.
As I understand it some (many?) fax modems do not send the required tone
when making a fax call.
On Wed, 2004-03-10 at 10:47, Jim Sneeringer wrote:
I'm having the same symptoms using X100P's, TDM400P's and WinFax, and have
and have had no luck correcting it. I don't have a standalone fax machine to
test with.
Does anyone know if this a problem whenever faxes are sent and received with
a modem, or is it specifically WinFax? Is there any other way to accomplish
the goal of computer faxing with Asterisk?
Jim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bisker, Scott
(7805)
Sent: Wednesday, March 10, 2004 8:50 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Asterisk mangling faxes
Michiel,
Are you using WinFax? or one of the Products Based on Winfax? I've seen
this on all of our WinFax Stations, but none of our standalone Fax machines.
-sb
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of michiel betel
Sent: Wednesday, March 10, 2004 9:40 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk mangling faxes
Hi list,
Faxes come in over an E1 line (on an TE410P) here and then are sent to
an analog fax machine attached to a T1 (also on the TE410P)
channelbank (CAC1).
Problem is that almost all faxes we send out and receive are mangled...
either only halve pages or very stretched text etc.
Setup in extensions.conf is just:
exten = ${NN_FAX},1,Answer
exten = ${NN_FAX},2,Dial(Zap/49,80)
exten = ${NN_FAX},3,Hangup
echocancel is off for Zap/49 since the path is TDM only
Any pointers to where to look??
Thanks, Michiel
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Re: [Asterisk-Users] 3com NBX phones
Tim Sailer wrote: Just found on nwfusion.com: 3Com plans to announce the 3102 Business Phone, a SIP-based handset that works with the vendor's VCX IP PBX, technology borrowed from 3Com's now-defunct carrier softswitch business. The phone is also compatible with 3Com's small- and midsize-site NBX IP PBX. The phones support a G.723 wideband audio codec, which 3Com says provides clearer voice than previous 3Com IP phones. The phone is expected to be available March 19 for $310. See also http://www.tmcnet.com/usubmit/2004/Mar/1024782.htm ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: simple H323 question
Ron McMillan wrote:
One way to do it is to use a sniffer, such as ethereal, to capture the
traffic. You should see it in capability exchange, but also easily see in
RTP packets. There might be better ways. But if you're interested in
pursuing it this way and not sure how to do, please follow up with another
question...
Ron
On Fri, 27 Feb 2004, T. Chan wrote:
Hi, all
I wonder when passing calls through asterisk with H323, is there anyway to
find out what codec the calls are using, anyone can help please, thanks alot
!
TC
TC, When using chan_oh323 the codec used is stored in the variable
${OH323_CHANCODEC}
Michiel
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Re: [Asterisk-Users] Failed to start asterisk
Did you change the Makefile to set the processor to i586??? The Via C3
(up to the 900Mhz model) identifies itself as a i686, but misses an
instruction..
dkwok wrote:
I am using mini-itx motherboard and I installed asterisk stable from
cvs. However below is the messages when starting asterisk by
safe_asterisk. Anyone spotted the cause of not starting.
Last login: Fri Feb 27 10:40:44 2004
[EMAIL PROTECTED] root]# safe_asterisk
[EMAIL PROTECTED] root]# /usr/sbin/safe_asterisk: line 77: 3448 Illegal
instruction (core dumped) asterisk ${ASTARGS} 1/dev/${TTY}
/dev/${TTY}
Asterisk ended with exit status 132
Asterisk exited on signal 4.
Automatically restarting Asterisk.
/usr/sbin/safe_asterisk: line 77: 3463 Illegal instruction (core
dumped) asterisk ${ASTARGS} 1/dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 132
Asterisk exited on signal 4.
Automatically restarting Asterisk.
/usr/sbin/safe_asterisk: line 77: 3478 Illegal instruction (core
dumped) asterisk ${ASTARGS} 1/dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 132
Asterisk exited on signal 4.
Automatically restarting Asterisk.
/usr/sbin/safe_asterisk: line 77: 3493 Illegal instruction (core
dumped) asterisk ${ASTARGS} 1/dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 132
Asterisk exited on signal 4.
Automatically restarting Asterisk.
/usr/sbin/safe_asterisk: line 44: /dev/tty9: Input/output error
/usr/sbin/safe_asterisk: line 45: /dev/tty9: Input/output error
Asterisk ended with exit status 1
Asterisk died with code 1. Aborting.
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[Asterisk-Users] Register to h323 gk
Hello group, I am trying to register to a opengk h323 gatekeeper using chan_h323. The gatekeeper expects me to register a username like [EMAIL PROTECTED] with a password secret and an e164 of 31201234567. Thus I put the following in the config file: [general] gatekeeper=w.x.y.z. AllowGKRouted=yes [EMAIL PROTECTED] type=h323 e164=31201234567 secret=geheim context=default However it looks like the gatekeeper doesn't like me and I expect the security to be the reason. Is there a way to specify how the password is communicated? or is plaintext the only supported way at this moment? Thanks! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI - IVR - Time Clock
Jason A. Pattie wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 PBX wrote: | I wanted to post the beginings of my latest IVR Project for an automated | Time Clock software. Not to alarm you too much, but MCI WorldCom has a patent on this kind of thing and is suing people that develop/implement/use these kinds of systems. I wasn't aware that an idea could be patented or even an implementation, but basically, from what I saw they patented a flowchart (!!). Basically, check into the relevant patents for this kind of system. Just giving you a heads up. - -- Jason A. Pattie [EMAIL PROTECTED] Xperience, Inc. (http://www.xperienceinc.com) -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) Comment: Using GnuPG with Debian - http://enigmail.mozdev.org iD8DBQE/+aQOuYsUrHkpYtARAo7CAJ9G38JW4qvLtj1k7RrY3Hc7rEcnRACdFecw IUU5MkfgLHeGpOkoxUrXKPU= =Sadr -END PGP SIGNATURE- If thats true, and I'm old enough to know that with patents anything is possible, the for an Asterisk clock it sounds like REAL -FULL- time employment for Allison :-) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: transfer with threeway calling
Cees de Groot wrote: Matteo Brancaleoni [EMAIL PROTECTED] said: because is different new. Has new powerful features, and old functions has been abandoned for new ones. Yeah, so much is clear. However, because flash doesn't work at a certain moment *and*, AFAIK, has no other functions at that time, I'm simply wondering what the design constraint here is. Because if there is no design constraint, the old-style behavior could simply be added (should, even, IMO) and everyone would be happy... I agree with Cees, however, not wanting to throw away the 3 way conference feature, but giving the user a config choice might be best. Therefore I'm now testing a patch which will allow/disallow the 3 way conference. When disallowed it will fallback to normal old fashioned PBX behaviour namely FLASH puts caller on hold, FLASH again gets caller back. Michiel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AVM ISDN Fritz!Card USB works
Anthony Wood wrote: On Tue, Dec 16, 2003 at 11:28:38AM +1100, Gonzalo Servat wrote: On Tue, 2003-12-16 at 10:34, Michiel Betel wrote: Is case anyone wants to know... The Fritz! USB ISDN box works fine with Asterisk! I'm running CAPI 0.3.0 and love it, because the mini ITX server I have only takes one PCI slot which is now filled with a 4 port Digium card. Is this the micro PABX model 4FXS + 1 ISDN FXO (USB 3.0) or the plain 1 ISDN FXO (USB 2.0)? cheers, Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users It's the little single BRI FXO that AVM calls the FRITZ!CARD USB. Have not tested the other one, athough there is a CAPI driver available for that box too. No need for the mico PABX when running Asterisk is there? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] transfer with threeway calling
Hi, We are using threewaycalling flash transfers over a CAC channelbank. The following happens: Call comes in to my extension I talk to a party and press flash party goes on hold, I get get dail tone I dial internal number internal party answers I press flash once more we are now in a three party conference Or I hang up, and thus transfer the call. Thats fine, but What if the internal party is busy and answers with voicemail how then do I get my original call back to me? Pressing flash will conf in the voicemail prompts of the internal party. Hanging up will transfer the party to the voicemail... Or am I missing somthing essential here?? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] transfer with threeway calling
Read the source luke Just found out the answer to my own question... press FLASH once more... Now the big question is, which part of the source to comment out to stop the 3 way conference... so you get a normal consult and flash will get back the caller. It's very confusing for my users. Michiel Betel wrote: Hi, We are using threewaycalling flash transfers over a CAC channelbank. The following happens: Call comes in to my extension I talk to a party and press flash party goes on hold, I get get dail tone I dial internal number internal party answers I press flash once more we are now in a three party conference Or I hang up, and thus transfer the call. Thats fine, but What if the internal party is busy and answers with voicemail how then do I get my original call back to me? Pressing flash will conf in the voicemail prompts of the internal party. Hanging up will transfer the party to the voicemail... Or am I missing somthing essential here?? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AVM ISDN Fritz!Card USB works
Is case anyone wants to know... The Fritz! USB ISDN box works fine with Asterisk! I'm running CAPI 0.3.0 and love it, because the mini ITX server I have only takes one PCI slot which is now filled with a 4 port Digium card. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Budgetone phones
Todd Wallace wrote: Anyone ever have the Ethernet port on a Budgetone phone quit working. For some reason, it stopped link'ing up and I can't get an address from DHCP or when I set a static address, it would ping. I have reset to factory defaults and nothing seems to work. Feels like the port died, but nothing else is failing. Happened to one of ours too... exaclty the same issues, everyting seens to work except the network port.. Todd Wallace ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_queue different behaviour
Anton, Take a look at the latest version of the patch in: http://bugs.digium.com/bug_view_advanced_page.php?bug_id=214 Good luck! Michiel Anton Yurchenko wrote: Hello, is there a way to make app queue to first try to ring the agents and start music on hold only when they are all talking to other callers? So when the caller calls, and there are free operators he hears ringing, and * is not picking up until call is answere, or specified timeout. And if the caller calls , and there are no free operators , some message like please wait for next avalable operator and them the music on hold start. thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Crash - What is happening here???
The following transfer led to a crash of asterisk, without leaving a core or any utterances in messages or debug file. It looks like the zombie which was created during the MASQ-transfer was not cleaned up... But why did it start a Dial??? And... why does Asterisk die when this happens?? Thanks!!! Michiel -- Zap/32-1 answered Zap/6-1 -- Stopped music on hold on Zap/6-1 -- Starting simple switch on 'Zap/32-2' -- Started three way call on channel 32 -- Started music on hold, class 'default', on Zap/6-1 -- Executing Macro(Zap/32-2, stdexten2|6112|Zap/34|20|tT) in new stack -- Executing SetLanguage(Zap/32-2, nl) in new stack -- Executing DBget(Zap/32-2, fwdexten=FEAT/6112/CFWD/CFU) in new stack -- DBget: varname=fwdexten, family=FEAT, key=6112/CFWD/CFU -- DBget: Value not found in database. -- Executing Goto(Zap/32-2, s|5) in new stack -- Goto (macro-stdexten2,s,5) -- Executing Dial(Zap/32-2, Zap/34|20|tT) in new stack -- Called 34 -- Zap/34-1 is ringing -- Zap/34-1 is ringing -- Stopped music on hold on Zap/6-1 -- Hungup 'Zap/6-1MASQ' -- Hungup 'Zap/32-1' == Spawn extension (netland_admin, s, 3) exited non-zero on 'Zap/32-2ZOMBIE' -- Executing Macro(Zap/32-2ZOMBIE, record-cleanup) in new stack -- Executing SetVar(Zap/32-2ZOMBIE, MONITORDIR=/var/spool/asterisk/monitor) in new stack -- Executing GotoIf(Zap/32-2ZOMBIE, 1?5:3) in new stack -- Goto (macro-record-cleanup,s,5) -- Executing NoOp(Zap/32-2ZOMBIE, ) in new stack -- Executing ChanIsAvail(Zap/32-2ZOMBIE, Zap/32) in new stack -- Hungup 'Zap/32-1' -- Executing Dial(Zap/32-2ZOMBIE, Zap/32|40|tm) in new stack -- Called 32 -- Started music on hold, class 'default', on Zap/32-2ZOMBIE -- Zap/32-1 is ringing -- Zap/32-1 is ringing -- Zap/34-1 is ringing -- Zap/34-1 answered Zap/6-1 -- Attempting native bridge of Zap/6-1 and Zap/34-1 -- Zap/32-1 is ringing -- Zap/32-1 is ringing -- Zap/32-1 is ringing -- Zap/32-1 answered Zap/32-2ZOMBIE -- Stopped music on hold on Zap/32-2ZOMBIE n010205*CLI Disconnected from Asterisk server ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Crash - What is happening here???
Matteo, I AM running -gc and ulimit -c unlimited (from safe_asterisk) on RH7.2 Thats the weird thing... it crashed without any message. And looking through the source I still don't see how the Dial could start on a Zombie channel... But you are right, I'll try to reproduce it tomorrow morning (Its a production system) Michiel Matteo Brancaleoni wrote: Small tutorial: these errors are too generic to be solved in such way... hey my asterisk crashed, why it did?... there're many reasons... First: set ulimit -c unlimited on the console from which * starts, to let it dump cores. Then start it with 'g' in his parms , like asterisk -vvvgc, to enable debugging... then when it crashed, run gdb on the core and backtrace it also: try to find a way to reproduce the crash. random crashed aren't very useful... and... report also asterisk version, kernel, distro, blah blah blah Michiel, that message isn't only for you, but your post triggered my thoughts to how to report a crash, for anyone that just jump on th ML and say my asterisk crashed. please say me why... bye, matteo Scrive Michiel Betel [EMAIL PROTECTED]: The following transfer led to a crash of asterisk, without leaving a core or any utterances in messages or debug file. It looks like the zombie which was created during the MASQ-transfer was not cleaned up... But why did it start a Dial??? And... why does Asterisk die when this happens?? Thanks!!! Michiel -- Zap/32-1 answered Zap/6-1 -- Stopped music on hold on Zap/6-1 -- Starting simple switch on 'Zap/32-2' -- Started three way call on channel 32 -- Started music on hold, class 'default', on Zap/6-1 -- Executing Macro(Zap/32-2, stdexten2|6112|Zap/34|20|tT) in new stack -- Executing SetLanguage(Zap/32-2, nl) in new stack -- Executing DBget(Zap/32-2, fwdexten=FEAT/6112/CFWD/CFU) in new stack -- DBget: varname=fwdexten, family=FEAT, key=6112/CFWD/CFU -- DBget: Value not found in database. -- Executing Goto(Zap/32-2, s|5) in new stack -- Goto (macro-stdexten2,s,5) -- Executing Dial(Zap/32-2, Zap/34|20|tT) in new stack -- Called 34 -- Zap/34-1 is ringing -- Zap/34-1 is ringing -- Stopped music on hold on Zap/6-1 -- Hungup 'Zap/6-1MASQ' -- Hungup 'Zap/32-1' == Spawn extension (netland_admin, s, 3) exited non-zero on 'Zap/32-2ZOMBIE' -- Executing Macro(Zap/32-2ZOMBIE, record-cleanup) in new stack -- Executing SetVar(Zap/32-2ZOMBIE, MONITORDIR=/var/spool/asterisk/monitor) in new stack -- Executing GotoIf(Zap/32-2ZOMBIE, 1?5:3) in new stack -- Goto (macro-record-cleanup,s,5) -- Executing NoOp(Zap/32-2ZOMBIE, ) in new stack -- Executing ChanIsAvail(Zap/32-2ZOMBIE, Zap/32) in new stack -- Hungup 'Zap/32-1' -- Executing Dial(Zap/32-2ZOMBIE, Zap/32|40|tm) in new stack -- Called 32 -- Started music on hold, class 'default', on Zap/32-2ZOMBIE -- Zap/32-1 is ringing -- Zap/32-1 is ringing -- Zap/34-1 is ringing -- Zap/34-1 answered Zap/6-1 -- Attempting native bridge of Zap/6-1 and Zap/34-1 -- Zap/32-1 is ringing -- Zap/32-1 is ringing -- Zap/32-1 is ringing -- Zap/32-1 answered Zap/32-2ZOMBIE -- Stopped music on hold on Zap/32-2ZOMBIE n010205*CLI Disconnected from Asterisk server ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zt_rec: Unknown error 500
I have a number of Zap/ extensions defined in a queue with ringall strategy. When this queue is called sometimes Asterisk seems to think that one of these channels is busy, while it is NOT. The following is shown on the console: --Called 44 -- Called 36 -- Called 41 -- Called 35 -- Called 38 -- Zap/44-1 is ringing -- Zap/36-1 is ringing -- Zap/41-1 is ringing -- Zap/35-1 is ringing -- Zap/38-2 is ringing -- Zap/44-1 is ringing -- Zap/36-1 is ringing -- Hungup 'Zap/35-1' -- Zap/41-1 is ringing -- Zap/44-1 is ringing -- Zap/36-1 is ringing While a WARNING[165916]: File chan_zap.c, Line 3331 (zt_read): zt_rec: Unknown error 500 is generated in /var/log/asterisk/messages Any ideas on how to fix this?? Thanks! Michiel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zt_rec: Unknown error 500
My Zap channels having the problems are on a T1 connected to a CAC channelbank, But it looks like the zt_rec in chan_zap error uses the lowlevel zaptel ioctl's which are the same for T1 PRI... Scott Stingel wrote: Hi Michiel- This may be related to a PRI frame buffer overflow problem that I get in high-volume IVR applications. I get a lot of these errors mixed in with frame errors. In my case its load related. Mark and Martin at Digium have said they'll be looking into improving the buffering mechanism. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel Betel Sent: Tuesday, November 25, 2003 1:43 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] zt_rec: Unknown error 500 I have a number of Zap/ extensions defined in a queue with ringall strategy. When this queue is called sometimes Asterisk seems to think that one of these channels is busy, while it is NOT. The following is shown on the console: --Called 44 -- Called 36 -- Called 41 -- Called 35 -- Called 38 -- Zap/44-1 is ringing -- Zap/36-1 is ringing -- Zap/41-1 is ringing -- Zap/35-1 is ringing -- Zap/38-2 is ringing -- Zap/44-1 is ringing -- Zap/36-1 is ringing -- Hungup 'Zap/35-1' -- Zap/41-1 is ringing -- Zap/44-1 is ringing -- Zap/36-1 is ringing While a WARNING[165916]: File chan_zap.c, Line 3331 (zt_read): zt_rec: Unknown error 500 is generated in /var/log/asterisk/messages Any ideas on how to fix this?? Thanks! Michiel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zt_rec: Unknown error 500
I know... bad form to add to my own posting but: I found out that the Unknown error only appears when ringing multiple extensions at nearly the same time. When ringing two Zap channels (with ) it takes a little longer but eventually the error will crop up and one of the ringing channels will hang up. Michiel Betel wrote: I have a number of Zap/ extensions defined in a queue with ringall strategy. When this queue is called sometimes Asterisk seems to think that one of these channels is busy, while it is NOT. The following is shown on the console: --Called 44 -- Called 36 -- Called 41 -- Called 35 -- Called 38 -- Zap/44-1 is ringing -- Zap/36-1 is ringing -- Zap/41-1 is ringing -- Zap/35-1 is ringing -- Zap/38-2 is ringing -- Zap/44-1 is ringing -- Zap/36-1 is ringing -- Hungup 'Zap/35-1' -- Zap/41-1 is ringing -- Zap/44-1 is ringing -- Zap/36-1 is ringing While a WARNING[165916]: File chan_zap.c, Line 3331 (zt_read): zt_rec: Unknown error 500 is generated in /var/log/asterisk/messages Any ideas on how to fix this?? Thanks! Michiel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Netphone SIP phone
Does anyone have experience using the Netphone SIP phone from Ortena Networks (http://www.ortena.com). I contacted them, and they will sell me 10 units for 95 euros/unit. At least i -looks- better then the Grandstream :-) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Still TDM400P problem
JanM wrote: Hi again all, I have searched the list for help with my problem but I can´t find an answer. I only manage to get one port of my TDM400P card working. When I do dmesg I get following, seems like four discovered ports: --- Zapata Telephony Interface Registered on major 196 PCI: Found IRQ 11 for device 02:00.0 PCI: Sharing IRQ 11 with 02:07.1 PCI: Sharing IRQ 11 with 02:0c.0 Freshmaker version: 63 Freshmaker passed register test Module 0: Installed -- AUTO Module 1: Installed -- AUTO Module 2: Installed -- AUTO Module 3: Installed -- AUTO Found a Wildcard FXS: Wildcard TDM400P REV E/F (4 modules) Registered tone zone 5 (Finland) But when I do ztcfg -vv I only get one port configured: Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) 1 channels configured. How do I configure/load the rest of the ports? Add them in /etc/zaptel.conf... ztcfg reads this file and configures zap ports accordingly Michiel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco to use * as a gateway?
Joseph Finley wrote:
I'm not sure if I am wording this correctly, but I'll try.
I have a Cisco 2621 w/ a couple FXO and FXS ports. I have a couple cheap
analog phones plugged into the FXS ports. I am able to get * to ring those
phones when a call comes in, but I cannot get the phones to dial out. I
guess it's all syntax that I'm doing wrong. Does someone have a couple
small snip-its to accomplish this?
Thanks
Joe
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are you using SIP??
if so...
exten = _0XX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5060)
exten = _0XX,2,Congestion
where W.X.Y.Z is the IP address of your Cisco
Michiel
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Re: [Asterisk-Users] soxmix/gsm
Dave,
I use:
exten = s,1,SetVar(MONITORDIR=/var/spool/asterisk/monitor)
exten = s,2,GotoIf($[${CALLFILENAME} = ${FOO}]?6:3)
exten = s,3,System(soxmix ${MONITORDIR}/${CALLFILENAME}-in.wav
${MONITORDIR}/${CALLFILENAME}-out.wav ${MONITORDIR}/${CALLFILENAME}.gsm)
exten = s,4,System(/bin/rm ${MONITORDIR}/${CALLFILENAME}-in.wav
${MONITORDIR}/${CALLFILENAME}-out.wav)
exten = s,5,NoOp
Which records wav files, but mixes them into a single .gsm. Quality is
good, no choppyness.
Michiel
David C. Troy wrote:
All --
I'm still having serious trouble mixing two gsm files together. They are
generated from Monitor(gsm).
The only way I can get successful results right now is to do Monitor(wav)
and then use soxmix to convert the resulting two wav files into a single
gsm file, but the sound quality is warbly and the wav files unnecessarily
large.
Is anyone successfully using soxmix to merge two existing gsm files into a
third gsm file? If so, what version of soxmix, etc?
Cheers,
Dave
=
David C. Troy [EMAIL PROTECTED] 410-384-2500 Sales
ToadNet - Want to go fast?410-544-1329 FAX
570 Ritchie Highway, Severna Park, MD 21146-2925 www.toad.net
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[Asterisk-Users] Cisco router/SIP gateway registration
Is there a way to have a Cisco SIP gateway register with Asterisk? The current setup just drops calls into the sip.conf default context which works fine but has some security risks since anyone who can install XTEN and has access to my LAN can then use this context to drop calls in I'd like to be able to get inbound calls from the cisco in a from_gw context, then I can just set the default context to a simple Congestion dialplan... Thanks! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] recording files for menues
Shoval Tomer wrote: How do you suggest doing that? How can I convert wav files to gsm files? thanks #!/bin/sh for i in *.wav; do sox $i -r 8000 `basename $i *.wav`.gsm resample -ql; done ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] recording files for menues
Shoval Tom wrote: Either it's not working, or I don't know what I'm doing. It's giving me the error sox: effect '.gsm' is no known! Sounds like your copy of sox was not compiled with gsm enabled.. or you put a space between the ...wav`.gsm bit check with a single file like this: $ sox file.wav -r 8000 file.gsm resample -ql Michiel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] recording files for menues
Sorry... I was a bit in a hurry, and indeed I cannot expect all list readers to know about shell scripts... will elaborate a bit more in the future. I noticed you removed the sox resample -ql options, which on my studio recorded .wav files helped a bit, also It might be sensible to add a -c 1 to make sure sox will convert a stereo file to a single channel .gsm Regards, Michiel Shoval Tom wrote: Olle, I can't reach the faq page, and haven't been able to for the last four days. I'm getting 504 gateway timeout errors. Any ideas? Btw, the first answer I got worked, I mistook ` for ' (newbie error, I know...) To be more specific for you newbies out there Create a file containing: copy below this line #!/bin/sh for i in *.wav; do sox $i `basename $i .wav`.gsm;done up to this line save it in your path, or in the directory containing the files you want to convert do a chmod +x filename (where filename is the name of your saved file) now you can run it while in the directory and it'll convert all *.wav files for you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Answering Machine Detection
See http://resource.intel.com/telecom/support/documentation/unix/SR50_linux/html _files/vox_feat/contents.html#TopOfPage chapter 2 for a basic insight on Dialogic does it... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: woensdag 29 oktober 2003 3:12 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Answering Machine Detection Humans tend to say Hello? (short burst of audio followed by silence), and answering machines tend to say I'm sorry I'm not here right now, please leave a message after the beep (long burst of audio followed by a beep and silence). So, basically you need to decide 1) what is audio and what is background noise and 2) how long should there be audio followed by silence. On Tue, 2003-10-28 at 19:25, Alastair Maw wrote: On 27/10/03 21:57, DUSTIN WILDES wrote: Does anyone have any recommendations on implementing Answering Machine detection for call generation programs? There's obviously no nice way of doing this. If you're doing telemarketing, and you're playing pre-recorded audio, which of course is a nasty thing to do, the algorithm is something like: 1. Dial out. 2. Wait for answer. 3. Start playing audio. 4. If you hear something that sounds like a beep, either hang up and try again later, or stop the audio, pause for two seconds and start playing it again. 5. Hang up when finished playing audio. Step 4 is accomplished by doing a FFT on the incoming audio into frequency buckets and taking a rolling average of the mean and standard deviation, such that you can detect when a fixed monotone beep occurs at the other end. If you don't want to play audio files and wait for beeps, and want to connect real humans to each other, then there's no decent way to do this, as the only difference between humans and arbitrary answering machines is that the answering machines give you a beep prompt to record your message. Regards, -- Sample configs, scripts, more : http://www.fnords.org/~eric/asterisk/ BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] core dump in app_dial
My asterisk suddenly died in ast_verbose, called from app_dial... leaving a core which told me the following: (gdb) where #0 0x4011a7c3 in chunk_free () from /lib/libc.so.6 #1 0x4011a548 in free () from /lib/libc.so.6 #2 0x080529fa in ast_verbose () #3 0x40493b76 in wait_for_answer (in=0x811f8d0, outgoing=0x81154e0, to=0xbcdff094, allowredir_in=0xbcdff098, allowredir_out=0xbcdff09c, allowdisconnect=0xbcdff0a0) at app_dial.c:322 #4 0x40494a3a in dial_exec (chan=0x811f8d0, data=0xbcdff77c) at app_dial.c:619 #5 0x08061666 in pbx_exec () #6 0x0806324a in pbx_extension_helper () #7 0x08063eee in ast_pbx_run () #8 0x08069dbf in pbx_thread () #9 0x40023f77 in pthread_start_thread () from /lib/libpthread.so.0 This is on RedHat 7.3, using CVS of 7 october 2003 The log around the crash time shows: Oct 27 12:57:23 WARNING[65556]: File ast_expr.y, Line 346 (ast_yyerror): ast_yyerror(): syntax error: parse error Oct 27 12:58:15 WARNING[67604]: File chan_zap.c, Line 3331 (zt_read): zt_rec: Unknown error 500 Oct 27 12:58:15 WARNING[67604]: File chan_zap.c, Line 3331 (zt_read): zt_rec: Unknown error 500 Oct 27 12:58:56 WARNING[67604]: File ast_expr.y, Line 346 (ast_yyerror): ast_yyerror(): syntax error: parse error Any ideas?? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] single dialplan for multiple Asterisk machines
Matt, It's done by using the switch keyword in extensions.conf Thus if you fill in the stuff below correctly and make the appropriate settings in iax.conf: switch = IAX/username:[EMAIL PROTECTED]/context Will send all extensions which cannot be resolved in the local dialplan, over IAX to the asterisk instance where your switch statement is pointing to. They will end up in the context you specify. Michiel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mattf Sent: woensdag 1 oktober 2003 15:59 To: '[EMAIL PROTECTED]' Subject: [Asterisk-Users] single dialplan for multiple Asterisk machines I have heard it mentioned several times by different people but can anyone explain to me how you can set up a single dialplan for 2 or more than asterisk boxes located on the same local network? MATT--- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Config TE410P + TDM400
When configuring a TE410P which is only attached to a single E1 together with a TDM400, how should one count the channels for the next Zap interface? Must I put 4 span lines in zapata.conf and define all channels up to 124? thus having the TDM400's start at 125? Or can I comment out the 3 spans I don't use and start at channel 32 for the TDM400? (this would get nasty when adding extra lines, but would stop asterisk from trying to look at E1's which are not connected) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] list of voice prompts
Me = stupid!!! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut . Sent: woensdag 24 september 2003 11:22 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] list of voice prompts Take a look at sounds.txt in the root of your Asterisk source.. Does there exist a text file with all the 'standard' Asterisk voice messages? I'm planning to get them recorded in dutch, but need to know the exact text of each prompt... Michiel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco Gateways
I'm using cico's with SIP... And it works great :-) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Edward Gomez Sent: dinsdag 16 september 2003 15:52 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco Gateways Hi all, Just wondering if * can work with Cisco Gateways such as Cisco 2600/3600 routers or a VG200? -- Edward J. Gomez Director of Network Services ProxyMed, Inc 2555 Davie Road, Suite 110 Fort Lauderdale, Florida 33317 (954) 473-1001 x315 (954) 473-1656 FAX http://www.proxymed.com/ Confidential, unpublished property of ProxyMed, Inc. (c) copyright as of the date of this email. ProxyMed, Inc. CONFIDENTIALITY NOTICE: This e-mail message, including any attachments and files transmitted with it, are confidential and are intended solely for the use of the individual or entity to whom they are addressed. It may contain information that is privileged, confidential and exempt from disclosure under applicable laws. Moreover, this communication may contain the original sender's personal views and opinions, which do not necessarily reflect those of ProxyMed, Inc. . If the reader of this message is not the intended recipient, or the employee or agent responsible for delivering the message to the intended recipient, or if you have received this communication in error, please notify us immediately by return e-mail and delete the original message and any copies of it from your system. If you are not the intended recipient, be advised that you have received this e-mail in error, and that any unauthorized review, use, disclosure, distribution, forwarding, printing, or copying of this e-mail is strictly prohibited without our prior, written permission. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outgoing SIP trunk
First it helps to read the documentation.. Read up on the Dial
application.
Then put something more elaborate then this example in extensions.conf:
exten= _9.,1,Dial(SIP/{EXTEN:[EMAIL PROTECTED],20,t)
Have fun!
On Mon, 2003-09-15 at 04:39, Juan J. Sierralta P. wrote:
Hi,
I´m new to Asterisk. What I´m trying to set up is to use SER as a SIP
provider for Asterisk and route all non-local calls through SER (which
is connected to Cisco Gateways), I was able to register Asterisk on SER.
But I don´t know how to tell Asterisk to use the SIP channels as the
outbound trunk.
I was able to set the Console to SIP/[EMAIL PROTECTED] but I need
Asterisk to change someuser with the number actually dialed by the
local users.
Any suggestions ?
TIA
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RE: [Asterisk-Users] Ztdummy not loaded
It might help if you have the module compiled installed :-) Check the Makefile in /usr/src/zaptel and uncomment ztdummy. Then do a make install and modprobe ztdummy Michiel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chad Brown Sent: zondag 14 september 2003 10:23 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Ztdummy not loaded I was having problems with conferencing when I found a list which suggested that ztdummy might not be loaded. I checked using lsmod and sure enough it was not loaded. When trying to load ztdummy I get an error saying Can't locate module ztdummy. I am using Asterisk CVS-09/13/03-23:21:19 Any help would be appreciated. Thanks, Chad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dialplan question
Title: Message
Fredrik,
Your
dialplan looks correct, however you disallow 112, the emergency
number!
Does
it fail for local or interlocal calls?
I
use:
[dutchdial];;emergency (112) and other 11x
numbers;exten = _1XX,1,Dial(${ISDN}:${EXTEN})exten =
_1XX,2,Congestionexten = _01XX,1,Dial(${ISDN}:${EXTEN:1})exten =
_01XX,2,Congestion;;0900 0800 numbers;exten =
_00[89]00.,1,SetCIDNum(0206408219)exten =
_00[89]00.,2,Dial(${ISDN}:${EXTEN:1})exten =
_00[89]00.,3,Congestion;;International;exten =
_000.,1,Dial(${ISDN}:${EXTEN:1})exten =
_000.,2,Congestion;;Local (7 digits, add area code);exten =
_0XXX,1,SetCIDNum(0206408219)exten =
_0XXX,2,Dial(${ISDN}:020${EXTEN:1})exten =
_0XXX,3,Congestion;;Interlocal, 10 digits;exten =
_0XX,1,SetCIDNum(0206408219)exten =
_0XX,2,Dial(${ISDN}:${EXTEN:1})exten =
_0XX,3,Congestion
And it
works fine
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of fredrik
chabotSent: zaterdag 6 september 2003 18:25To:
[EMAIL PROTECTED]Subject: [Asterisk-Users] Dialplan
questionHi,Dialplan QuestionI'm in
holland and I have:[naarbuiten]ignorepat = 0;
interlocaalexten =
_00[1-9],1,Dial(Modem/g1:${EXTEN})
exten = _00[1-9],2,Congestion; locaalexten =
_0[1-9]XX,1,Dial(Modem/g1:${EXTEN})
exten = _0[1-9]XX,2,CongestionAnd sometimes I can get
out, most of the time however I get a busy signal halfway throu the
number.It works more often if I change Early Dial:
No Yes (use "Yes" only if proxy supports 484
response)to No. In the Budgetone 100
phone.regards,fredrik chabot---*CLI
show dialplan [ Context 'default' created by 'pbx_config' ]
Include =
'demo'
[pbx_config][ Context 'demo' created by 'pbx_config' ] '#'
= 1.
Playback(demo-thanks)
[pbx_config]
2.
Hangup()
[pbx_config] '100'
= 1.
Dial(SIP/100)
[pbx_config] '101'
= 1.
Dial(SIP/101)
[pbx_config] '190'
= 1.
Dial(Modem/g1:006400)
[pbx_config] '8500'
= 1.
VoicemailMain()
[pbx_config]
2.
Goto(s|6)
[pbx_config] 'i'
= 1.
Playback(invalid)
[pbx_config] 's'
= 1.
Wait(1)
[pbx_config]
2.
Answer()
[pbx_config]
3.
DigitTimeout(5)
[pbx_config]
4.
ResponseTimeout(10)
[pbx_config]
5.
BackGround(demo-congrats)
[pbx_config]
6.
BackGround(demo-instruct)
[pbx_config] 't'
= 1.
Goto(#|1)
[pbx_config] Include
=
'naarbuiten'
[pbx_config][ Context 'naarbuiten' created by 'pbx_config' ]
'_00[1-9]' = 1.
Dial(Modem/g1:${EXTEN})
[pbx_config]
2.
Congestion()
[pbx_config] '_0[1-9]XX' = 1.
Dial(Modem/g1:${EXTEN})
[pbx_config]
2.
Congestion()
[pbx_config] Ignore pattern =
'0'
[pbx_config][ Context 'vanbuiten' created by 'pbx_config' ]
's' = 1.
Wait(1)
[pbx_config]
2.
Answer()
[pbx_config]
3.
DigitTimeout(5)
[pbx_config]
4.
ResponseTimeout(10)
[pbx_config]
5.
Playback(tt-weasels)
[pbx_config]
6.
Dial(SIP/100|4)
[pbx_config]
7.
Dial(SIP/100SIP/101|10)
[pbx_config]
8. Dial(SIP/100SIP/101Modem/g1:0064000)
[pbx_config]___
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http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Change include contexts runtime
All the errors you get are associated with not having the prompts recorded... If you do a show database at the CLI you'll see that it actually made the entry in the database.. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomas Prybil Sent: woensdag 3 september 2003 15:06 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Change include contexts runtime John Congdon wrote: Here is an example I stole off the list awhile back exten = *5,1,DBget($Night=GlobalSettings/Night) ; if not night then set it exten = *5,2,DBdel(GlobalSettings/Night) exten = *5,3,Playback(night_off) exten = *5,4,Hangup exten = *5,102,DBput(GlobalSettings/Night=true) exten = *5,103,Playback(night_on) exten = *5,104,Hangup Set Night and then have a voice response saying Night on or Night Off accordingly... Do yu need to initiate any external db in anyway. Exept from not having the right prompt recorded right know I get error messages like this: Executing Hangup(SIP/9002-4f88, ) in new stack == Spawn extension (siphone, *5, 4) exited non-zero on 'SIP/9002-4f88' -- Executing DBget(SIP/9002-2e67, $Night=GlobalSettings/Night) in new stack -- DBget: varname=$Night, family=GlobalSettings, key=Night -- DBget: Value not found in database. -- Executing DBput(SIP/9002-2e67, GlobalSettings/Night=true) in new stack -- DBput: family=GlobalSettings, key=Night, value=true -- Executing Playback(SIP/9002-2e67, night_on) in new stack WARNING[442385]: File file.c, Line 443 (ast_openstream): File night_on does not exist in any format WARNING[442385]: File file.c, Line 717 (ast_streamfile): Unable to open night_on (format 4): No such file or directory WARNING[442385]: File app_playback.c, Line 83 (playback_exec): ast_streamfile failed on SIP/9002-2e67 for night_on -- Executing Hangup(SIP/9002-2e67, ) in new stack == Spawn extension (siphone, *5, 104) exited non-zero on 'SIP/9002-2e67' /t ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DBSaveTree DBLoadTree
OOOPS Indeed! My fault... They do persist if you the system it correctly. Sorry, Michiel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: maandag 1 september 2003 5:12 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] DBSaveTree DBLoadTree On Sunday 31 August 2003 16:49, Michiel Betel wrote: The db entries persist on reload, on a restart (or crash...) they are gone... Are you perhaps running Asterisk as a user other than root? Sounds like you might not have permission to write to /var/lib/asterisk/. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DBSaveTree DBLoadTree
Title: Message Hi all, Has anyone already written something which allows saving and loading the internalDB settings? All users CFWD and speeldial settings are stored in the DBin my setup which makes it a pain to restart Asterisk Looking at showtree in db.c (why isn't that exposed in the CLI?) It shouldn't be too difficult, but I don't want to reinvent the wheel. On the same track, I am also looking at exposing DBput DBget to the manager interface, thus making it easy to st global stuff like nightsettings... Michiel Betel ConsultancyAbelenlaan 19 T: +31 20 640 30181185 RT Amstelveen E: [EMAIL PROTECTED]The NetherlandsW: www.betel.nl Confidentiality Notice - The information contained in this e-mail is intended for the named recipient(s) only. It may contain privileged and confidential information, and if you are not the addressee or the person responsible for delivering this to the Addressee, you may not copy, distribute or take action in reliance on it. If you have received this e-mail in error, please notify us immediately by returning the original message to the sender by e-mail!
RE: [Asterisk-Users] DBSaveTree DBLoadTree
The db entries persist on reload, on a restart (or crash...) they are gone... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Weis Sent: zondag 31 augustus 2003 20:39 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] DBSaveTree DBLoadTree On Sun, 31 Aug 2003, Michiel Betel wrote: Has anyone already written something which allows saving and loading the internal DB settings? All users CFWD and speeldial settings are stored in the DB in my setup which makes it a pain to restart Asterisk Looking at showtree in db.c (why isn't that exposed in the CLI?) It shouldn't be too difficult, but I don't want to reinvent the wheel. Doesn't the db stuff persist on disk during reloads? On the same track, I am also looking at exposing DBput DBget to the manager interface, thus making it easy to st global stuff like nightsettings... I will be doing that if you don't get to it within the next week or so. dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_capi compile errors with latest CVS
Title: Message Did something change in lock.h lately? I get all kind of ast_mutex errors when trying to compile chan capi 0.24c with the latest asterisk code Betel ConsultancyAbelenlaan 19 T: +31 20 640 30181185 RT Amstelveen E: [EMAIL PROTECTED]The NetherlandsW: www.betel.nl Confidentiality Notice - The information contained in this e-mail is intended for the named recipient(s) only. It may contain privileged and confidential information, and if you are not the addressee or the person responsible for delivering this to the Addressee, you may not copy, distribute or take action in reliance on it. If you have received this e-mail in error, please notify us immediately by returning the original message to the sender by e-mail!
[Asterisk-Users] zaptel sync
Title: Message Simple Q but I can't find the answer in the archives (and am too lazy to look in the source, but then its 32 Celcius here... Do all digium cards provide the zapata timing? e.g.also the analogs (including the X100P)or only the E1/T1 -ones or do I need to use ztdummy on the analog cards? Thanks, Michiel Betel ConsultancyAbelenlaan 19 T: +31 20 640 30181185 RT Amstelveen E: [EMAIL PROTECTED]The NetherlandsW: www.betel.nl Confidentiality Notice - The information contained in this e-mail is intended for the named recipient(s) only. It may contain privileged and confidential information, and if you are not the addressee or the person responsible for delivering this to the Addressee, you may not copy, distribute or take action in reliance on it. If you have received this e-mail in error, please notify us immediately by returning the original message to the sender by e-mail!
[Asterisk-Users] Retry dial when busy
Title: Message Some switches provide the functionality to try a number till it becomes available. Thus whenone dials a number and get a busy, one enters a *XX# code and the switch will call your extension when the called party becomes available. Has somebody already built this in/for Asterisk, otherwise I'll look into it. Michiel Betel ConsultancyAbelenlaan 19 T: +31 20 640 30181185 RT Amstelveen E: [EMAIL PROTECTED]The NetherlandsW: www.betel.nl Confidentiality Notice - The information contained in this e-mail is intended for the named recipient(s) only. It may contain privileged and confidential information, and if you are not the addressee or the person responsible for delivering this to the Addressee, you may not copy, distribute or take action in reliance on it. If you have received this e-mail in error, please notify us immediately by returning the original message to the sender by e-mail!
[Asterisk-Users] srv.c + srv.h
I just downloaded the latetst CVS. A compile now complains about a missing srv.c srv.h used in chan_sip.c. Can they be added? -- Betel Consultancy Abelenlaan 19 1185 RT Amstelveen The Netherlands http://www.betel.nl tel. +31 621 858 469 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ATA losing registration problems solved by setting tftp
For all thos Asterisk users not on the FWD list, it works for me!: -Original Message- From: Free World Dialup - The Future of Dialing [mailto:[EMAIL PROTECTED] On Behalf Of Leonidas Piagkos Sent: donderdag 12 juni 2003 0:58 To: [EMAIL PROTECTED] Subject: Re: [FWD] FWD losing Registration Hi Don, All you have to do with your ATA is to set the following parameters as : UseTftp = 1 CfgInterval = 1800 TftpURL = 192.168.0.x (any internal IP) This parameters enables the auto-provisioning of your ATA device though a TFTP server every 30 minutes. This operation helps your ATA to be always alive and of caurse solves your problem :) Even If you dont have any TFTP server installed on your local network, your ATA will try to provision itself from the IP that you enter on TftpURL and always stays alve :))) Of caurse, this is a way that I found (after thousands of hours of research and telephone speaking with Cisco support), it does not solves realy the 'bug' of ATA's formware (in all versions). All it does, is that make a simple automatic-reset on your device every 1800 seconds (30 minutes). And this because its time the ATA provisions itself, makes and a reset also!!! End even if 'hungs-up' again in between 30minutes, it will be fine after the next auto-provisioning You cannot imaging how many hours I spend from my life in the telephones, for describing this problem of ATA to the Cisco Tech Support. And from then, Cisco had release many new firmwares for ATAs... But until now, I never see something relative on the 'fix - lists' of eatch release Who knows Im still so young... :( Enjoy Leonidas Piagkos -Original Message- From: Free World Dialup - The Future of Dialing [mailto:[EMAIL PROTECTED] Behalf Of Russell, Don Sent: Wednesday, June 11, 2003 7:04 PM To: [EMAIL PROTECTED] Subject: Re: FWD losing Registration I have a similar problem with a Cisco 186 turning on debug trace on the ATA, shows that it gets a 403 Forbidden reply from fwd.pulver.com When that happens, the ATA stops registering at RegInterval seconds. When I reset the ATA ( via http: //ata ip address/reset ) then everything is OK (for a while) I can make outgoing calls, but not incoming because my firewall closes up if the registration is not periodic. Not much help here, but you're not alone, and maybe I gave you something to look into (403 Forbidden?) Don Russell -Original Message- From: W Hills [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 10, 2003 9:58 PM To: [EMAIL PROTECTED] Subject: [FWD] FWD losing Registration With a Grandstream Sip phone I lose registration after about one day ,sometimes less .This means that I am no longer visible as being on the FWD network and cannot receive calls .I have spoken to people with an ATA and some of them have a similar problem yet others do not . Is this an issue with the FWD servers or with the outbound Proxy and is this something that will get fixed any time soon ? The people with this problem need to reboot their devices often to be able to be registered on FWD to receive calls and this is obviously not ideal . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN4Linux + Asterisk and Europe
My Fritz paasive PCI hasn't crashed so far and works fine, relatively low latency so not too much echo. However for professional use, get an active CAPI card so you can use the CAPI echo supp. routines. Michiel Oliver Brandt said: On Mon, Jun 02, 2003 at 12:33:06AM +0200, Piotr Adamiak wrote: Hello, Anyone on this group using / implementing * and hardware certified for use in Europe ? I believe that ISDN4Linux cards mostly have telecomm certificates, so using them should be safe on the client side. Are there any major issues / problems associated with using such cards with * ? I am talking about a small / very small office with single - few lines. I tried ISDN4Linux but I had the problem that high voices were recognized as DTMF signal wich ended up in beping through the whole call. I belive there is a patch out (maybe eve imcluded in the regular asterisk code) but I have not tried it. I'm using chan_capi and since I swiched to an AVM B1 it works great. With the AVM Fritz (passive) it's suppose to work but it actually cause my whole system to crash every once in a while... Just buy a B1 or so at ebay and you should be fine. CU Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] manager interface change request
I concur! It would also help in parsing out the occasional junk I get on the socket. (I'm currently writing a wxwindows version of gastman) Also... I'm still not sure wheter I can be absolutely sure that the Responses will always be in the correct order... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roy Sigurd Karlsbakk Sent: vrijdag 30 mei 2003 10:29 To: Asterisk mailing list Subject: [Asterisk-Users] manager interface change request hi all I'm trying to use the manager interface to do some nagios (http://nagios.org/) integration, and I find some parts of it not really optimal. What I'd like to change, is to make \r\n\r\n an actual terminator, something it isn't today, AFACS. Below is the Status output - it shows Response, Message, \r\n, Status post, \r\n, Status post etc etc. Without a parsable terminator, I need to use some select/poll interfaces, and I just don't like that :P May I suggest changing the \r\n between status (and other) output sections to something like '---\r\n'? regards roy action: status Response: Success Message: Channel status will follow Event: Status Channel: CAPI[contr2/22545070] CallerID: 22545070 State: Up Link: MGCP/aaln/[EMAIL PROTECTED] Event: Status Channel: MGCP/aaln/[EMAIL PROTECTED] CallerID: 22545070 State: Up Context: default Extension: 98013356 Priority: 1 Link: CAPI[contr2/22545070] -- Roy Sigurd Karlsbakk, Datavaktmester ProntoTV AS - http://www.pronto.tv/ Tel: +47 9801 3356 Computers are like air conditioners. They stop working when you open Windows. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CE certification for Europe
sigh And all I actually did was ask if anybody else was interested to share costs of the european certification of the (note!) digium cards. Haven't seen any replys on that :-( I'll call digium on Monday to discuss on how to proceed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gregg Lebovitz Sent: zondag 6 april 2003 0:03 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] CE certification for Europe dwayne, I didn't interpret Tilgman's email as instigating this kind of response. You may be smart, but I really don't buy any of your arguments. I hope someone at your company has better business/PR sense than you. Otherwise, you are doomed. Gregg On Sat, 2003-04-05 at 15:00, d hinton wrote: to tilghman: Then contribute already. Don't troll the list you bozoo, before you call someone a troll you should prob read the post. and if that's you opinion arfter that then fine, so be it. but it's always some punk, who got bullied on as a kid that hides behind the internet and slings pot-shots ;-( And don't volunteer to sell Digium's cards at a price that severely undercuts their product (and revenue and development) THESE ARE NOT DIGIUMS CARDS they are GPL'ed released by the zapata project: http://zapatatelephony.org try reading and learning before you write. and further more i never said that i would sell to the public, just that we COULD make these cards and sell them for $850 and still turn a profit. it was a hint to digium. in fact, the most expensive part only cost $100.00 USD. see http://www.maxim-ic.com/index.cfm If you can't afford $1500 for a card, you jolly well aren't going to be able to afford a T1 line anyway, so having the card would be completely irrelevant at the price digium charges for the CARDS, they would cost the most of the whole project. for example: i can get a local T1 for ~$250/month with 100 DID's for $56 dollars extra. and a FULL T1 backbone for $395/month see: http://www.theplanet.com/solutions/access.php in these hard eco times we should ALL be good shepards of our money. ALSO for all of you who keep comparing the zapata cards to intel's dialogic cards please read up on the subject first, it's like comparing apples and oranges. the best reason to chose the zapata IMHO, is the fact that their cheap good cards that you can buy (or make) alot of them and provide redundancy and at lot of coverage area because of the low upfront cost. THANK GOD FOR THE GPL MOVEMENT. TO MARK thanks for the offer but, yes we are to far into the making our own process, sorry. dwayne - Original Message - From: Tilghman Lesher [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, April 05, 2003 12:47 PM Subject: Re: [Asterisk-Users] CE certification for Europe On Saturday 05 April 2003 12:03, d hinton wrote: look, i like this project, and hope to contribute to it in the near future, Then contribute already. Don't troll the list. And don't volunteer to sell Digium's cards at a price that severely undercuts their product (and revenue and development), as the GNU telephony movement greatly benefits from Digium's continued development and resources (including this list). but i just believe that making the cards more affordable ads more value, It's already more affordable than any other telephony card. The value is undisputed by everyone here, except for you. by allowing more people to be able to afford to develope on this cards, If you can't afford $1500 for a card, you jolly well aren't going to be able to afford a T1 line anyway, so having the card would be completely irrelevant. and for the global GPL telephony movement to continue to grow. It's growing. Buy from Digium and it'll grow some more. i hope i didn't offend Too late. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Patch to enable syslog logging
Title: Message Oops just noticed my patch removes the changes made in CVS to threadsafe the time routines... I'll create a new diff which compares the latest trees... Michiel
[Asterisk-Users] corrected syslog patch
Title: Message NOTE: This one does not touch localtime_r.. Here's a small patch to logger.ci wrote which enables Asterisk to log to syslog. put a line like below in your logger.conf: syslog = notice,warning,error andAsterisk will write it's logging to /var/log/messages too.. Note that event-log logging is not yet included. (should it be?) Michiel Betel patch.syslog Description: Binary data
RE: [Asterisk-Users] CE certification for Europe
Thats what I specifically asked... Should the whole system be approved (eg. Computer, cards, software) or just the components. The answer I got back from two agencies was that hardware approval under RTTE should be sufficient. Same as ISDN BRI card manufacturers do... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaus-Peter Junghanns Sent: donderdag 3 april 2003 17:58 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] CE certification for Europe Hi d hintion, hmmm...getting approvals for europe isnt that easy. because you get the approval for a combination of hardware and driver software, so when you change the driver you loose the approval. oh yes, sure you can produce the cards and sell them cheaper, but that doesnt take the development time of the zaptel drivers into account. opening up a competition against digium based on their software and GPLed hardware design doesnt sound good to me . rather sounds like M$ style to me. regards kapejod -- Klaus-Peter Junghanns CEO,CTO Junghanns.NET GmbH Breite Strasse 13 - 12167 Berlin - Germany fon:+49 30 79705390 fax:+49 30 79705391 iaxtel: 1-700-157-8753 email: [EMAIL PROTECTED] Am Don, 2003-04-03 um 17.15 schrieb d hinton: you know, i asked diguim guys for this info and got nothing. so my company decided to produce these cards for our own use and have them tested for FCC, CA and CE certs est. for testing we got back, range between $7,500 - $9,500 (USD) for all three. we are also surprised that the quote we got for producing these cards was so cheap, that we could produce them and sell them for just under $850(USD). we believe that this would be more in line with the reason zap tel guy's released the plans GPL. so that average developers could afford the card. if there's enough intrest i'm sure i could get my boss to sponsor a project that provides driver support for wider use of the zapata card and lower cost hardware, unless diguim wishes to do it (HINT). shout out to me before the end of this week, cause our cards go to the manufacturer on 4/10/2003 and the testing starts no soon as the manufacturer sends us the prototype. the total led time we got back was 3-6 weeks. dwayne - Original Message - From: Michiel Betel [EMAIL PROTECTED] To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Sent: Thursday, April 03, 2003 8:18 AM Subject: RE: [Asterisk-Users] CE certification for Europe I called around and got some rough quotes for RTTE testing and certification for europe. It seems to boil down to euro 2400,- per card to be tested. They would also need the tech. doc and design from digium... Any european users want to help? I'd like to be able to legally use the E1 cards Michiel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vlasis Hatzistavrou Sent: woensdag 2 april 2003 15:25 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] CE certification for Europe Thank you very much for your reply and for clarifying this point. Does anyone know if there is any effort for approval of the boards as communications equipment in Europe then? Best regards, Vlasis. Klaus-Peter Junghanns wrote: Hi Vlasis, CE is no certification, it is just a decleration of conformity from the manufacturer. It has nothing to do with getting an ITU / ETSI (whatever...) approval for communication equipment. regards kapejod -- Klaus-Peter Junghanns CEO,CTO Junghanns.NET GmbH Breite Strasse 13 - 12167 Berlin - Germany fon:+49 30 79705390 fax:+49 30 79705391 iaxtel: 1-700-157-8753 email: [EMAIL PROTECTED] Am Die, 2003-04-01 um 10.13 schrieb Vlasis Hatzistavrou: Hello, I'd like to ask if there are any news about CE certification of the E1 boards. I know that the T1 boards are FCC certified but I'd also like to know what is the status for CE certification. Thanks for any input, Vlasis Hatzistavrou. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL
RE: [Asterisk-Users] CE certification for Europe
As far as I can find out (till now that is) all it actually takes (since 2001) is conforming to the RTTE As in http://www.radio.gov.uk/topics/conformity/document/rtte/rtteman/rtteman.htm Annex 4 states that An application for an Opinion should be accompanied by (amongst others): Version of any software or firmware supplied with the equipment which may affect compliance with the RTTE must be declared. However, and thats the big difference since this RTTE came into effect: Unlike the previous Telecommunications Terminal Equipment Directive (TTE Directive 91/263/EEC and 98/13/EC) in which each compliance procedure included a third party continuing compliance element, there is no formal third party continuing compliance requirement in Annex IV of the RTTE Directive. However, the manufacturer does have a responsibility for ensuring continuing compliance. This requirement is invoked by Annex II. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Underwood Sent: donderdag 3 april 2003 18:29 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] CE certification for Europe Unless things have simplified since I was last involved in European approvals (which is quite a long time) things are worse than that. If your factory has not previously produced approved telecoms products, you probably need to pay for a factory inspection; each new protocol you want to support needs its own approvals testing of the software; the software drivers must be locked down against uncontrolled changes; your own changes require some level of reapproval; etc. The list can get quite long and painful, unless you are producing a series of products and can get into the proper swing of things. If you only want CTR4 the protocol list might not be a problem. On the driver side you can look at the i4l stuff and see what they had to do to get a driver through approvals for dumb BRI ISDN cards - and every tiny change means some level of reapproval. The US used to be comparable, but these days approval there may not even be necessary. It depends how you read the rules. Approving the hardware certainly makes life easier, though. Getting UL and FCC approval for the hardware seems to be all that is needed. The protocols don't seem to need any approvals. The figures the original poster quoted seem much cheaper than any real approval I have seen go through. It sounds like he hasn't been through the approvals minefield before. It can be a slow and costly place to navigate for the beginner. Regards, Steve Klaus-Peter Junghanns wrote: Hi d hintion, hmmm...getting approvals for europe isnt that easy. because you get the approval for a combination of hardware and driver software, so when you change the driver you loose the approval. oh yes, sure you can produce the cards and sell them cheaper, but that doesnt take the development time of the zaptel drivers into account. opening up a competition against digium based on their software and GPLed hardware design doesnt sound good to me . rather sounds like M$ style to me. regards kapejod ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Nightsettings
Based on James suggestion to use the DB functions I made the following and thought it might be nice to share: ; exten = s,1,DBget($Night=GlobalSettings/Night) ; if not night jump to +101 exten = s,2,Goto(closed,s,1) ;Night has been set, we're closed exten = s,102,Goto(open,s,1) ;Night has not been set so we are open ; ; night settings ; calling 6502 toggles the Night-settings ; exten = 6502,1,Authenticate(/etc/asterisk/password.conf) exten = 6502,2,DBget($Night=GlobalSettings/Night) ; if not night then set it exten = 6502,3,DBdel(GlobalSettings/Night) exten = 6502,4,Playback(night_off) exten = 6502,5,Hangup exten = 6502,103,DBput(GlobalSettings/Night=true) exten = 6502,104,Playback(night_on) exten = 6502,105,Hangup ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] syslog patch
Hi I've written a small patch to logger.c which enables * to log to syslog ased on a setting in the logger.conf file. However, Asterisk uses the same macros as sys/sylog does. Changing the asterisk defines won't help since they also do the macro expansion to filename linenumber. Not being an experienced C programmer what is the normal way to solve this? Change all asterisk LOG_WARNING/NOTICE/ERROR to AST_LOG_WARNING? or include a modified version of /sys/syslog? or just ignore the overwrites, meaning logger.c can't itself call ast_log? Michiel -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] night setting
Hi, We want to be able to switch to a night context when the last person leaves the office. (by having this person call a special extension) This context would then disallow anything but local outbound calls and wil handle inbound calls differently. Currently we use includes with a set time, but thats not ideal. I could do an agi which looks at a variable, checking the night-setting but that would mean it gets called for every in outbound call and I'm a bit afraid of the overhead involved. The other option I see is modifying the include functions in pbx.c to look at a settable variable or Asterisk database value. But helpfull suggestionsare very welcome! Michiel Betel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How could I get * from CVS if I am not on the Linux platform?
http://www.wincvs.org/download.html or http://www.cvshome.org/cyclic/cvs/windows.html haven't tried them myself but know people using them Michiel Betel it said: Hi,I want to get the latest asterisk code from CVS. But the computer OS I used for travelling internet is Windows. I don't know how to I deal with the CVS. Thanks. john -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ATA186 v2.15mm help...
You have the skinny/mgcp firmware loaded, not the SIP one... Get ata18x-v2-15-020927a-2.zip from cisco (or contact me off-list) Michiel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Archer Sent: vrijdag 14 maart 2003 22:46 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ATA186 v2.15mm help... Hi All... I have a brand new ATA186 with the following firmware: Version: v2.15.ms ata186 (Build 020919a) I have been through the archives about how to configure it, but my colorful configuration web page does not have the same fields that people say I need to adjust. Even the examples on Cisco's web site don;t match. For example, I don't have the GtkOrProxy field, which is probably an important one. Am I missing something very simple? Thanks... Jim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ATA186 MGCP or SIP?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Archer Sent: zaterdag 15 maart 2003 9:51 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ATA186 MGCP or SIP? Yup, that's the problem. So now the question is, do I convert to SIP or stick with MGCP? I have MGCP working but, as reported by someone else, transfer does not work. If I convert to SIP can I go back? Yes, just load the other firmware again, SIP transfers work with the #,T method. Also, I have noticed when I dial from one extension to another the quality is good, but when I dial out it is horrific. I wonder if this is because my CPU is too slow? I have a 1.1GHz on order, but until it comes in I am testing on an old 266MHz. 266Mhz is too slow, voiceprompts get choppy, and codec conversion is not fast enough I would appreciate any suggestions. --On Saturday, March 15, 2003 9:15 AM +0100 Michiel Betel [EMAIL PROTECTED] wrote: You have the skinny/mgcp firmware loaded, not the SIP one... Get ata18x-v2-15-020927a-2.zip from cisco (or contact me off-list) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] chan_capi: advice needed on isdn card
Note that the CAPI driver for the Fritz! only supports a single card. So if you want to expand your asterisk in future you can't just add a 2nd cheap card... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaus-Peter Junghanns Sent: zaterdag 15 maart 2003 9:31 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] chan_capi: advice needed on isdn card Morning Chris, 1. it's not predictable if you will get echo with the passive AVMs (and probably also not predictable with the actives). 2. yes. 3. the passive Eicons (those without the word server in the name) have no echo cancelation and not even a capi driver (so you're stuck with i4l). but not all active Eicons support echo cancelation! Bicster: can you shed some light on this? :-) regards kapejod -- Klaus-Peter Junghanns CEO,CTO Junghanns.NET GmbH Breite Strasse 13 - 12167 Berlin - Germany fon:+49 30 79705390 fax:+49 30 79705391 iaxtel: 1-700-157-8753 email: [EMAIL PROTECTED] Am Fre, 2003-03-14 um 22.20 schrieb Chris Wetemans: Since chan_capi now supports echo_cancellation on (some) eicon cards, i'm considering buying another isdn-card. On the moment I'm using an old Teles card with isdn4linux, but i get a terrible echo when calling analog counterparts, and the delay is also quite heafty. 1. If I get a (cheap) AVM-card (Fritz), and use CAPI, would the delay (latency) be so small that an echo isn't noticeable anymore? 2. If the echo would still be noticeable would an EICON-card with echo cancellation on board help a lot? 3. Which EICON-cards have echo cancellation and linux CAPI-support, the cheaper client cards( DIVA Pro, DIVA+CT,...) or the expensive Server cards (BRI-server, BRI-server voice, ) Thanks, Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] True believer
Title: Message Hi All, I am now a true believer in Asterisk... I just made a call which went like this: Analog Phone - ATA-186 --SIP-- home asterisk --IAX over DSL 1024/512-- office asterisk -- digium E1 -- PSTN -- gsm cellphone whilst the gsm user who had no idea of all codec conversions involved compimented me on the sound quality! Michiel
[Asterisk-Users] ATA beginners question
Title: Message
When dialing a port
on my ATA-186 I get:
== Spawn extension
(default, s, 1) exited non-zero on 'SIP/ata1-1-0c77' --
Executing Macro("SIP/ata1-2-4fc0", "stdexten|6200|SIP/ata1-1") in new
stack -- Executing Dial("SIP/ata1-2-4fc0",
"SIP/ata1-1|30") in new stack -- Called
ata1-1 -- Got SIP response 488 "Not Acceptable Here" back
from 192.168.1.100 == No one is available to answer at this
time
Instead of a ringing
telephone... Both ata1-1 and ata1-2 are registered with
asterisk.
Can anyone tell me
how to get rid of the 488?
Thanks!
RE: R: [Asterisk-Users] Cisico ATA licence
Err... I just wanted a Cisco ATA and did not want to start a war :-( -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: vrijdag 7 maart 2003 0:11 To: [EMAIL PROTECTED] Subject: Re: R: [Asterisk-Users] Cisico ATA licence On Thursday 06 March 2003 12:42, Florian Overkamp wrote: While they may not prosecute an individual for having loaded an unlicensed stack on the hardware, it is unwise to suggest it in a publicly available and archived list. Do remember here in the US we have to now worry more about John Ashcroft than the company whose software we use/abuse since John can bring charges on his own without the company. Fine. Luckily, not all of us are in the US. Michiel and I can happily toy around with cisco firmware :-) You haven't been paying attention to world news recently, have you? George, John, Donald, and the rest of the gang think they can invade any country anytime they want (see Afghanistan, Iraq). -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisico ATA licence
I can buy a new ATA186 here, but it is sold with a 1-port user license UK, for euro 192, but does that license stop me from using both ports? I can't read the license agreement till I buy the thing, so I don't know what i'm buying... Michiel -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: [Asterisk-Users] Cisico ATA licence
Thanks! Is there a safe way to identify (cisco secret part number or something) what SIP loaded ATA to order, or should I call Cisco? I don't really trust the mailorder company guys to sort it out for me as they probably don't sell that many of these units and will probably go uh??? on me if I start questioning Steven Critchfield said: On Thu, 2003-03-06 at 04:25, Matteo Brancaleoni wrote: the license is needed only with cisco callmanager. so you can ignore it and use both ports with asterisk ;-) Thats wrong according to the debates here and on the FWD mailing list. The unit that Michiel was looking at contains software that connects to the Cisco Call Manager, probably using skinney. What Michiel needs is one with the SIP or H323 software load on it. The units with SIP or H323 loaded on it usually have the license for both ports to use that software. -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Michiel Betel Inviato: giovedì 6 marzo 2003 10.43 A: [EMAIL PROTECTED] Oggetto: [Asterisk-Users] Cisico ATA licence I can buy a new ATA186 here, but it is sold with a 1-port user license UK, for euro 192, but does that license stop me from using both ports? I can't read the license agreement till I buy the thing, so I don't know what i'm buying... Michiel -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call recording
I don't know (haven't tried myself) but Kostya V. Ivanov's 'R' patch to the dial application (december 2002) might be of help for you. Check the archives for Barge (Intrusion) Capabilities. It might be some manual work to apply after all the allmost daily CVS changes but worth a try! Michiel Betel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian J. Schrock Sent: woensdag 5 maart 2003 17:55 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Call recording Hello, How would I go ahead a record all phone calls into and out of my asterisk server. I know the legality issues behind it, but I could always play a recording to let people know they will be recorded. Brian J. Schrock Network Engineer, RHCE, CCNA Anistone Technologies Phone: 614-537-2817 FAX: 614-573-7165 6926 Avery Rd. Dublin, OH 43017 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Interest in E1 channel banks?
Very interested! Are you planning for european certification?(expensive!) If so even more interested! Florian Overkamp said: At 18:45 27-2-2003 +1100, you wrote: Our company manufactures an E1 channel bank that is approved for use in Australia (it should also be compatible with Euro standards). It is modular and available in 10, 20 or 30 analog port configurations. Signal monitoring and configuration is via Ethernet. These units are manufactured in low quantities for specific telco requirements. However if there was enough interest, we would be able to manufacture and sell the units at pricing levels under US $2000. So how much interest is out there? /me raises hand (well, we've done our current infra, but it may be a consideration none the less - at these levels of pricing) Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
