[asterisk-users] Re: 1.4 branch on OSX?
On 2006-10-21 22:20:51 -0700, Joshua Colp [EMAIL PROTECTED] said: Okay folks, give the latest 1.4 branch a try. I spent some time this morning isolating the issue and think I have it. OK! Thanks Josh, that builds and seems to work a bit, but it's easting my whole CPU... Any ideas on how to figure out why? Still, progress! Is it during calls, just sitting there at the CLI... Just sitting at the command line it's using 80% of my mighty G3/500 ;~) how is it being started, I am starting it manually at this point as my launchd thing is another issue. I have been using asterisk -v start it. if during calls - what are you doing? Doesn't change during calls, and the audio seems to be ok on the few calls I have tried. I have a Mac Mini here now that I'm using for Asterisk development so I can lab it up and hopefully isolate the issue. As well - is tab completion working for you? I received a report from another individual saying it just hung for him but it didn't for me... quite odd. Even odder my console seems to be taking commands but doesn't respond. Except exit works? Tab completion is not working here either. But the fact the console doesn't except any fully typed commands is more worry some. Let me know if I can help with any info. Marty ___ --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] Unique call ID's across several systems
Conrad Wood wrote:
On Sat, 2006-10-21 at 19:16 +0100, Julian Lyndon-Smith wrote:
hi guys. Is there anyway of generating a universal / global unique id
from the dialplan (A uuid or guid). I want to have several asterisk
servers sharing a cdr database, and want a unique reference for each
call. Obviously, ${UNIQUEID} doesn't work across several * systems/
couldn't you set a variable in the local dialplan and combine ith with
uniqueid?
e.g.
LOCALID=asterisk1
and then
Set(GLOBALID=${LOCALID}-${UNIQUEID}) or so?
Doh. Sometimes things are just so simple I seem to miss them. ;)
Thanks for the help.
Julian
or even use the hostname of the machine?
Conrad
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Re: [asterisk-users] Unique call ID's across several systems
Matt Florell wrote:
For several releases now the BRIStufft version of Asterisk will
prepend the hostname of the server to the beginning of the base
uniqueid by default and use that new value as the uniqueid of the call
everywhere it is used. This shouldn't be that difficult to do in
regular Asterisk.
Is this the case for the cdr record in an odbc database ?
Julian
MATT---
On 10/21/06, Julian Lyndon-Smith [EMAIL PROTECTED] wrote:
hi guys. Is there anyway of generating a universal / global unique id
from the dialplan (A uuid or guid). I want to have several asterisk
servers sharing a cdr database, and want a unique reference for each
call. Obviously, ${UNIQUEID} doesn't work across several * systems/
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[asterisk-users] new g.729a codecs for asterisk 1.2/1.4 and glibc
Hello! It seems that the new codec is not backward-compatible to glibc 2.1/2.2 so I receive the following error: [codec_g729a.so]Oct 22 02:37:18 WARNING[3433]: loader.c:325 __load_resource: /u sr/local/glibc/libc.so.6: version `GLIBC_2.3' not found (required by /opt/files/ usr/lib/asterisk/modules/codec_g729a.so) Oct 22 02:37:18 WARNING[3433]: loader.c:554 load_modules: Loading module codec_g 729a.so failed! Ouch ... error while writing audio data: : Broken pipe I use some licenses on glibc 2.2 systems (I can't change that in the moment)! -- regards Holger Hornung mailto:[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] bristuff-0.3.0-PRE-1u for Asterisk 1.2.13 on junghanns downloads now
On Sat, 21 Oct 2006, Michiel van Baak wrote: On 20:15, Sat 21 Oct 06, Tzafrir Cohen wrote: Interesting. Latest bristuff chenges the default Zaptel echo canceller to MG2 (which is also the recommendation of Digium now). BTW: as an alternative to zaphfc+flotz, consider vzaphfc. It seems that the only place from which you can download an up-to-date version nowadays is the Debian zaptel package: http://svn.debian.org/wsvn/pkg-voip/zaptel/trunk/vzaphfc/ http://packages.debian.org/zaptel-source Tzafrir, Are you in the position to get stuff into asterisk ? The patch you sent me is working great. I think we need this stuff into the normal asterisk. BRI is something a lot of asterisk users depend on, how odd this may sound to USA ppl. Which patches are that vzaphfc? I'm interested too :) ___ --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] bristuff-0.3.0-PRE-1u for Asterisk 1.2.13 on junghanns downloads now
On Sun, Oct 22, 2006 at 12:56:18PM +0200, Remco Barendse wrote: On Sat, 21 Oct 2006, Michiel van Baak wrote: On 20:15, Sat 21 Oct 06, Tzafrir Cohen wrote: Interesting. Latest bristuff chenges the default Zaptel echo canceller to MG2 (which is also the recommendation of Digium now). BTW: as an alternative to zaphfc+flotz, consider vzaphfc. It seems that the only place from which you can download an up-to-date version nowadays is the Debian zaptel package: http://svn.debian.org/wsvn/pkg-voip/zaptel/trunk/vzaphfc/ http://packages.debian.org/zaptel-source Tzafrir, Are you in the position to get stuff into asterisk ? The patch you sent me is working great. I think we need this stuff into the normal asterisk. BRI is something a lot of asterisk users depend on, how odd this may sound to USA ppl. Which patches are that vzaphfc? I'm interested too :) vZapHFC is Daniele Vihai Orlandis take on ZapHFC. However over the time Daniele decided that he needs to V a bigger target and started vISDN. vZapHFC is now maintained by Jens Wilke. Though appears nowhere on his homepage ( http://wilke.org/jens/ ). If it works for you, ask him to post it somewhere... -- Tzafrir Cohen iax:[EMAIL PROTECTED]/tzafrir icq#16849755 mailto:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] new g.729a codecs for asterisk 1.2/1.4 and glibc
On Sun, Oct 22, 2006 at 12:03:08PM +0200, Holger Hornung wrote: Hello! It seems that the new codec is not backward-compatible to glibc 2.1/2.2 so I receive the following error: [codec_g729a.so]Oct 22 02:37:18 WARNING[3433]: loader.c:325 __load_resource: /u sr/local/glibc/libc.so.6: version `GLIBC_2.3' not found (required by /opt/files/ usr/lib/asterisk/modules/codec_g729a.so) Oct 22 02:37:18 WARNING[3433]: loader.c:554 load_modules: Loading module codec_g 729a.so failed! Ouch ... error while writing audio data: : Broken pipe I use some licenses on glibc 2.2 systems (I can't change that in the moment)! What OS? What distribution? Self-built glibc? Installed in /usr/local? -- Tzafrir Cohen iax:[EMAIL PROTECTED]/tzafrir icq#16849755 mailto:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Strange FXS disconnection problem.
Hi Guys, Sorry to prod this again, but just wondering if anyone had any thoughts? If I'm not supplying useful information, please let me know what I can do to make the debug easier. Very best regards, Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Bath Sent: 15 October 2006 01:53 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Strange FXS disconnection problem. Hi there, Thanks for the reply. I pasted everything from the console, with logger set to all, and verbose set to around 15. (as asked in original email) I have the whole session, from placing the call to the call hanging up, but it's pretty long, so I wasn't sure if you wanted to full thing. I'm sorry if I've misunderstood what you asked for - would you like it all? If not, would you mind please clarifying what I should post? Many thanks for your patience, Cheers, Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: 15 October 2006 01:35 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Strange FXS disconnection problem. On Fri, Oct 13, 2006 at 09:05:54PM +0100, David Bath wrote: Hey all, Just as an update, incoming calls are fine. I have had several long calls today inbound on the PSTN with no drops. From the log it does sort of look like a hangup is being detected.. but its certainly not correct! Could anyone help me debug this a little further? As you probably recall, I have asked you to post debug log snippets just above the ones you posted. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.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 ___ --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
[asterisk-users] Using variable as a context extension ?
Hi,
Is it possible to use a variable as a context extension? For exemple:
[some-context]
exten = s,1,Background(some_prompt)
exten = ${key1},1,Noop(User pressed ${key1})
exten = ${key2},1,Noop(User pressed ${key2})
If now anyone can suggest how I could achieve this?
--
Andre Courchesne
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Re: [asterisk-users] Unique call ID's across several systems
On 10/22/06, Julian Lyndon-Smith [EMAIL PROTECTED] wrote:
Matt Florell wrote:
For several releases now the BRIStufft version of Asterisk will
prepend the hostname of the server to the beginning of the base
uniqueid by default and use that new value as the uniqueid of the call
everywhere it is used. This shouldn't be that difficult to do in
regular Asterisk.
Is this the case for the cdr record in an odbc database ?
Yes, I even had to modify the database schema I was using to allow for
the longer uniqueids. It is changed in the code of Asterisk somewhere
I just never went in to find the location of the change. Just wanted
to mention that is how the BRIstufft version does uniqueid and it does
allow for a truely unique identifier across multiple machines.
MATT---
Julian
MATT---
On 10/21/06, Julian Lyndon-Smith [EMAIL PROTECTED] wrote:
hi guys. Is there anyway of generating a universal / global unique id
from the dialplan (A uuid or guid). I want to have several asterisk
servers sharing a cdr database, and want a unique reference for each
call. Obviously, ${UNIQUEID} doesn't work across several * systems/
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Re: [asterisk-users] Using variable as a context extension ?
It should because it is a variable however you would have to set it some
where. Either by creating an extension that sets it or by setting it
staticly.
- Original Message -
From: Andre Courchesne - Consultant [EMAIL PROTECTED]
To: Asterisk Users List asterisk-users@lists.digium.com
Sent: Sunday, October 22, 2006 2:29 PM
Subject: [asterisk-users] Using variable as a context extension ?
Hi,
Is it possible to use a variable as a context extension? For exemple:
[some-context]
exten = s,1,Background(some_prompt)
exten = ${key1},1,Noop(User pressed ${key1})
exten = ${key2},1,Noop(User pressed ${key2})
If now anyone can suggest how I could achieve this?
--
Andre Courchesne
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[asterisk-users] G.729 operating on outgoing only
Greetings list, I have an older Dell Poweredge server running Asterisk 1.2.13. I have installed 5 licenses for G.729 from Digium. I have 5 SIP trunks through a US provider. When my system makes outgoing calls, they go out as G.729. However, when an incoming call comes in, my server does not indicate to the providers server that G.729 is an option, so the remote server sends the call in ULAW. My sip.conf file has both the remote server my calls come from, and the remote server we send calls to listed, with disallow=all then allow=g729, but only the outgoing seems to be doing what its supposed to. Any suggestions? Joel Lansden Solutions Architect [EMAIL PROTECTED] tel 205.533.2039 fax 866.602.9130 digitalparadisesystems http://www.digitalparadise.net Could it be any easier? ___ --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] Re: 1.4 branch on OSX?
On 22 Oct 2006, at 07:02, Martin Joseph wrote: On 2006-10-21 22:20:51 -0700, Joshua Colp [EMAIL PROTECTED] said: Okay folks, give the latest 1.4 branch a try. I spent some time this morning isolating the issue and think I have it. OK! Thanks Josh, that builds and seems to work a bit, but it's easting my whole CPU... Any ideas on how to figure out why? Still, progress! Is it during calls, just sitting there at the CLI... Just sitting at the command line it's using 80% of my mighty G3/500 ;~) how is it being started, I am starting it manually at this point as my launchd thing is another issue. I have been using asterisk -v start it. if during calls - what are you doing? Doesn't change during calls, and the audio seems to be ok on the few calls I have tried. I have a Mac Mini here now that I'm using for Asterisk development so I can lab it up and hopefully isolate the issue. As well - is tab completion working for you? I received a report from another individual saying it just hung for him but it didn't for me... quite odd. Even odder my console seems to be taking commands but doesn't respond. Except exit works? Tab completion is not working here either. But the fact the console doesn't except any fully typed commands is more worry some. I get different results if I run : asterisk -c; # (which works ok - except for the high CPU) and asterisk ; asterisk -r # (where the console doesn't work properly) Tim Panton www.mexuar.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.729 operating on outgoing only
disallow=Ulaw in the trunk conf Best regards, Al Bochter Bochter Services Toll Free: 866-638-1254 EXT: 250 Free World DialUp: 780217 EXT: 250 Cellular: 206-203-5801 http://www.BochterServices.com/?t=Email - - - - we BUY and sell Coins, Silver, Sterling Silver and Gold http://www.bochterservices.com/?j=goldt=email - - - - For new and used security items http://www.bochterservices.com/?j=storet=email_security - - - - 24kt GOLD PLATING http://www.bochterservices.com/?j=platingt=email - - - - Joel Lansden wrote: Greetings list, I have an older Dell Poweredge server running Asterisk 1.2.13. I have installed 5 licenses for G.729 from Digium. I have 5 SIP trunks through a US provider. When my system makes outgoing calls, they go out as G.729. However, when an incoming call comes in, my server does not indicate to the providers server that G.729 is an option, so the remote server sends the call in ULAW. My sip.conf file has both the remote server my calls come from, and the remote server we send calls to listed, with disallow=all then allow=g729, but only the outgoing seems to be doing what its supposed to. Any suggestions? Joel Lansden Solutions Architect [EMAIL PROTECTED] tel 205.533.2039 fax 866.602.9130 digitalparadisesystems http://www.digitalparadise.net Could it be any easier? ___ --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: 0642-6, 10/22/2006 - 10/22/2006 10:17:02 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
[asterisk-users] Re: 1.4 branch on OSX?
On 2006-10-22 09:16:04 -0700, Tim Panton [EMAIL PROTECTED] said: On 22 Oct 2006, at 07:02, Martin Joseph wrote: On 2006-10-21 22:20:51 -0700, Joshua Colp [EMAIL PROTECTED] said: Okay folks, give the latest 1.4 branch a try. I spent some time this morning isolating the issue and think I have it. snip Even odder my console seems to be taking commands but doesn't respond. Except exit works? Tab completion is not working here either. But the fact the console doesn't except any fully typed commands is more worry some. I get different results if I run : asterisk -c; # (which works ok - except for the high CPU) and asterisk ; asterisk -r # (where the console doesn't work properly) YES! Same here. Using the color console I can now see that show channels is deprecated ;~) Seems quite odd to me that the color console would work while the regular console is broken. Tab completion works too. Asterisk is still consuming the whole available CPU. Anybody know how I can figure out what is up with the CPU hogging? I don't seen anything particularly odd in top or in the activity monitor... Marty ___ --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] Re: G.729 operating on outgoing only
On 2006-10-22 07:14:46 -0700, Joel Lansden [EMAIL PROTECTED] said: This is a multi-part message in MIME format. Greetings list, =20 I have an older Dell Poweredge server running Asterisk 1.2.13. I have installed 5 licenses for G.729 from Digium. I have 5 SIP trunks through a US provider. When my system makes outgoing calls, they go out as G.729. However, when an incoming call comes in, my server does not indicate to the provider's server that G.729 is an option, so the remote server sends the call in ULAW. My sip.conf file has both the remote server my calls come from, and the remote server we send calls to listed, with disallow=3Dall then allow=3Dg729, but only the outgoing = seems to be doing what it's supposed to. =20 Any suggestions? Make sure that your [general] section in SIP.conf includes allow G729. ___ --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] checking 'voicemail externally - doesn't work
Can not check voice mail-box externally. I'm trying to log-in externally (from PSTN line) to check my voice-mail so I created context to authenticate log-in ... exten = 7,4,Authenticate(01894546) exten = 7,5,DISA(4789|disa-access) Authentication works OK, I get inside dial none enter my mailbox extension but it doesn't accept my mailbox password even though it is the correct one. It keep asking me for mailbox, password: Executing VoiceMailMain(SIP/pstn-1270-0819a1f0, pstn1270) in new stack -- Playing 'vm-login' (language 'en') -- Playing 'vm-password' (language 'en') -- Incorrect password '123' for user 'tn12701' (context = default) Could the problem be that the user: pstn1270 is truncated 'tn12701' ? When I try to check the mailbox internally for user who doesn't have a mailbox, it works OK: vm_execmain: Specified user '218' not found (check voicemail.conf and/or realtime config). Falling back to authentication mode. -- Playing 'vm-login' (language 'en') -- Playing 'vm-password' (language 'en') Though, when I log-in externally, it doesn't display that message: Specified user '218' not found... It just asking me for password and telling me it is incorrect one; even though it is the correct one. Using latest Asterisk-1.2.13 -- #Joseph ___ --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] embedded asterisk
Dovid B wrote: anyone know if any of these could handle asterisk ? http://www.axotec.com/embedded-server.htm Dovid, Most of them, running with ARMs running at 70mhz wouldn't be very practical. The SPIDER-III could work out better, but even still 200 MIPS isn't exactly impressive... There are much better solutions for running Asterisk on embedded hardware. -- Kristian Kielhofner ___ --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] Using variable as a context extension ?
Actually that would be using call files. Will try it and let everyone know.
--
Message: 25
Date: Sun, 22 Oct 2006 15:01:41 +0200
From: Dovid B [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Using variable as a context extension ?
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; format=flowed; charset=iso-8859-1;
reply-type=response
It should because it is a variable however you would have to set it some
where. Either by creating an extension that sets it or by setting it
staticly.
- Original Message -
From: Andre Courchesne - Consultant [EMAIL PROTECTED]
To: Asterisk Users List asterisk-users@lists.digium.com
Sent: Sunday, October 22, 2006 2:29 PM
Subject: [asterisk-users] Using variable as a context extension ?
Hi,
Is it possible to use a variable as a context extension? For exemple:
[some-context]
exten = s,1,Background(some_prompt)
exten = ${key1},1,Noop(User pressed ${key1})
exten = ${key2},1,Noop(User pressed ${key2})
If now anyone can suggest how I could achieve this?
--
Andre Courchesne
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RE: [asterisk-users] [SOLVED] 1.2.12.1 crashing
On Fri, 2006-10-13 at 10:50 -0600, Joseph wrote: On Fri, 2006-10-13 at 07:27 +0200, Remco Barendse wrote: On Thu, 12 Oct 2006, Eric ManxPower Wieling wrote: Matt Florell wrote: If you downgrade, let us know if it fixes things for you. It's strange that there were so many changes in the 1.2 SVN branch after 1.2.7.1 that seem to be complete changes in how some things operate(like the transcoding optimization mess for Asterisk 1.2.11 and 1.2.12 that was fixed in 1.2.12.1). I wish that such radical changes would not be made in a release branch at the expense of reliabitily. Maybe Digium can run the next release for 7 days on their PRODUCTION Asterisk box before a release. I guess they did, and it probably worked. Then they run it for several months, and if it works they label it Business Edition and actually sell it because they know it will work. What hardware are they testing it with, just Digium cards? Asterisk 1.2.12.1 definitely doesn't run correctly with Sipura 3000, as it crashes on second call to PSTN line. Crashing problem SOLVED. It my case it seems to be related usage of NVFaxdetect. Upgrading from Asterisk-1.0... to Asterisk-1.2... one need to recompile the NVFaxdetect module. So if somebody is upgrading and from Asterisk-1.0... and using NVFaxdetect this information might save you few hours. -- #Joseph ___ --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] Audiocodes MP-20x
Has anyone used the AudioCodes MP-20x? http://audiocodes.com/Objects/Analog_Telephone_Adapter_Series_MP_20X.pdfSeems like a good device, but I can't seem to find anyone actually using them... ___ --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] chan_bluetooth, mobile handset as VoIP terminal?
The Nokia E60 and E61 can do this, however it's over WiFiOn 10/17/06, Brian Candler [EMAIL PROTECTED] wrote: I have been looking at chan_bluetooth, so far being unable to make itcompile with Asterisk SVN trunk. I was wondering about the different ways it can be used. What I have read sofar implies two possibilities:1. Asterisk pretends to be a handsfree unit, and can use the cell phone for placing calls over the mobile network, or answer inbound calls from the mobile network.2. Asterisk pretends to be a phone, and you can use a headset as a VoIP terminal (unfortunately only useful for receiving calls, as headsets don't have keypads)However, the possibility which really interests me is: 3. Can I use my mobile phone as a terminal, originating calls over bluetooth via Asterisk, using the phone's keypad to dial? And answering inbound calls from Asterisk?This makes my mobile phone into a cordless phone replacement - avoiding mobile charges while at home, and being able to receive PSTN and VoIP callsvia Asterisk.I notice BT's Fusion service appears to work in this way -http://www.bt.com/btfusion/ - as it looks like you get a normal mobile phonewhich can route VoIP calls via Bluetooth and DSL when in range of the basestation.So the question is:- has anyone got Asterisk working this way? - what bluetooth profile would the phone need to support to do this?- does that limit me to particular models of mobile phone?If this is possible, it's not clear to me how the phone would know that Iwanted to set up a call over bluetooth rather than over the mobile network. Would I need to load some sort of app onto the phone?Thanks,Brian.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] VoicePulse Connect 4 Channel Limit?
I would worry about using Voicepulse as your primary provider, even if they didn't impose their draconian policies. You could have 20 numbers paying $220/month in your account and you still get only four calls,. However if you were to open 20 voicepulse connect accounts and put one number on each, you would still pay the same $220/month however you could get up to EIGHTY -- thats 20 times more! -- for the SAME EXACT PRICES Also VoicePulse DOES NOT use Tier 1 providers. We notice every week or so during peak hours a very bad degrigation of the voice quality. If you have an IVR and call it from a landline, it will sound like crap. It's the quality of service you would expect from a free provider. Aggrivated to this, when you contact them they try to blame YOU for their issues. They told me I HAD to run PingPlotter (a WINDOWS program, besides the fact this is VoicePulse Connect for Asterisk and Asterisk is software for Linux) which was not possible on a co-located machine. Also we ported a bunch of phone numbers and the DTMF does not work. If you dial 5551212 VoicePulse might recognise and pass to us 55112 and again instead of trying to troubleshoot the issue (from the SAME phone it always produced CONSISTANT behavior -- the ported number does not accet DTMF correctly, assigned # work!) they blame us and the phones we use. I went as far as going to Sprint PCS store and EVERY CDMA phone in the store would produce the same result! In the end, don't bother with VoicePulse. The quality of the service and the support and just the treatment you get is not worth the price. For $11/month per number and their draconian channels and also billing policy (I wont even get into that) I expect a PREMIUM service and they deliver something about par for a free service. Here's some typical behavior from their servers: ug 24 14:55:48 NOTICE[13944]: chan_iax2.c:7813 iax2_poke_noanswer: Peer 'voicepulse01' is now UNREACHABLE! Time: 71 Aug 24 14:55:59 NOTICE[13944]: chan_iax2.c:7145 socket_read: Peer 'voicepulse01' is now REACHABLE! Time: 1059 Aug 24 15:11:05 NOTICE[13944]: chan_iax2.c:7813 iax2_poke_noanswer: Peer 'voicepulse01' is now UNREACHABLE! Time: 40 Aug 24 15:11:15 NOTICE[13944]: chan_iax2.c:7145 socket_read: Peer 'voicepulse01' is now REACHABLE! Time: 49 Aug 21 15:33:08 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 39 Aug 21 15:33:18 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 43 Aug 22 13:40:41 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 45 Aug 22 13:40:52 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 1064 Aug 22 16:57:15 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 41 Aug 22 16:57:25 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 43 Aug 23 11:02:36 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 43 Aug 23 11:03:49 NOTICE[13944] chan_iax2.c: Peer 'voicepulse02' is now UNREACHABLE! Time: 37 Aug 23 11:08:19 NOTICE[13944] chan_iax2.c: Peer 'voicepulse02' is now REACHABLE! Time: 39 Aug 23 11:08:26 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 39 Aug 23 11:16:31 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 46 Aug 23 12:10:25 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 1246 Aug 23 14:01:37 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 45 Aug 23 14:04:27 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 258 Aug 23 15:28:36 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 44 Aug 23 15:28:37 NOTICE[13944] chan_iax2.c: Peer 'voicepulse02' is now UNREACHABLE! Time: 39 Aug 23 15:31:27 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 41 Aug 23 15:31:27 NOTICE[13944] chan_iax2.c: Peer 'voicepulse02' is now REACHABLE! Time: 56 Aug 23 15:40:31 NOTICE[13944] chan_iax2.c: Peer 'voicepulse02' is now UNREACHABLE! Time: 40 Aug 23 15:43:31 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 43 Aug 23 15:49:42 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 42 Aug 23 15:57:22 NOTICE[13944] chan_iax2.c: Peer 'voicepulse02' is now REACHABLE! Time: 40 Aug 23 16:37:49 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 44 Aug 23 16:38:00 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 48 Aug 23 17:31:07 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 43 Aug 23 17:32:17 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 42 Aug 23 17:34:21 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 42 Aug 23 17:34:31 NOTICE[13944] chan_iax2.c: Peer 'voicepulse02' is now UNREACHABLE! Time: 40 Aug 23 17:36:01 NOTICE[13944] chan_iax2.c: Peer 'voicepulse02' is now REACHABLE! Time: 39 Aug 23 17:36:11 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01'
Re: [asterisk-users] noise gate for asterisk?
Whar sort of interface are you using? Whats at both ends of the calls? Are you sure that noise is generated by Asterisk itself and not by any of the interfaces?On 10/20/06, Lenz [EMAIL PROTECTED] wrote:Hi list, I have a client with a strange requirement: putting a noise gate on theAsterisk channel. For those who are not familiar with them, noise gatesare used in musical instruments to avoid entering low-level noise into the amp system. What they basically do is, they measure the volume of thechannel, and when it's too low they just let the channel close, i.e sendperfect silence, therefore killing low-level buzzing sounds. My client has such a need because they have analog voice-operated push-to-talkhalf-duplex devices on the other side, and low level noise from theAsterisk side will keep the channel open.I will try diminishing the TXgain, but I wondered if there were other options too.l.--Loway Research - Home of QueueMetricshttp://queuemetrics.loway.it___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] Re: G.729 operating on outgoing only
That did it! Thank you very much!! ~Joel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Joseph Sent: Sunday, October 22, 2006 11:56 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: G.729 operating on outgoing only On 2006-10-22 07:14:46 -0700, Joel Lansden [EMAIL PROTECTED] said: This is a multi-part message in MIME format. Greetings list, =20 I have an older Dell Poweredge server running Asterisk 1.2.13. I have installed 5 licenses for G.729 from Digium. I have 5 SIP trunks through a US provider. When my system makes outgoing calls, they go out as G.729. However, when an incoming call comes in, my server does not indicate to the provider's server that G.729 is an option, so the remote server sends the call in ULAW. My sip.conf file has both the remote server my calls come from, and the remote server we send calls to listed, with disallow=3Dall then allow=3Dg729, but only the outgoing = seems to be doing what it's supposed to. =20 Any suggestions? Make sure that your [general] section in SIP.conf includes allow G729. ___ --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] VoicePulse Connect 4 Channel Limit?
So, What´s your recommendation for a production environment? I was looking for good prices, good voice quality for USA Origination and I´d like to hear about good experiences thanks in advance.. R.R. Libera Andrew Joakimsen escribió: I would worry about using Voicepulse as your primary provider, even if they didn't impose their draconian policies. You could have 20 numbers paying $220/month in your account and you still get only four calls,. However if you were to open 20 voicepulse connect accounts and put one number on each, you would still pay the same $220/month however you could get up to EIGHTY -- thats 20 times more! -- for the SAME EXACT PRICES Also VoicePulse DOES NOT use Tier 1 providers. We notice every week or so during peak hours a very bad degrigation of the voice quality. If you have an IVR and call it from a landline, it will sound like crap. It's the quality of service you would expect from a free provider. Aggrivated to this, when you contact them they try to blame YOU for their issues. They told me I HAD to run PingPlotter (a WINDOWS program, besides the fact this is VoicePulse Connect for Asterisk and Asterisk is software for Linux) which was not possible on a co-located machine. Also we ported a bunch of phone numbers and the DTMF does not work. If you dial 5551212 VoicePulse might recognise and pass to us 55112 and again instead of trying to troubleshoot the issue (from the SAME phone it always produced CONSISTANT behavior -- the ported number does not accet DTMF correctly, assigned # work!) they blame us and the phones we use. I went as far as going to Sprint PCS store and EVERY CDMA phone in the store would produce the same result! In the end, don't bother with VoicePulse. The quality of the service and the support and just the treatment you get is not worth the price. For $11/month per number and their draconian channels and also billing policy (I wont even get into that) I expect a PREMIUM service and they deliver something about par for a free service. Here's some typical behavior from their servers: ug 24 14:55:48 NOTICE[13944]: chan_iax2.c:7813 iax2_poke_noanswer: Peer 'voicepulse01' is now UNREACHABLE! Time: 71 Aug 24 14:55:59 NOTICE[13944]: chan_iax2.c:7145 socket_read: Peer 'voicepulse01' is now REACHABLE! Time: 1059 Aug 24 15:11:05 NOTICE[13944]: chan_iax2.c:7813 iax2_poke_noanswer: Peer 'voicepulse01' is now UNREACHABLE! Time: 40 Aug 24 15:11:15 NOTICE[13944]: chan_iax2.c:7145 socket_read: Peer 'voicepulse01' is now REACHABLE! Time: 49 Aug 21 15:33:08 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 39 Aug 21 15:33:18 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 43 Aug 22 13:40:41 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 45 Aug 22 13:40:52 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 1064 Aug 22 16:57:15 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 41 Aug 22 16:57:25 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 43 Aug 23 11:02:36 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 43 Aug 23 11:03:49 NOTICE[13944] chan_iax2.c: Peer 'voicepulse02' is now UNREACHABLE! Time: 37 Aug 23 11:08:19 NOTICE[13944] chan_iax2.c: Peer 'voicepulse02' is now REACHABLE! Time: 39 Aug 23 11:08:26 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 39 Aug 23 11:16:31 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 46 Aug 23 12:10:25 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 1246 Aug 23 14:01:37 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 45 Aug 23 14:04:27 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 258 Aug 23 15:28:36 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 44 Aug 23 15:28:37 NOTICE[13944] chan_iax2.c: Peer 'voicepulse02' is now UNREACHABLE! Time: 39 Aug 23 15:31:27 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 41 Aug 23 15:31:27 NOTICE[13944] chan_iax2.c: Peer 'voicepulse02' is now REACHABLE! Time: 56 Aug 23 15:40:31 NOTICE[13944] chan_iax2.c: Peer 'voicepulse02' is now UNREACHABLE! Time: 40 Aug 23 15:43:31 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 43 Aug 23 15:49:42 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 42 Aug 23 15:57:22 NOTICE[13944] chan_iax2.c: Peer 'voicepulse02' is now REACHABLE! Time: 40 Aug 23 16:37:49 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 44 Aug 23 16:38:00 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 48 Aug 23 17:31:07 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 43 Aug 23 17:32:17 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 42 Aug 23 17:34:21 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 42
Re: [asterisk-users] [SOLVED] 1.2.12.1 crashing
Joseph wrote: Crashing problem SOLVED. It my case it seems to be related usage of NVFaxdetect. Upgrading from Asterisk-1.0... to Asterisk-1.2... one need to recompile the NVFaxdetect module. So if somebody is upgrading and from Asterisk-1.0... and using NVFaxdetect this information might save you few hours. I was under the impression that whenever you upgrade asterisk, any modules that you added needed to be recompiled with the new source. That's the impression I got from the huge warning you get when you compile asterisk and it finds modules in /var/lib/asterisk/modules that the install process didn't put there. ___ --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] [SOLVED] 1.2.12.1 crashing
On Sun, 2006-10-22 at 22:56 +0100, Thomas Kenyon wrote: Joseph wrote: Crashing problem SOLVED. It my case it seems to be related usage of NVFaxdetect. Upgrading from Asterisk-1.0... to Asterisk-1.2... one need to recompile the NVFaxdetect module. So if somebody is upgrading and from Asterisk-1.0... and using NVFaxdetect this information might save you few hours. I was under the impression that whenever you upgrade asterisk, any modules that you added needed to be recompiled with the new source. That's the impression I got from the huge warning you get when you compile asterisk and it finds modules in /var/lib/asterisk/modules that the install process didn't put there. I'm using Getnoo ebuild (currently - asterisk-1.2.13) but when I upgraded from 1.0.11 to 1.2.12.1 it should automatically re-emerge asterisk-app_nv_faxdetect but it didn't. So, after upgrade it was crashing instantly when it hit extension: exten = s,4,NVBackgroundDetect So it is definitely, Gentoo ebuild problem that should be corrected. -- #Joseph ___ --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] wrong outgoing caller id with PRI lines: maybe usecallingpres involved?
Shouldn't it also acquire the callerid of whatever device placed the call? IE: IP phones callerid= in sip.conf or even the phone itself?On 10/21/06, Leo Ann Boon [EMAIL PROTECTED] wrote:Massimiliano Stucchi wrote: On 201006, 10:06, Giorgio Incantalupo wrote: Hi Doug, I do not use extensions.conf so I cannot show anything but I can assure that I do not set the callerid except for parameters inside zapata.conf: usecallerid = yes callerid = asreceived I guess the problem is at the telco's side, since the CLI that is shown seems to be the first one of the numbering scheme.I suppose what you That's the default ISDN behavior. If you don't set the caller id, itwill always send the pilot number. The usecallerid and callerid are forincoming, when used on a trunk. They don't apply to outgoing calls. You need to set the callerid if you want the DDI number to be sent insteadof the pilot number.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] voicemail usernames can't begin with j letter?
vm-isunavail is played because you set the u flag which does exactly that, play the vm-isunavail file.On 10/21/06,
[EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Thanks to all that replayed, I made like Mr Watkins told me, and my problem isapparently solved, although, because of the usage of the syntaxVoiceMail(${EXTEN}|u), now, two more sound files are played: vm-theperson and
vm-isunavail, while before were only played vm-intro and beep.Is there a way to disable this two other files that get played every time?Regards,Ricardo.Quoting Watkins, Bradley
[EMAIL PROTECTED]: I playing a bit with this, it seems that if you use the new syntax it works: exten = _[a-z].,3,VoiceMail(${EXTEN}|u)
You can, of course, also use the b, j, s, and g flags. Even using the VoiceMail(u${EXTEN}) still elides the 'j'. Regards, - Brad -Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
] On Behalf Of Eric ManxPower Wieling Sent: Friday, October 20, 2006 1:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] voicemail usernames can't begin
with j letter? Ricardo Carvalho wrote: I'm running Asterisk version 1.2.10. I also tried with version 1.2.4 and got same problem.
I use SIP and in my extensions.conf I have the following code: exten = _[a-z].,1,Answer exten = _[a-z].,2,Wait(1) exten = _[a-z].,3,VoiceMail(${EXTEN}) exten = _[a-z].,4,Hangup
Through my testing I found that the problem is that when someone enters for example john's voicemail, Asterisk thinks that j letter
is jump flag to n+1 priority. How can I disable, (if possible) this erroneous interpretation that Asterisk does? Have you tried exten = _[a-z].,3,VoiceMail(u${EXTEN})
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[asterisk-users] embedded asterisk
anyone know if any of these could handle asterisk ? http://www.axotec.com/embedded-server.htm ___ --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] wrong outgoing caller id with PRI lines: maybe usecallingpres involved?
Andrew Joakimsen wrote: Shouldn't it also acquire the callerid of whatever device placed the call? IE: IP phones callerid= in sip.conf or even the phone itself? A little clarification: when you send callerid out to the PSTN, it has to match the format of the PSTN. Suppose your phone extension is 1000, naturally its caller ID is 1000. That's fine and well for internal calls. For an external call: when the PBX sends 1000 to the PSTN, the exchange will reject it and send the pilot number instead. You'll need a translation rule to map the your extension to the DDI you want to send to the PSTN. It's similar in Cisco CM, you need to apply a Calling Party Mask to make sure your caller ID will be sent in the correct format. Leo. On 10/21/06, *Leo Ann Boon * [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Massimiliano Stucchi wrote: On 201006, 10:06, Giorgio Incantalupo wrote: Hi Doug, I do not use extensions.conf so I cannot show anything but I can assure that I do not set the callerid except for parameters inside zapata.conf: usecallerid = yes callerid = asreceived I guess the problem is at the telco's side, since the CLI that is shown seems to be the first one of the numbering scheme. I suppose what you That's the default ISDN behavior. If you don't set the caller id, it will always send the pilot number. The usecallerid and callerid are for incoming, when used on a trunk. They don't apply to outgoing calls. You need to set the callerid if you want the DDI number to be sent instead of the pilot number. ___ --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 http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Findme problem
Something was broken with the script on the Wiki... this worked for me, however I have not touched it as I had more interesting projects on the table. I do know when you reject the call it will send the caller to your voicemail (or whatever else you wish to define).
[macro-screen]exten = s,1,Playback(silence/1)exten = s,2,Playback(screen-from)exten = s,3,SayDigits(${ARG1})exten = s,4,Read(ACCEPT|screen-accept|1||1)exten = s,5,GotoIf($[${ACCEPT} = 1 ] ?7:6)
exten = s,6,SetVar(MACRO_RESULT=CONTINUE)exten = s,7,NoOpOn 10/18/06, Eric Jacksch
[EMAIL PROTECTED] wrote:
Greetings all,
I've been working on having Asterisk put a call
through to two different numbers, and give the call to the first one that
acknowledges by pressing the 1 key. I found an example on the wiki, but I
can't get it working.
When I answer the call I hear the message telling
me to press 1 to connect, and as soon as the message is done, the call is
connected. In other words, it is not waiting for me to press a
key.
I'm sure this is a forehead slapper, but I just
can't see it...can anyone help? Here's the relevant portion of the
dialplan, It executes the NoOp(Waiting) and then the macro seems to
immediately exit and the call is connected.
[default]exten =
_XX,1,Dial(SIP/provider/${EXTEN:4},40,M(screen))exten =
_XX,2,Hangup
[macro-screen]exten = s,1,Wait(1)exten
= s,2,Set(TIMEOUT(digit)=5)exten =
s,3,Set(TIMEOUT(response)=10)exten = s,4,Background(press-1)exten
= s,5,NoOp(Waiting)
exten = 1,1,NoOp(Caller accepted)
exten = i,1,NoOp(Invalid response)
exten =
i,2,Set(MACRO_RESULT=CONTINUE)
exten = t,1,NoOp(Timeout)exten =
t,2,Set(MACRO_RESULT=CONTINUE)
[find-eric]exten =
s,1,Playback(pls-wait-connect-call)exten =
s,n,Dial(LOCAL/6135551212LOCAL/6135551313,40,m)
(I have replaced the phone numbers with bogus
ones).
Thanks,
Eric
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[asterisk-users] Re: some transfers dropped.
Could there be something going on in asterisk to make the first request fail, so that the SIP device cancels and retries the transfer(refer)? Could it be manager overuse? -- -- Steven http://www.glimasoutheast.org BerkHolz, Steven [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] We are having an issue with transferred calls being dropped. Looking at the asterisk 1.2.10 logs, it appears that when it is dropped, the SIP unit send a CANCEL message to the server. On successful transfers this is not seen. The errors logged in the SIP Unit error log, I believe are from the second attempt to transfer the call, after it has actually been disconnected. Nothing is deferent in the logs above the CANCEL request for successful or failed transfers. So, I am not sure why the CANCEL is being sent. I can not discern what may be different when it fails. Thank You, Steven BerkHolz Board member of www.glimasoutheast.org ref: from SIP Phone (I think these are the second invite after it is hung up) 2006-OCT-20 17:49:52 GMT +++ Current Timestamp +++ 2006-OCT-20 17:19:47 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-20 15:56:37 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-20 15:50:00 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-20 15:45:38 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-20 15:11:28 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-20 15:10:58 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-20 14:59:26 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-20 12:45:30 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-19 19:53:25 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-19 18:40:52 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-19 18:03:45 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-19 17:55:55 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-19 15:09:13 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-19 15:04:33 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-19 14:52:12 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-19 14:34:35 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-19 14:20:17 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-19 13:45:33 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER ref. from asterisk 1.2.10 logs: Oct 20 13:19:45 VERBOSE[10652] logger.c: -- Requested transfer capability: 0x00 - SPEECH Oct 20 13:19:45 DEBUG[8159] channel.c: Avoiding initial deadlock for 'Zap/25-1' Oct 20 13:19:45 VERBOSE[10652] logger.c: -- Called g2/5155 Oct 20 13:19:45 VERBOSE[10652] logger.c: Transmitting (no NAT) to 172.16.8.200:5065: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.16.8.200:5065;branch=z9hG4bKline0-2425957956;received=172.16.8.200 From: From Desksip:[EMAIL PROTECTED];tag=2425948795 To: sip:[EMAIL PROTECTED];tag=as279eb184 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Oct 20 13:19:45 DEBUG[10658] app_queue.c: Device 'Zap/25' changed to state '2' (In use) but we don't care because they're not a member of any queue. Oct 20 13:19:45 DEBUG[8159] devicestate.c: Changing state for Zap/25 - state 2 (In use) Oct 20 13:19:45 DEBUG[10659] app_queue.c: Device 'Zap/25' changed to state '2' (In use) but we don't care because they're not a member of any queue. Oct 20 13:19:45 DEBUG[8167] chan_zap.c: Enabled echo cancellation on channel 25 Oct 20 13:19:45 VERBOSE[10652] logger.c: -- Zap/25-1 is ringing Oct 20 13:19:45 DEBUG[8159] devicestate.c: Changing state for Zap/25 - state 6 (Ringing) Oct 20 13:19:45 DEBUG[10660] app_queue.c: Device 'Zap/25' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. Oct 20 13:19:45 DEBUG[8171] chan_sip.c: Header 0: (0) Oct 20 13:19:46 VERBOSE[8171] logger.c: -- SIP read from 172.16.8.200:5065: CANCEL sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 172.16.8.200:5065;branch=z9hG4bKline0-2425957956 To: sip:[EMAIL PROTECTED] From: From Desksip:[EMAIL PROTECTED];tag=2425948795 Call-Id: [EMAIL PROTECTED] Max-Forwards: 70 CSeq: 2 CANCEL Content-Length: 0 Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 0: CANCEL sip:[EMAIL PROTECTED] SIP/2.0 (36) Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 1: Via: SIP/2.0/UDP 172.16.8.200:5065;branch=z9hG4bKline0-2425957956 (65) Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 2: To: sip:[EMAIL PROTECTED] (27)
Re: [asterisk-users] compiling libunicall
More machine translator humor :) Somehow, a chica (the bookstore clerk?) and a commando got involved in Diego's attempt to compile libunicall. This should go into the annals of mistranslation. Diego, You should post in point form rather than a single long paragraph. Leo. DiegoF wrote: good, after proving of everything it did not want to work, it installs the sound card to him and not if that I influence in something, but I was convinced that she was not by the versions of the bookstores because compiled them in other equipment and they worked to me without no problem, but good I work them now story that step since I sent the previous mail until I work myself first that everything installs the sound card, compiles and I do not work, later I put myself to review and agreed me that between so many compiled in I put --prefix and sure I had left bookstores by all the sides and when she went to compile the one of libunicall it gave error me, I did grep ' dtmf_put' * that was one of the errors that appeared to me and I saw that in the compilations that had done before to not it appeared me nothing whereas in it completes it if what I did it were to erase all the bookstores that they had watered and leaves you complete them followed without working to me. I am called on myself to create symbolic connections to them to /usr/lib, because they were in /usr/local/lib. later reading I saw that he was not necessary that, single is to create a file in /etc/ld.so.conf.d/ with extension conf and inside I put /usr/local/lib, it followed without working and it is that I lack myself to execute the commando ldconfig. later it erases all the bookstores, it compiles and ready. with respect to which it is not necessary to publish in Spanish, I do for that know ingles and Spanish to it and has been translated by babelfish.altavista.com/tr http://babelfish.altavista.com/tr badly, better understand it in Spanish, in addition because it does not lack the one that little of ingles and reads these messages, like I. / bueno, después de probar de todo no quería funcionar, le instale la tarjeta de sonido y no se si eso influyo en algo, pero yo estaba convencido de que no era por las versiones de las librerías porque las compilaba en otros equipo y me funcionaban sin ningún problema, pero bueno funciono. ahora les cuento que paso desde que envié el correo anterior hasta que me funciono. primero que todo instale la tarjeta de sonido, compile y no funciono, después me puse a revisar y me acorde que entre las tantas compiladas en unas les puse --prefix y claro, me quedaron librerías por todos los lados y cuando iba a compilar la de libunicall me daba error, le hice un grep 'dtmf_put' * que era uno de los errores que me aparecía y vi que en las compilaciones que había hecho antes no me aparecía nada mientras que en la ultima si. lo que hice fue borrar todas las librerías que habían regadas y deje las ultimas. seguía sin funcionarme. me toco crearles enlaces simbólicos al /usr/lib, porque estaban en /usr/local/lib. después leyendo vi que no era necesario eso, solo es crear un archivo en /etc/ld.so.conf.d/ con extensión .conf y dentro puse /usr/local/lib , seguía sin funcionar y es que me falto ejecutar el comando ldconfig. después borre todas las librerías, compile y listo. respecto a que no es necesario publicar en español, lo hago para los que saben ingles y español y haya quedado mal traducido por babelfish.altavista.com/tr http://babelfish.altavista.com/tr, lo entiendan mejor en español, ademas porque no falta el que poco de ingles y lee estos mensajes, como yo. On 10/13/06, *Moises Silva* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: good, it now seems that again it was mistaken, the difference that I see between the equipment in that if I work with which I do not work, is that it does not have sound card. I am going to him to install one and it will comment to them if that were the problem until the next one. You are completly lost, the errors you posted here have NOTHING to do with the kernel or the sound card. As I told you, you are trying to compile non matching API versions of libraries. Regards On 10/12/06, DiegoF [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: hello to all good it seems that the problem is not by the version of spand but by the version of kernel of linux, I proved it in two equipment with fedora4 and works, but I need to compile it in fedora5 and when I do it it marks the error to me that mentions before. Somebody knows where encounter an updated version of these archives? On 10/11/06, Moises Silva [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Diego, this is an english mailing list, there is no need to
[asterisk-users] Re: Asterisk + Huawei
Thanks for your answer, here is some more debug information, if is a codec interrupt issue, how can i fix it?My Sipura uses UID 1234. The huawei softswitch IP address is 10.220.0.2. The Asterisk IP address is 10.223.6.98.The Sipura is registered to the Asterisk box and the Asterisk box is registered to the Huawei softswitch. Thanks a lot for your help,Carlos Andres Medina--- INCOMING -- -- Executing Macro("SIP/10.220.0.2-08191e48", "incoming|SIP/1234") in new stack -- Executing Dial("SIP/10.220.0.2-08191e48", "SIP/1234|30") in new stackWe're at 10.223.6.98 port 19404Adding codec 0x4 (ulaw) to SDPAdding codec 0x8 (alaw) to SDPAdding non-codec 0x1 (telephone-event) to SDP13 headers, 11 linesReliably Transmitting (no NAT) to 10.223.6.99:5150:INVITE
sip:[EMAIL PROTECTED]:5150 SIP/2.0Via: SIP/2.0/UDP 10.223.6.98:5060;branch=z9hG4bK3eb83872;rportFrom: "Anonymous" sip:[EMAIL PROTECTED];tag=as448023d0To: sip:[EMAIL PROTECTED]:5150Contact: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Thu, 19 Oct 2006 01:56:50 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContent-Type: application/sdpContent-Length: 236v=0o=root 1760 1760 IN IP4 10.223.6.98s=sessionc=IN IP4 10.223.6.98t=0 0m=audio 19404 RTP/AVP 0 8 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -- Called 1234-- SIP read from 10.223.6.99:5150:SIP/2.0 100 TryingTo: sip:[EMAIL PROTECTED]:5150From: "Anonymous"
sip:[EMAIL PROTECTED];tag=as448023d0Call-ID: [EMAIL PROTECTED]CSeq: 102 INVITEVia: SIP/2.0/UDP 10.223.6.98:5060;branch=z9hG4bK3eb83872Server: Sipura/SPA2000-2.0.10(e)Content-Length: 0--- (8 headers 0 lines)- SIP read from 10.223.6.99:5150:SIP/2.0 180 RingingTo: sip:[EMAIL PROTECTED]:5150;tag=e2a724add55f408bi0From: "Anonymous" sip:[EMAIL PROTECTED];tag=as448023d0Call-ID: [EMAIL PROTECTED]CSeq: 102 INVITEVia: SIP/2.0/UDP 10.223.6.98:5060;branch=z9hG4bK3eb83872Server: Sipura/SPA2000-2.0.10(e)Content-Length: 0--- (8 headers 0 lines)--- -- SIP/1234-08197388 is ringingTransmitting (no NAT) to 10.220.0.2:5061:SIP/2.0 180 RingingVia: SIP/2.0/UDP
10.220.0.2:5061;branch=z9hG4bK94161ad88;received=10.220.0.2From: Anonymoussip:[EMAIL PROTECTED];tag=961d1a68To: sip:[EMAIL PROTECTED];user=phone;tag=as40afbad8Call-ID: [EMAIL PROTECTED]CSeq: 1 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContact: sip:[EMAIL PROTECTED]Content-Length: 0-- SIP read from 10.223.6.99:5150:SIP/2.0 200 OKTo: sip:[EMAIL PROTECTED]:5150;tag=e2a724add55f408bi0From: "Anonymous" sip:[EMAIL PROTECTED];tag=as448023d0Call-ID: [EMAIL PROTECTED]CSeq: 102 INVITEVia: SIP/2.0/UDP 10.223.6.98:5060;branch=z9hG4bK3eb83872Contact: sip:[EMAIL PROTECTED]:5150Server: Sipura/SPA2000-2.0.10(e)Content-Length: 229Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFERSupported:
x-sipuraContent-Type: application/sdpv=0o=- 78549 78549 IN IP4 10.223.6.99s=-c=IN IP4 10.223.6.99t=0 0m=audio 21101 RTP/AVP 8 100 101a=rtpmap:8 PCMA/8000a=rtpmap:100 NSE/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15a=ptime:30a=sendrecv--- (12 headers 12 lines)---Found RTP audio format 8Found RTP audio format 100Found RTP audio format 101Peer audio RTP is at port 10.223.6.99:21101Found description format PCMAFound description format NSEFound description format telephone-eventCapabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)list_route: hop: sip:[EMAIL PROTECTED]:5150set_destination: Parsing sip:[EMAIL PROTECTED]:5150 for address/port to send toset_destination: set destination to
10.223.6.99, port 5150Transmitting (no NAT) to 10.223.6.99:5150:ACK sip:[EMAIL PROTECTED]:5150 SIP/2.0Via: SIP/2.0/UDP 10.223.6.98:5060;branch=z9hG4bK403f58ec;rportFrom: "Anonymous" sip:[EMAIL PROTECTED];tag=as448023d0To: sip:[EMAIL PROTECTED]:5150;tag=e2a724add55f408bi0Contact: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0--- -- SIP/1234-08197388 answered SIP/10.220.0.2-08191e48We're at 10.223.6.98 port 15322Adding codec 0x4 (ulaw) to SDPAdding codec 0x8 (alaw) to SDPAdding non-codec 0x1 (telephone-event) to SDPReliably Transmitting (no NAT) to 10.220.0.2:5061:SIP/2.0 200 OKVia: SIP/2.0/UDP 10.220.0.2:5061;branch=z9hG4bK94161ad88;received=10.220.0.2From:
Anonymoussip:[EMAIL PROTECTED];tag=961d1a68To: sip:[EMAIL PROTECTED];user=phone;tag=as40afbad8Call-ID: [EMAIL PROTECTED]CSeq: 1 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContact: sip:[EMAIL PROTECTED]Content-Type: application/sdpContent-Length: 233v=0o=root 1760 1760 IN IP4 10.223.6.98s=sessionc=IN IP4 10.223.6.98t=0 0m=audio 15322 RTP/AVP 0 8 97a=rtpmap:0
[asterisk-users] Re: Asterisk + Huawei
need debug * and Huawei, not * and client___ --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] How to deploy a PBX in such a condition ?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 My organization has a LAN now , and there almost a computer in each office for each employee . And in such a situation , what the most economic way to deploy a PBX with asterisk ? Is there good tutorials for me to learn how to do ? -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFPCpm7tZp58UCwyMRAtGvAJ9koQF0Gzg8wxM8K+U01lwBOyenbACfcWu5 jZ68myehj2wrbzYosClWVCg= =Zg/t -END PGP SIGNATURE- ___ --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] wrong outgoing caller id with PRI lines: maybe usecallingpres involved?
On Mon, Oct 23, 2006 at 08:48:21AM +0800, Leo Ann Boon wrote: A little clarification: when you send callerid out to the PSTN, it has to match the format of the PSTN. Suppose your phone extension is 1000, naturally its caller ID is 1000. That's fine and well for internal calls. For an external call: when the PBX sends 1000 to the PSTN, the exchange will reject it and send the pilot number instead. You're an optimist. Lots of switches don't do any verification at all. Or do you know someone who's CNID is *really* 000-123-4567? :-) Last time I got into this (which was about 10 years ago), 5ESS's are really picky, and DMS-100's don't much give a crap (or didn't then) Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --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] How to deploy a PBX in such a condition ?
http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 regards, PaulH On Mon, 2006-10-23 at 10:35 +0800, Bo Yang wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 My organization has a LAN now , and there almost a computer in each office for each employee . And in such a situation , what the most economic way to deploy a PBX with asterisk ? Is there good tutorials for me to learn how to do ? -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFPCpm7tZp58UCwyMRAtGvAJ9koQF0Gzg8wxM8K+U01lwBOyenbACfcWu5 jZ68myehj2wrbzYosClWVCg= =Zg/t -END PGP SIGNATURE- ___ --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
[asterisk-users] Asterisk PBX to a Nortel MICS PBX
What's the best way to connect an Asterisk PBX to a Nortel MICS PBX. I have two offices that I want to link together. TTFN ___ --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] embedded asterisk
The the largest one only has 128m ram. But I think the largest factor is what you intend to doOn 10/22/06, Kristian Kielhofner [EMAIL PROTECTED] wrote:Dovid B wrote: anyone know if any of these could handle asterisk ? http://www.axotec.com/embedded-server.htmDovid,Most of them, running with ARMs running at 70mhz wouldn't be verypractical.The SPIDER-III could work out better, but even still 200 MIPS isn't exactly impressive...There are much better solutions forrunning Asterisk on embedded hardware.--Kristian Kielhofner___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] checking 'voicemail externally - doesn't work
Why are you using Disa to get to the voicemail? What if you dont use it? How are you dialing pstn on your phone anyways?On 10/22/06, Joseph [EMAIL PROTECTED] wrote:Can not check voice mail-box externally.I'm trying to log-in externally (from PSTN line) to check my voice-mail so I created context to authenticate log-in...exten = 7,4,Authenticate(01894546)exten = 7,5,DISA(4789|disa-access)Authentication works OK, I get inside dial none enter my mailbox extension but it doesn't accept my mailbox password even though it isthe correct one.It keep asking me for mailbox, password:Executing VoiceMailMain(SIP/pstn-1270-0819a1f0, pstn1270) in new stack-- Playing 'vm-login' (language 'en')-- Playing 'vm-password' (language 'en')-- Incorrect password '123' for user 'tn12701' (context = default)Could the problem be that the user: pstn1270 is truncated 'tn12701' ? When I try to check the mailbox internally for user who doesn't have amailbox, it works OK:vm_execmain: Specified user '218' not found (check voicemail.conf and/orrealtime config).Falling back to authentication mode. -- Playing 'vm-login' (language 'en')-- Playing 'vm-password' (language 'en')Though, when I log-in externally, it doesn't display that message:Specified user '218' not found...It just asking me for password and telling me it is incorrect one; even though it is the correct one.Using latest Asterisk-1.2.13--#Joseph___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] checking 'voicemail externally - doesn't work
On 23/10/2006, at 10:13 AM, Joseph wrote: I'm trying to log-in externally (from PSTN line) to check my voice-mail so I created context to authenticate log-in Just create an inbound route to VoiceMailMain(). Then, press * during the outbound message and it'll prompt you for a password. Hey presto, you're inside your voicemail! cYa, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9235 5400 Fitzroy, VIC F: +61 (0) 3 9235 5444 3065 W: http://www.squiz.net . Open Source - Own It - Squiz.net .. / ___ --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] asterisk guru needed for job in Chicago area
Hello, I run a small network consulting company in the Chicago area and I have a client who is interested in doing an asterisk based VOIP installation. My company does not have the necessary experience to carry out the project alone so I am looking for an asterisk guru to lead the project. I'm interested in someone from the Chicago or northwest Indiana area who is very experienced with Asterisk deployements in multi-site scenarios connected via VPN tunnels. The person must be very experienced with the following; - Working with various telcos to order and troubleshoot circuits and phone lines - Analog based VOIP gateways - Asterisk PBX on Linux - VOIP in general - SIP and IAX VOIP protocols - Solid experience with IP networks, routers, switches, firewalls The person must also be willing to come on site during deployement to ensure smooth integration but a good portion of the work may possibly be done remotely since we can handle some of it. This is for a one project job initially but if it goes well it could definitely open the door for other VOIP related projects. For anyone who might be interested, please email me your resume. Kind regards, Elvar ___ --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 Realtime... Help Me!!!
Make additional checks : 1) ensure u've unixodbc, unixodbc-devel installed, use this command rpm -qa | grep -i unixodbc MUST see unixodbc and unixodbc-devel in the output!!!, else get unixodbc and unixodbc-devel(am kinda guessing u do have that perfect). 2) /etc/odbcinst.ini and /etc/odbc.ini should be correct. cross check 3) Aaahh.. revelation!! I think, I know where you've gone wrong. In your res_odbc.conf , you have given the database context as mysql(see [mysql]). This should be the same as the 2nd argument in ur extconfig.conf line for realtime for your sipusers. i.e. it should be sipusers = odbc,mysql,sipusers instead of sipusers = odbc,asterisk,sipusers This should work fine. If it doesn't, paste your odbc.ini and odbcinst.ini files as well over here. or give me ssh login access to your machine.(dont wory, wont mess up ur machine). cheerz - Ben. Maurizio Pederneschi wrote: These are my conf file: res_odbc.conf ;;; odbc setup file ; ENV is a global set of environmental variables that will get set. ; Note that all environmental variables can be seen by all connections, ; so you can't have different values for different connections. [ENV] INFORMIXSERVER = my_special_database INFORMIXDIR = /opt/informix ; All other sections are arbitrary names for database connections. ;[asterisk] ;enabled = yes ;dsn = asterisk ;;username = myuser ;;password = mypass ;pre-connect = yes [mysql] enabled = yes dsn = MySQL-asterisk username = root password = pre-connect = yes - extconfig.conf ; ; Static and realtime external configuration ; engine configuration ; ; Please read doc/README.extconfig for basic table ; formatting information. ; [settings] ; ; Static configuration files: ; ; file.conf = driver,database[,table] ; ; maps a particular configuration file to the given ; database driver, database and table (or uses the ; name of the file as the table if not specified) ; ;uncomment to load queues.conf via the odbc engine. ; ;queues.conf = odbc,asterisk,ast_config ; ; The following files CANNOT be loaded from Realtime storage: ; asterisk.conf ; extconfig.conf (this file) ; logger.conf ; ; Additionally, the following files cannot be loaded from ; Realtime storage unless the storage driver is loaded ; early using 'preload' statements in modules.conf: ; manager.conf ; cdr.conf ; rtp.conf ; ; ; Realtime configuration engine ; ; maps a particular family of realtime ; configuration to a given database driver, ; database and table (or uses the name of ; the family if the table is not specified ; ;example = odbc,asterisk,alttable ;iaxusers = odbc,asterisk ;iaxpeers = odbc,asterisk sipusers = odbc,asterisk,sipusers ;sippeers = odbc,asterisk voicemail = odbc,asterisk ;extensions = odbc,asterisk ;queues = odbc,asterisk ;queue_members = odbc,asterisk extensions = odbc,asterisk,extensions This is my table sipusers | id | name | username | context | host| port | secret | allow | ipaddr | type | password | | 1 | pippo| pippo| tutorial | dynamic | | password | g729;ilbc;gsm;ulaw;alaw | NULL | friend | password | | 2 | testAsterisk | testAsterisk | tutorial | dynamic | | password | g729;ilbc;gsm;ulaw;alaw | NULL | friend | password | This is the output of the realtime load command: realtime load sipusers name pippo No rows found matching search criteria. Thank's Maury - Original Message - From: Benjamin Jacob [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, October 20, 2006 12:39 PM Subject: Re: [asterisk-users] Asterisk Realtime... Help Me!!! Maurizio Pederneschi wrote: Hi, i have implemented Asterisk Realtime architecture with Odbc and MySql DB. I have followed all the step of the documentation I found on the Internet. On the CLI, if I make odbc show I see that the DB connection is UP, but if I make realtime load family column value both with extensions family or with sipusers family, I can't find anything in the db. Why it happens? What can I check in my configuration? Someone know if there is a way to test if asterisk make effectively the query to the DB when I make the realtime load command? Please, help me! Maury ___ --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 paste your relevant config files and also an example command (realtime load etc)
[asterisk-users] Re: checking 'voicemail externally - doesn't work
On 2006-10-22 20:58:46 -0700, Avi Miller [EMAIL PROTECTED] said: On 23/10/2006, at 10:13 AM, Joseph wrote: I'm trying to log-in externally (from PSTN line) to check my voice-mail so I created context to authenticate log-in Just create an inbound route to VoiceMailMain(). Then, press * during the outbound message and it'll prompt you for a password. Hey presto, you're inside your voicemail! Well, I was excited to see this response, since I haven't implemented a way to check my voice mail from PSTN, but figure it would work based on your description. Short version: It doesn't work. pressing * during my outgoing message does nothing. Oh well, I guess there really is no free lunch. Marty PS 1.2.12 running on OSX 10.4.8 through Wellgate 3701 PSTN gateway. ___ --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] Re: checking 'voicemail externally - doesn't work
Martin Joseph wrote: On 2006-10-22 20:58:46 -0700, Avi Miller [EMAIL PROTECTED] said: On 23/10/2006, at 10:13 AM, Joseph wrote: I'm trying to log-in externally (from PSTN line) to check my voice-mail so I created context to authenticate log-in Just create an inbound route to VoiceMailMain(). Then, press * during the outbound message and it'll prompt you for a password. Hey presto, you're inside your voicemail! Well, I was excited to see this response, since I haven't implemented a way to check my voice mail from PSTN, but figure it would work based on your description. Short version: It doesn't work. pressing * during my outgoing message does nothing. Oh well, I guess there really is no free lunch. Marty The previous poster is obviously running some Asterisk GUI. You need to read the info on show application voicemailmain in the Asterisk CLI. Add a | at the end of the mailbox name in your extensions.conf and that *should* fix it. ___ --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] Re: checking 'voicemail externally - doesn't work
On 23/10/2006, at 2:24 PM, Martin Joseph wrote: It doesn't work. pressing * during my outgoing message does nothing. Works for me. 1.2.12.1 with FreePBX. When I press *, I get a password prompt. Entering my password gets me into the main voicemail menu. cYa, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9235 5400 Fitzroy, VIC F: +61 (0) 3 9235 5444 3065 W: http://www.squiz.net . Open Source - Own It - Squiz.net .. / ___ --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] Re: checking 'voicemail externally - doesn't work
Avi Miller wrote: On 23/10/2006, at 2:24 PM, Martin Joseph wrote: It doesn't work. pressing * during my outgoing message does nothing. Works for me. 1.2.12.1 with FreePBX. When I press *, I get a password prompt. Entering my password gets me into the main voicemail menu. FreePBX is NOT Asterisk. ___ --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 guru needed for job in Chicago area
Call CompuNetWorld. +1 (704) 644-5528 -Original Message- From: Elvar[EMAIL PROTECTED] Sent: 10/23/06 12:03:07 AM To: asterisk-users@lists.digium.comasterisk-users@lists.digium.com Subject: [asterisk-users] asterisk guru needed for job in Chicago area Hello, I run a small network consulting company in the Chicago area and I have a client who is interested in doing an asterisk based VOIP installation. My company does not have the necessary experience to carry out the project alone so I am looking for an asterisk guru to lead the project. I'm interested in someone from the Chicago or northwest Indiana area who is very experienced with Asterisk deployements in multi-site scenarios connected via VPN tunnels. The person must be very experienced with the following; - Working with various telcos to order and troubleshoot circuits and phone lines - Analog based VOIP gateways - Asterisk PBX on Linux - VOIP in general - SIP and IAX VOIP protocols - Solid experience with IP networks, routers, switches, firewalls The person must also be willing to come on site during deployement to ensure smooth integration but a good portion of the work may possibly be done remotely since we can handle some of it. This is for a one project job initially but if it goes well it could definitely open the door for other VOIP related projects. For anyone who might be interested, please email me your resume. Kind regards, Elvar ___ --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] Re: checking 'voicemail externally - doesn't work
On 23/10/2006, at 2:26 PM, Eric ManxPower Wieling wrote: The previous poster is obviously running some Asterisk GUI. Yes, sorry. I am running FreePBX, but I didn't notice the | in the call to VoiceMailMain, otherwise I would've mentioned it. :( My bad. -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9235 5400 Fitzroy, VIC F: +61 (0) 3 9235 5444 3065 W: http://www.squiz.net . Open Source - Own It - Squiz.net .. / ___ --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] Audiocodes MP-20x
On 10/23/06, Andrew Joakimsen [EMAIL PROTECTED] wrote: Has anyone used the AudioCodes MP-20x? http://audiocodes.com/Objects/Analog_Telephone_Adapter_Series_MP_20X.pdf Seems like a good device, but I can't seem to find anyone actually using them... I am using an AudioCodes Mediant1000 and now trying to configure MP-118. The mediant1000 works well, and I will update the wiki some time soon with the exact configurations to get it working. raj ___ --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] Re: checking 'voicemail externally - doesn't work
On 23/10/2006, at 2:35 PM, Eric ManxPower Wieling wrote: Works for me. 1.2.12.1 with FreePBX. When I press *, I get a password prompt. Entering my password gets me into the main voicemail menu. FreePBX is NOT Asterisk. Yes, I know that. Hence the 1.2.12.1 *with* FreePBX statement. I.E. Asterisk v1.2.12.1 *with* FreePBX *added* I know what FreePBX is. I also know the differences between Asterisk, FreePBX, [EMAIL PROTECTED] and TrixBox. :) cYa, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9235 5400 Fitzroy, VIC F: +61 (0) 3 9235 5444 3065 W: http://www.squiz.net . Open Source - Own It - Squiz.net .. / ___ --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] Re: checking 'voicemail externally - doesn't work
Avi Miller wrote: On 23/10/2006, at 2:35 PM, Eric ManxPower Wieling wrote: Works for me. 1.2.12.1 with FreePBX. When I press *, I get a password prompt. Entering my password gets me into the main voicemail menu. FreePBX is NOT Asterisk. Yes, I know that. Hence the 1.2.12.1 *with* FreePBX statement. I.E. Asterisk v1.2.12.1 *with* FreePBX *added* I know what FreePBX is. I also know the differences between Asterisk, FreePBX, [EMAIL PROTECTED] and TrixBox. :) Pray, tel me difference!! ___ --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] Re: checking 'voicemail externally - doesn't work
On Sun, 2006-10-22 at 23:26 -0500, Eric ManxPower Wieling wrote: Martin Joseph wrote: On 2006-10-22 20:58:46 -0700, Avi Miller [EMAIL PROTECTED] said: On 23/10/2006, at 10:13 AM, Joseph wrote: I'm trying to log-in externally (from PSTN line) to check my voice-mail so I created context to authenticate log-in Just create an inbound route to VoiceMailMain(). Then, press * during the outbound message and it'll prompt you for a password. Hey presto, you're inside your voicemail! Well, I was excited to see this response, since I haven't implemented a way to check my voice mail from PSTN, but figure it would work based on your description. Short version: It doesn't work. pressing * during my outgoing message does nothing. Oh well, I guess there really is no free lunch. Marty The previous poster is obviously running some Asterisk GUI. You need to read the info on show application voicemailmain in the Asterisk CLI. Add a | at the end of the mailbox name in your extensions.conf and that *should* fix it. What does the | do like this: exten = s,8,Voicemail(11|) From the CLI show application voicemailmain Description] VoiceMailMain([EMAIL PROTECTED]|options]): This application allows the calling party to check voicemail messages. A specific mailbox, and optional corresponding context, may be specified. If a mailbox is not provided, the calling party will be prompted to enter one. If a context is not specified, the 'default' context will be used. Is Voicemail the same as VoiceMailMain -- #Joseph ___ --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] Re: checking 'voicemail externally - doesn't work
Joseph wrote: What does the | do like this: exten = s,8,Voicemail(11|) From the CLI show application voicemailmain Description] VoiceMailMain([EMAIL PROTECTED]|options]): This application allows the calling party to check voicemail messages. A specific mailbox, and optional corresponding context, may be specified. If a mailbox is not provided, the calling party will be prompted to enter one. If a context is not specified, the 'default' context will be used. Is Voicemail the same as VoiceMailMain Voicemail allows the caller to leave voicemail. Voicemailmain allows you to check your voicemail. 1.0.x Asterisk mailbox options were put as a prefix to the mailbox, such as Voicemail(u11) would play the unavailable message to the caller. 1.2 (I think) changed this to make it more like all the other applications, i.e., use a , or | before the options. ___ --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 Realtime... Help Me!!!
Thanks alot! Indeed it was the 3th solution, I changed sipusers = odbc,MySQL-asterisk,sip_buddies sippeers = odbc,MySQL-asterisk,sip_buddies into sipusers = odbc,mysql2,sip_buddies sippeers = odbc,mysql2,sip_buddies And realtime load sipusers username 1006 now returns data :-) greets Tijl Van den Broeck On 10/23/06, Benjamin Jacob [EMAIL PROTECTED] wrote: Make additional checks : 1) ensure u've unixodbc, unixodbc-devel installed, use this command rpm -qa | grep -i unixodbc MUST see unixodbc and unixodbc-devel in the output!!!, else get unixodbc and unixodbc-devel(am kinda guessing u do have that perfect). 2) /etc/odbcinst.ini and /etc/odbc.ini should be correct. cross check 3) Aaahh.. revelation!! I think, I know where you've gone wrong. In your res_odbc.conf , you have given the database context as mysql(see [mysql]). This should be the same as the 2nd argument in ur extconfig.conf line for realtime for your sipusers. i.e. it should be sipusers = odbc,mysql,sipusers instead of sipusers = odbc,asterisk,sipusers This should work fine. If it doesn't, paste your odbc.ini and odbcinst.ini files as well over here. or give me ssh login access to your machine.(dont wory, wont mess up ur machine). cheerz - Ben. Maurizio Pederneschi wrote: These are my conf file: res_odbc.conf ;;; odbc setup file ; ENV is a global set of environmental variables that will get set. ; Note that all environmental variables can be seen by all connections, ; so you can't have different values for different connections. [ENV] INFORMIXSERVER = my_special_database INFORMIXDIR = /opt/informix ; All other sections are arbitrary names for database connections. ;[asterisk] ;enabled = yes ;dsn = asterisk ;;username = myuser ;;password = mypass ;pre-connect = yes [mysql] enabled = yes dsn = MySQL-asterisk username = root password = pre-connect = yes - extconfig.conf ; ; Static and realtime external configuration ; engine configuration ; ; Please read doc/README.extconfig for basic table ; formatting information. ; [settings] ; ; Static configuration files: ; ; file.conf = driver,database[,table] ; ; maps a particular configuration file to the given ; database driver, database and table (or uses the ; name of the file as the table if not specified) ; ;uncomment to load queues.conf via the odbc engine. ; ;queues.conf = odbc,asterisk,ast_config ; ; The following files CANNOT be loaded from Realtime storage: ; asterisk.conf ; extconfig.conf (this file) ; logger.conf ; ; Additionally, the following files cannot be loaded from ; Realtime storage unless the storage driver is loaded ; early using 'preload' statements in modules.conf: ; manager.conf ; cdr.conf ; rtp.conf ; ; ; Realtime configuration engine ; ; maps a particular family of realtime ; configuration to a given database driver, ; database and table (or uses the name of ; the family if the table is not specified ; ;example = odbc,asterisk,alttable ;iaxusers = odbc,asterisk ;iaxpeers = odbc,asterisk sipusers = odbc,asterisk,sipusers ;sippeers = odbc,asterisk voicemail = odbc,asterisk ;extensions = odbc,asterisk ;queues = odbc,asterisk ;queue_members = odbc,asterisk extensions = odbc,asterisk,extensions This is my table sipusers | id | name | username | context | host| port | secret | allow | ipaddr | type | password | | 1 | pippo| pippo| tutorial | dynamic | | password | g729;ilbc;gsm;ulaw;alaw | NULL | friend | password | | 2 | testAsterisk | testAsterisk | tutorial | dynamic | | password | g729;ilbc;gsm;ulaw;alaw | NULL | friend | password | This is the output of the realtime load command: realtime load sipusers name pippo No rows found matching search criteria. Thank's Maury - Original Message - From: Benjamin Jacob [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, October 20, 2006 12:39 PM Subject: Re: [asterisk-users] Asterisk Realtime... Help Me!!! Maurizio Pederneschi wrote: Hi, i have implemented Asterisk Realtime architecture with Odbc and MySql DB. I have followed all the step of the documentation I found on the Internet. On the CLI, if I make odbc show I see that the DB connection is UP, but if I make realtime load family column value both with extensions family or with sipusers family, I can't find anything in the db. Why it happens? What can I check in my configuration? Someone know if there is a way to test if asterisk make effectively the query to the DB when I make the realtime load command? Please, help me! Maury
Re: [asterisk-users] checking 'voicemail externally - doesn't work
On Sun, 2006-10-22 at 23:46 -0400, Andrew Joakimsen wrote:
Why are you using Disa to get to the voicemail? What if you dont use
it? How are you dialing pstn on your phone anyways?
Second authentication DISA is for additional security and it doesn't
cause any problem, the authentication is giving me access to voicemail
but password is not recognized.
[voicemail]
exten = 1000,1,NoCDR()
exten = 1000,2,Answer()
exten = 1000,3,VoicemailMain(${CALLERIDNUM})
[disa-access]
include = tollfree
include = voicemail
Anyhow, adding pipe | at the end of
exten = 1000,3,VoicemailMain(${CALLERIDNUM}|)
doesn't work?
--
#Joseph
On 10/22/06, Joseph [EMAIL PROTECTED] wrote:
Can not check voice mail-box externally.
I'm trying to log-in externally (from PSTN line) to check my
voice-mail so I created context to authenticate log-in
...
exten = 7,4,Authenticate(01894546)
exten = 7,5,DISA(4789|disa-access)
[snip]
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