[Asterisk-Users] Need help with perl script/agi for ringback
Hi,
I'm trying to write a simple perl script that will run
the following:
Action: Originate
Channel: local/[EMAIL PROTECTED]/r/n
Exten: 1234
Context: callback
Priority: 1
Extensions.conf
exten => 500,1,agi,callback.pl
callback perl script:
use Net::Telnet ();
$mgrUSERNAME='fred';
$mgrSECRET='bloggs';
$server_ip='127.0.0.1';
$tn->print("Action: originate\nExten: 1234\nContext:
user\nChannel: local/[EMAIL PROTECTED]/r/n\nPriority:
1\nCallerid: 1234\n\n");
$tn->waitfor('/Event: Newchannel.*/') or die "Unable
to determine call status", $tn->lastline;
# wait for asterisk to process
$tn->print("Action: Logoff\n\n");
I'm not a programmer (as u can probably tell) so any
pointers would be much appreciated.
Cheers,
Taff.
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[Asterisk-Users] inter asterisk
Hi,I am trying to forward calls to another * server with IAXHere is What I want to Do1- Call SERVER1, let say at 51412345678 2- SERVER1 should transfer the call to SERVER2 in a remote location3- SERVER2 Receive the call and transfer it to the PSTN number. I have one X100P card on each machine. What is happening is that when the remote party picks up the phone, all he can hear is a weird sound. CONFIGS: SERVER1: zaptel.conf - ~ [channels] ~ language=fr ~ context=montréal ~ signalling=fxs_ks ~ usercallerid=yes ~ callwaiting=yes ~ threewaycalling=yes ~ transfer=yes ~ cancellforward=yes ~ echocancel=yes ~ echocancelwhenbridged=yes ~ echotraining=yes ~ relaxdtmf=yes ~ busydetect=yes ~ busycount=4 ~ callprogress=yes ~ group=1 ~ channel=>1 -- (same for SERVER2) IAX.conf ~ [general] ~ bindport=4569 ~ delayreject=yes ~ language=fr ~ allow=all ~ jutterbuffer=no ~ register => username:[EMAIL PROTECTED] ~ tos=lowdelay ~ autokill=yes ~ ~ [quebec] ~ type=friends ~ username = username ~ password=password ~ context=montréal ~ host=Dynamic ~ secret = password ~ disallow = all ~ allow=ulaw ~ allow=gsm extensions.conf --(Same for SERVER2 but no registration) ~ [general] ~ static=yes ~ writeprotect=yes ~ autofallthrough=yes ~ [montréal] ~ exten=>s,1,Answer ~ exten=>s,2,Playback(message-transfer) ~ exten=>s,3,Dial(IAX2/username:[EMAIL PROTECTED]/[EMAIL PROTECTED]) ; always the same number ~ exten=>s,4,Hangup My remote server receive the call, answer the line and then Dial(ZAP/1/51412345678). So far so good. But when 51412345678 pickup the phone,all she can hear is a weird sound.What am I doing wrong ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need help with perl script/agi for ringback
kill the line breaks?
On Sun, 6 Feb 2005, taf taffey wrote:
Hi,
I'm trying to write a simple perl script that will run
the following:
Action: Originate
Channel: local/[EMAIL PROTECTED]/r/n
Exten: 1234
Context: callback
Priority: 1
Extensions.conf
exten => 500,1,agi,callback.pl
callback perl script:
use Net::Telnet ();
$mgrUSERNAME='fred';
$mgrSECRET='bloggs';
$server_ip='127.0.0.1';
$tn->print("Action: originate\nExten: 1234\nContext:
user\nChannel: local/[EMAIL PROTECTED]/r/n\nPriority:
1\nCallerid: 1234\n\n");
$tn->waitfor('/Event: Newchannel.*/') or die "Unable
to determine call status", $tn->lastline;
# wait for asterisk to process
$tn->print("Action: Logoff\n\n");
I'm not a programmer (as u can probably tell) so any
pointers would be much appreciated.
Cheers,
Taff.
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[Asterisk-Users] IAXy ring frequency
Hello all, In changing the phone connected to my IAXy I remembered that this phone will not ring on the standard frquency (20hz?) when connected to the FXS. There is a patch, one line to change in wcfxs.c for use with a TDM400P, but obviously this doesn't change the FXS on the IAXy. Is there any possibiity of a "patch" for the provisioning data to raise the frequency to 25hz on the IAXy FXS? Maybe this is not part of provisioning data, though in which case there is no answer. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Xorcom Rapid 1.0 released
Hi folks Xorcom Rapid 1.0 is avilable for download Q: 1.0? A: Sure, better than 0.9.1: * Asterisk 1.0.5 * Base packages upgraded * Built with SpanDSP support * Improved Zaptel detection * ast-cmd with some useful command-line abilities provided * ssh installed by default * putty.exe is included on the CD * music-on-hold files removed due to potential licensing issues (http://bugs.debian.org/288429) * Country selection in the installer has been fixed * Kernel upgraded (may require manual upgrade) * Additional options available in the menu http://xorcom.com/release-notes-v1_0.html Q: Rapid? A: Debian/GNU/Linux/Asterisk: a distribution that features an auto-install for Debian Linux and pre-configured Asterisk. It quickly and effortlessly converts any PC to a functioning Asterisk PBX. http://www.xorcom.com/rapid.html Q: Xorcom? A: Xorcom Ltd., a contributing member of the Asterisk community, develops software and hardware for the open source Asterisk PBX environment. Our goal is to make Asterisk a friendly, easyâtoâinstall and easyâtoâuse system, enabling fast and simple installation and configuration of PBXs of all sizes. http://www.xorcom.com/ -- Tzafrir Cohen icq#16849755 +972-50-7952406 [EMAIL PROTECTED]http://xorcom.com - Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] not sharing IRQ's
but when i remove uhci_hcd module i will fall in a big trouble, look: the problem will solve when i load uhci_hcd again!! i've a TE405P card installed and modules loaded. Feb 6 08:11:16 WARNING[2907]: Failed to create new channel thread Feb 6 08:11:16 WARNING[2907]: Failed to start PBX :( Feb 6 08:13:57 WARNING[2907]: Failed to create update thread! Feb 6 08:14:12 WARNING[2907]: Failed to create new channel thread Feb 6 08:14:12 WARNING[2907]: Failed to start PBX :( Feb 6 08:14:12 WARNING[2907]: Failed to create update thread! Feb 6 08:14:18 WARNING[2907]: Maximum retries exceeded on call Feb 6 08:16:26 WARNING[2907]: Failed to create update thread! Feb 6 08:16:26 WARNING[2907]: Failed to create new channel thread Feb 6 08:16:26 WARNING[2907]: Failed to start PBX :( Feb 6 08:16:26 WARNING[2907]: Failed to create update thread! Feb 6 08:16:27 WARNING[2907]: Failed to create update thread! Feb 6 08:16:27 WARNING[2907]: Failed to create new channel thread Feb 6 08:16:27 WARNING[2907]: Failed to start PBX :( Feb 6 08:16:27 WARNING[2907]: Failed to create update thread! Feb 6 08:16:27 WARNING[2907]: Failed to create update thread! Feb 6 08:16:27 WARNING[2907]: Failed to create new channel thread Feb 6 08:16:27 WARNING[2907]: Failed to start PBX :( Feb 6 08:16:27 WARNING[2907]: Failed to create update thread! Feb 6 08:16:27 WARNING[2907]: Failed to create update thread! Feb 6 08:17:35 WARNING[2907]: Failed to create update thread! Feb 6 08:17:35 WARNING[2907]: Failed to create update thread! Feb 6 08:17:35 WARNING[2907]: Failed to create update thread! Feb 6 08:17:35 WARNING[2907]: Failed to create update thread! Feb 6 08:17:35 WARNING[2907]: Failed to create update thread! Feb 6 08:17:35 WARNING[2907]: Failed to create update thread! Feb 6 08:18:13 WARNING[2907]: Failed to create new channel thread Feb 6 08:18:13 WARNING[2907]: Failed to start PBX :( Feb 6 08:18:13 WARNING[2907]: Failed to create update thread! Feb 6 08:18:13 WARNING[2907]: Failed to create new channel thread Feb 6 08:18:13 WARNING[2907]: Failed to start PBX :( Feb 6 08:18:15 WARNING[2907]: Failed to create update thread! Feb 6 08:18:15 WARNING[2907]: Failed to create update thread! Feb 6 08:18:16 WARNING[2907]: Failed to create new channel thread Feb 6 08:18:16 WARNING[2907]: Failed to start PBX :( Feb 6 08:18:17 WARNING[2907]: Failed to create update thread! Feb 6 08:18:17 WARNING[2907]: Failed to create update thread! Feb 6 08:18:17 WARNING[2907]: Unable to start simple switch thread on channel 34 Feb 6 08:18:17 WARNING[2907]: Cannot kill myself Feb 6 08:18:17 WARNING[2907]: Failed to create update thread! Feb 6 08:18:20 WARNING[2907]: Failed to create update thread! Feb 6 08:18:20 WARNING[2907]: Failed to create new channel thread Feb 6 08:18:20 WARNING[2907]: Failed to start PBX :( Feb 6 08:18:21 WARNING[2907]: Cannot kill myself Feb 6 08:18:26 WARNING[2907]: Failed to create update thread! Feb 6 08:18:26 WARNING[2907]: Unable to start simple switch thread on channel 34 Feb 6 08:18:26 WARNING[2907]: Cannot kill myself Feb 6 08:18:26 WARNING[2907]: Failed to create update thread! Feb 6 08:18:26 WARNING[2907]: Failed to create new channel thread Feb 6 08:18:26 WARNING[2907]: Failed to start PBX :( Feb 6 08:18:26 WARNING[2907]: Failed to create update thread! On Wed, 12 Jan 2005 20:08:15 -0500, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > > just to make sure: > > when i have zaptel devices on my box and i also use meetme and iax2, > > do i need to have USB device enabled and it's modules loaded? > No > your zaptel device will provide the needed hardware timer > > the USB timer hack is for when you don't have any digium card > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FIX YOUR AUTO-RESPONDERS!!!
> > They can always check the archives to read up on missed posts, and it > > would save us all the trouble in the mean time ;-) Isn't it obvious that with a choice of hundreds of free email providers, anyone who wants to avois this problem need only use a throwaway account like gmail with mailing lists and avoid checking the "away message" stuff. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] inter asterisk
> I am trying to forward calls to another * server with IAX > Here is What I want to Do > 1- Call SERVER1, let say at 51412345678 > 2- SERVER1 should transfer the call to SERVER2 in a remote location > 3- SERVER2 Receive the call and transfer it to the PSTN number. > > I have one X100P card on each machine. What is happening is that when the > remote party picks up the phone, all he can hear > is a weird sound. > > CONFIGS: > > SERVER1: > zaptel.conf >- >~ [channels] >~ language=fr >~ context=montréal >~ signalling=fxs_ks >~ usercallerid=yes >~ callwaiting=yes >~ threewaycalling=yes >~ transfer=yes >~ cancellforward=yes >~ echocancel=yes >~ echocancelwhenbridged=yes >~ echotraining=yes >~ relaxdtmf=yes >~ busydetect=yes >~ busycount=4 >~ callprogress=yes >~ group=1 >~ channel=>1 >-- (same for SERVER2) > > IAX.conf > >~ [general] >~ bindport=4569 >~ delayreject=yes >~ language=fr >~ allow=all >~ jutterbuffer=no >~ register => username:[EMAIL PROTECTED] >~ tos=lowdelay >~ autokill=yes >~ >~ [quebec] >~ type=friends >~ username = username >~ password=password >~ context=montréal >~ host=Dynamic >~ secret = password >~ disallow = all >~ allow=ulaw >~ allow=gsm > > extensions.conf >--(Same for SERVER2 but no > registration) >~ [general] >~ static=yes >~ writeprotect=yes >~ autofallthrough=yes >~ [montréal] >~ exten=>s,1,Answer >~ exten=>s,2,Playback(message-transfer) >~ exten=>s,3,Dial(IAX2/username:[EMAIL PROTECTED]/[EMAIL PROTECTED]) ; > always the same number >~ exten=>s,4,Hangup > > > > My remote server receive the call, answer the line and then > Dial(ZAP/1/51412345678). So far so good. But when 51412345678 pickup the phone, > all she can hear is a weird sound. > What am I doing wrong ? Difficult to tell without some feedback from the CLI. If you actually copy/pasted the above config statements, I'm assuming you manually added all those "~" at the front of each line. If they are actually in your config, get rid of them. The statement "jutterbuffer=no" should be jitterbuffer=no. One thing you might try to at least eliminate possible problems is to change iax.conf to disallow=all and allow=gsm only. Get rid of the allow=ulaw and do another test. Might as well add trunk=no to this link as well. (Must stop and restart * after making these type changes.) You might try 'iax2 debug' from the CLI on both machines and look at the detail to see if you can spot any conflicts or problems. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help with extensions
Hello:
I'd like some help with defining extension rules. I want calls arriving
at Asterisk from my SIP proxy to be sent directly to voicemail. I'd
also like the appropriate greeting played when the call gets to
voicemail.
My proxy prefixes the extension with a "u" or "b" based on
SIP response codes before relaying to Asterisk. So when the
call arrives it is in the format [u|b][3|6|7|8]
If I hard code the following rules then calls get forwarded
as expected.
exten => u67501,1,VoiceMail2(${EXTEN})
exten => #,2,Hangup
However to save on typing I'd like a general rule. I've tried
the following but Asterisk cannot find the extension with this
set of rules. Can someone explain how what I want can be
accomplished?
exten => _[ub][3678].,2,VoiceMail2(${EXTEN})
exten => #,2,Hangup
Thanks
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RE: [Asterisk-Users] inter asterisk
One thing I do on remote sites is set up a soft phone so I can call myself, this proves out the link and quality before anything else. DIAX id good for this as you can connect to multiple sites, also good to see if you have problems before anyone else calls you to say there is a problem. It also helps in cases like this, if your return quality is good then the possible fault lies with the ZAP interface. Process of elimination, works for me every time. Regards Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Ousmane Doukara Sent: 06 February 2005 08:46 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] inter asterisk Hi, I am trying to forward calls to another * server with IAX Here is What I want to Do 1- Call SERVER1, let say at 51412345678 2- SERVER1 should transfer the call to SERVER2 in a remote location 3- SERVER2 Receive the call and transfer it to the PSTN number. I have one X100P card on each machine. What is happening is that when the remote party picks up the phone, all he can hear is a weird sound. CONFIGS: SERVER1: zaptel.conf - ~ [channels] ~ language=fr ~ context=montréal ~ signalling=fxs_ks ~ usercallerid=yes ~ callwaiting=yes ~ threewaycalling=yes ~ transfer=yes ~ cancellforward=yes ~ echocancel=yes ~ echocancelwhenbridged=yes ~ echotraining=yes ~ relaxdtmf=yes ~ busydetect=yes ~ busycount=4 ~ callprogress=yes ~ group=1 ~ channel=>1 -- (same for SERVER2) IAX.conf ~ [general] ~ bindport=4569 ~ delayreject=yes ~ language=fr ~ allow=all ~ jutterbuffer=no ~ register => username:[EMAIL PROTECTED] ~ tos=lowdelay ~ autokill=yes ~ ~ [quebec] ~ type=friends ~ username = username ~ password=password ~ context=montréal ~ host=Dynamic ~ secret = password ~ disallow = all ~ allow=ulaw ~ allow=gsm extensions.conf --(Same for SERVER2 but no registration) ~ [general] ~ static=yes ~ writeprotect=yes ~ autofallthrough=yes ~ [montréal] ~ exten=>s,1,Answer ~ exten=>s,2,Playback(message-transfer) ~ exten=>s,3,Dial(IAX2/username:[EMAIL PROTECTED]/[EMAIL PROTECTED] al) ; always the same number ~ exten=>s,4,Hangup My remote server receive the call, answer the line and then Dial(ZAP/1/51412345678). So far so good. But when 51412345678 pickup the phone, all she can hear is a weird sound. What am I doing wrong ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help with extensions
Steve,
I haven't tried this but can't you do something like.
[from-proxy]
exten => s,1,Answer
exten => s,2,VoiceMail2(${EXTEN:1})
exten => 3,3,Hangup
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve Blair
Sent: 06 February 2005 12:14
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Help with extensions
Hello:
I'd like some help with defining extension rules. I want calls arriving
at Asterisk from my SIP proxy to be sent directly to voicemail. I'd
also like the appropriate greeting played when the call gets to
voicemail.
My proxy prefixes the extension with a "u" or "b" based on
SIP response codes before relaying to Asterisk. So when the
call arrives it is in the format [u|b][3|6|7|8]
If I hard code the following rules then calls get forwarded
as expected.
exten => u67501,1,VoiceMail2(${EXTEN})
exten => #,2,Hangup
However to save on typing I'd like a general rule. I've tried
the following but Asterisk cannot find the extension with this
set of rules. Can someone explain how what I want can be
accomplished?
exten => _[ub][3678].,2,VoiceMail2(${EXTEN})
exten => #,2,Hangup
Thanks
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Re: [Asterisk-Users] Problems DIALING to IAXTEL
On Feb 6, 2005, at 8:18, Gonzalo Gasca wrote: Do the ECHO TEST dialing to 17002353660 Make 4 calls first one completed succesfully, second FAILED, third and fourth were succesful. For the second call i got this ERROR: -- Hungup 'IAX2[69.73.19.178:4569]/1' == No one is available to answer at this time This happens randomly when I dial to IAX (ie Digium numbers) not only this time It's busy. That's all. iaxtel is a *free* service, there cannot be any guarantees that all calls are successful. If you can't live with that you need to spend money and buy from a commercial provider. jens ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help with extensions
Steve,
Sorry bum information. Line 2 should read: -
exten => s,2,VoiceMail2(${EXTEN})
Don't need to strip the first digit as this is either u or b already,
(Unobtainable or Busy).
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of David J
Carter
Sent: 06 February 2005 12:30
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Help with extensions
Steve,
I haven't tried this but can't you do something like.
[from-proxy]
exten => s,1,Answer
exten => s,2,VoiceMail2(${EXTEN:1})
exten => 3,3,Hangup
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve Blair
Sent: 06 February 2005 12:14
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Help with extensions
Hello:
I'd like some help with defining extension rules. I want calls arriving
at Asterisk from my SIP proxy to be sent directly to voicemail. I'd
also like the appropriate greeting played when the call gets to
voicemail.
My proxy prefixes the extension with a "u" or "b" based on
SIP response codes before relaying to Asterisk. So when the
call arrives it is in the format [u|b][3|6|7|8]
If I hard code the following rules then calls get forwarded
as expected.
exten => u67501,1,VoiceMail2(${EXTEN})
exten => #,2,Hangup
However to save on typing I'd like a general rule. I've tried
the following but Asterisk cannot find the extension with this
set of rules. Can someone explain how what I want can be
accomplished?
exten => _[ub][3678].,2,VoiceMail2(${EXTEN})
exten => #,2,Hangup
Thanks
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[Asterisk-Users] Question about X100P card
Hello my brothers and sisters, Is "X100P" card suitable to VoIP? and if "yes", am i need to only "X100P" and "Asterisk" Package? or i need also to other cards or packages? and if "X100P" card not suitable to VoIP, please recommend a another card, (please take in your account that i would like to connect standard analog line to the card directly). Thanks in advance Eng. Yousri Farouk www.egypt.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Proxied SIP
I want to install Asterisk for an organization that wants it to do some call routing for them. They have a SIP provider that is going to provide one termination and one origination account. We are going to have to route a rather large number of calls (50-100,000 concurrent), but can't find any information on how to proxy calls adaquately. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Question about X100P card
> Is "X100P" card suitable to VoIP? and if "yes", am i need to only "X100P" > and "Asterisk" Package? or i need also to other cards or packages? > and if "X100P" card not suitable to VoIP, please recommend a another card, > (please take in your account that i would like to connect standard analog > line to the card directly). The x100p card was designed to work with the US telephone standards (600 ohm line impedence) and will likely cause echo if used in other countries. I don't have a clue whether Egypt telephone standards are the same as the US; probably not. The TDM04B card (four fxo ports) was designed to work with telephone standards in most countries. You should consider using that card instead and you will find it at www.digium.com web site. Each TDM card can have up to four daughter modules on it, and each module can be either a fxs module (for connecting to standard telephones) or fxo (for connection to telephone lines that come from your telephone company). To order a card with only one line, the part number would be TDM01B. A Linux PC with a TDM card installed and asterisk code will work just fine for small installations. Nothing else required. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Proxied SIP
Chris Tooley wrote: I want to install Asterisk for an organization that wants it to do some call routing for them. They have a SIP provider that is going to provide one termination and one origination account. We are going to have to route a rather large number of calls (50-100,000 concurrent), but can't find any information on how to proxy calls adaquately. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users look to SER but 100,000 calls requires a tremendious amount of bandwidth, make sure you have it! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Question about X100P card
Yousri, You may want to look at these two articles which will give you a good idea of what hardware you need for a simple system with one or two phones and phone lines: http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with extensions
You read my mind. I was about to compose a response asking
why you'd strip digits when this message arrived.
Thanks.
David J Carter wrote:
Steve,
Sorry bum information. Line 2 should read: -
exten => s,2,VoiceMail2(${EXTEN})
Don't need to strip the first digit as this is either u or b already,
(Unobtainable or Busy).
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of David J
Carter
Sent: 06 February 2005 12:30
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Help with extensions
Steve,
I haven't tried this but can't you do something like.
[from-proxy]
exten => s,1,Answer
exten => s,2,VoiceMail2(${EXTEN:1})
exten => 3,3,Hangup
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve Blair
Sent: 06 February 2005 12:14
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Help with extensions
Hello:
I'd like some help with defining extension rules. I want calls arriving
at Asterisk from my SIP proxy to be sent directly to voicemail. I'd
also like the appropriate greeting played when the call gets to
voicemail.
My proxy prefixes the extension with a "u" or "b" based on
SIP response codes before relaying to Asterisk. So when the
call arrives it is in the format [u|b][3|6|7|8]
If I hard code the following rules then calls get forwarded
as expected.
exten => u67501,1,VoiceMail2(${EXTEN})
exten => #,2,Hangup
However to save on typing I'd like a general rule. I've tried
the following but Asterisk cannot find the extension with this
set of rules. Can someone explain how what I want can be
accomplished?
exten => _[ub][3678].,2,VoiceMail2(${EXTEN})
exten => #,2,Hangup
Thanks
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Re: [Asterisk-Users] Help with extensions
Nope. Didn't work. When the call hits extensions.conf no matching
extension definition exists. I tried replacing line #2,
exten => s,2,VoiceMail2(${EXTEN}) with exten =>
_u[3678].,2,VoiceMail2(${EXTEN}) without success. Same error
in the sip debug. It seems like this pattern matching should solve
the problem. It is unclear why the pattern is not matched. Is the
syntax correct?
-Steve
David J Carter wrote:
Steve,
Sorry bum information. Line 2 should read: -
exten => s,2,VoiceMail2(${EXTEN})
Don't need to strip the first digit as this is either u or b already,
(Unobtainable or Busy).
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of David J
Carter
Sent: 06 February 2005 12:30
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Help with extensions
Steve,
I haven't tried this but can't you do something like.
[from-proxy]
exten => s,1,Answer
exten => s,2,VoiceMail2(${EXTEN:1})
exten => 3,3,Hangup
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve Blair
Sent: 06 February 2005 12:14
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Help with extensions
Hello:
I'd like some help with defining extension rules. I want calls arriving
at Asterisk from my SIP proxy to be sent directly to voicemail. I'd
also like the appropriate greeting played when the call gets to
voicemail.
My proxy prefixes the extension with a "u" or "b" based on
SIP response codes before relaying to Asterisk. So when the
call arrives it is in the format [u|b][3|6|7|8]
If I hard code the following rules then calls get forwarded
as expected.
exten => u67501,1,VoiceMail2(${EXTEN})
exten => #,2,Hangup
However to save on typing I'd like a general rule. I've tried
the following but Asterisk cannot find the extension with this
set of rules. Can someone explain how what I want can be
accomplished?
exten => _[ub][3678].,2,VoiceMail2(${EXTEN})
exten => #,2,Hangup
Thanks
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[Asterisk-Users] FYI - New firmware from Sipura
FYI & OT, Looks like Sipura released new firmware for the majority of their products. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FIX YOUR AUTO-RESPONDERS!!!
On Sat, Feb 05, 2005 at 01:54:58PM +0100, Stefan Gofferje wrote: > 2.) The BOFH-way of handling such things is more fun *evilgrin*. Another BOFH method is to archive all of those messages in a public place. I know of someone who did that (in the linux-s390 list, IIRC, I can't recall his name and can't find a link to that page) -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Polycom Auto-Answer and Call Transfers
I have my * and polycom system setup to do Auto-Answer
for internal SIP/Staff calls, and I am running into an
issue with this and the polycom call transfer feature.
* is seeing a new call come through from the polycom
and is then transferring the call over. I need to know
if there is some way I can grab a message from the SIP
header or something to determine if I should not set
the ALERT_INFO tag to A-A. I would greatly appreciate
it if someone could help me out with this, I need to
have this resolved by Monday.
Instead of trying to hack something out of the SIP headers, you could have
the dialplan take care of this. Maybe something like the solution indicated in
the WIKI for the Polycom auto-answer - dial 8 before an extension to
auto-answer, otherwise the extension will ring. Something like:
exten => _8XXX,1,SetVar(ALERT_INFO=A-A)
exten => _8XXX,2,Dial(SIP/${EXTEN:1},20)
exten => _8XXX,3,Hangup
I know this isn't exactly what you were looking for because it means the users
have to remember to dial a different number to intercom than to transfer with
ringing, but it works. If you are under a time crunch you could do this
immediately, and figure out the SIP header hacking as time permits.
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Re: [Asterisk-Users] Proxied SIP
zoa. Todd Lieberman wrote: > Chris Tooley wrote: > >> I want to install Asterisk for an organization that wants it to do >> some call routing for them. They have a SIP provider that is going to >> provide one termination and one origination account. >> >> We are going to have to route a rather large number of calls >> (50-100,000 concurrent), but can't find any information on how to >> proxy calls adaquately. >> ___ >> Asterisk-Users mailing list >> Asterisk-Users@lists.digium.com >> http://lists.digium.com/mailman/listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> > look to SER but 100,000 calls requires a tremendious amount of > bandwidth, make sure you have it! > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FIX YOUR AUTO-RESPONDERS!!!
I think most people have spent more time complaining about the AUTO-RESPONDERS than it takes to hit the delete key. > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Wilson Pickett > Sent: Sunday, February 06, 2005 4:35 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] FIX YOUR AUTO-RESPONDERS!!! > > > > They can always check the archives to read up on missed posts, and it > > > would save us all the trouble in the mean time ;-) > > Isn't it obvious that with a choice of hundreds of free email > providers, anyone who wants to avois this problem need only use a > throwaway account like gmail with mailing lists and avoid checking the > "away message" stuff. > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FIX YOUR AUTO-RESPONDERS!!!
On Sun, Feb 06, 2005 at 08:34:07AM -0700, Keith Burns wrote: > I think most people have spent more time complaining about the > AUTO-RESPONDERS than it takes to hit the delete key. $ grep -c ^From ~/Mail/outofoffice 168 And this is from a realitively short period of time in a mailing list with "OOO-ignorants". The result: I naturally added a procmail rule to filter it to a folder. So this means that hence forth it has not served its purpose for me when I did need to be informed about some exchange/notes user out-of-office. The funniest thing is the non-English messages. ××× × . -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FIX YOUR AUTO-RESPONDERS!!!
That's brilliant. So you want everyone to fix the symptom instead of the problem. I guess that makes sense to some people. Jeremiah C. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Keith Burns Sent: Sunday, February 06, 2005 9:34 AM To: 'Wilson Pickett'; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] FIX YOUR AUTO-RESPONDERS!!! I think most people have spent more time complaining about the AUTO-RESPONDERS than it takes to hit the delete key. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TAPI integration with * using Identapop software
I haven't tried identapop, but an alternative is to use netcat along with YAC listener on the windows PC. See the wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk%20call%20notification Works well for me. On Sat, 5 Feb 2005 11:49:03 +, John Middleton <[EMAIL PROTECTED]> wrote: > Hi, > I've got Outlook to call the number on * using the TAPI interface > documented on the Wiki. Its working OK. > > I have downloaded the Indentapop application, and it appears to > connect to * Ok using the Debug modes, but It isnt detecting incoming > calls. > > Has anyone git identapop working? > > Care to share configuration details? > > Thanks > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help with extensions
I have had the same issue when receiving a call from an IAX provider.
Here is what I did to solve it.
[from-proxy]
exten => .,1,Goto(voicemail-direct,s,1)
[voicemail-direct]
exten => s,1,Answer
exten => s,2,VoiceMail2(${EXTEN:1})
exten => 3,3,Hangup
Not sure it does not pattern match like it should be the above worked
for me.
Robert
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Steve Blair
> Sent: Sunday, February 06, 2005 8:43 AM
> To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
> Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Help with extensions
>
>
> Nope. Didn't work. When the call hits extensions.conf no
> matching extension definition exists. I tried replacing line
> #2, exten => s,2,VoiceMail2(${EXTEN}) with exten =>
> _u[3678].,2,VoiceMail2(${EXTEN}) without success. Same error
> in the sip debug. It seems like this pattern matching should
> solve the problem. It is unclear why the pattern is not
> matched. Is the syntax correct?
>
> -Steve
>
> David J Carter wrote:
>
> >Steve,
> >
> >Sorry bum information. Line 2 should read: -
> >
> >exten => s,2,VoiceMail2(${EXTEN})
> >
> >Don't need to strip the first digit as this is either u or b
> already,
> >(Unobtainable or Busy).
> >
> >Regards
> >
> >Dave
> >
> >-Original Message-
> >From: [EMAIL PROTECTED]
> >[mailto:[EMAIL PROTECTED] Behalf Of David J
> >Carter
> >Sent: 06 February 2005 12:30
> >To: Asterisk Users Mailing List - Non-Commercial Discussion
> >Subject: RE: [Asterisk-Users] Help with extensions
> >
> >
> >Steve,
> >
> >I haven't tried this but can't you do something like.
> >
> >[from-proxy]
> >exten => s,1,Answer
> >exten => s,2,VoiceMail2(${EXTEN:1})
> >exten => 3,3,Hangup
> >
> >Regards
> >
> >Dave
> >
> >-Original Message-
> >From: [EMAIL PROTECTED]
> >[mailto:[EMAIL PROTECTED] Behalf Of Steve
> >Blair
> >Sent: 06 February 2005 12:14
> >To: Asterisk Users Mailing List - Non-Commercial Discussion
> >Subject: [Asterisk-Users] Help with extensions
> >
> >
> >
> >Hello:
> >
> > I'd like some help with defining extension rules. I want calls
> >arriving at Asterisk from my SIP proxy to be sent directly to
> >voicemail. I'd also like the appropriate greeting played
> when the call
> >gets to voicemail.
> >
> > My proxy prefixes the extension with a "u" or "b" based on SIP
> >response codes before relaying to Asterisk. So when the call
> arrives it
> >is in the format [u|b][3|6|7|8]
> >
> > If I hard code the following rules then calls get forwarded as
> >expected.
> > exten => u67501,1,VoiceMail2(${EXTEN})
> > exten => #,2,Hangup
> >
> > However to save on typing I'd like a general rule. I've tried the
> >following but Asterisk cannot find the extension with this set of
> >rules. Can someone explain how what I want can be accomplished?
> >
> >exten => _[ub][3678].,2,VoiceMail2(${EXTEN})
> >exten => #,2,Hangup
> >
> >Thanks
> >___
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> >Asterisk-Users@lists.digium.com
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
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> >
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> >
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RE: [Asterisk-Users] Callerid problems with 1.0.5
Hi Remco, > -Original Message- > Isn't it possibly to change caller id 'on the fly' like most > pbx's do? If > you do a call transfer you can see the local extension first and the > caller id of the incoming call afterwards? Within the Asterisk core you probably could do that, but it is heavily dependant on the equipment used (ATA's, IP-phones, etc) wether or not the change can be made visible to the end user. Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Waiting on X100P
http://voip-info.org/tiki-index.php?page=Asterisk%20vertical%20service%20activation%20codes I have never done this, so YMMV But searching the wiki can be quite usefull and enlightening. Lyle - Original Message - From: "Derek Whitten" <[EMAIL PROTECTED]> To: "asterisk-users" Sent: Saturday, February 05, 2005 9:59 PM Subject: Re: [Asterisk-Users] Call Waiting on X100P > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Question about X100P card
Define standard! The X100P was designed for NorthAmerican phone standards and can be used to bring a single dialtone line into asterisk. Lyle - Original Message - From: "Yousri Farouk" <[EMAIL PROTECTED]> To: Cc: "Asterisk Developers Mailing List" Sent: Sunday, February 06, 2005 6:56 AM Subject: [Asterisk-Users] Question about X100P card > Hello my brothers and sisters, > > Is "X100P" card suitable to VoIP? and if "yes", am i need to only "X100P" > and "Asterisk" Package? or i need also to other cards or packages? > and if "X100P" card not suitable to VoIP, please recommend a another card, > (please take in your account that i would like to connect standard analog > line to the card directly). > > Thanks in advance > Eng. Yousri Farouk > www.egypt.com > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] inter asterisk
I think it has to do with my ZAP interface. Before my DIAL(ZAP/1/51412345678) I have a Playback(message-transfert) which play nicely. As soon as the ZAP start ringing the PSTN phone, i have that helicopter sound. - Original Message - From: "David J Carter" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Sunday, February 06, 2005 7:18 AM Subject: RE: [Asterisk-Users] inter asterisk One thing I do on remote sites is set up a soft phone so I can call myself, this proves out the link and quality before anything else. DIAX id good for this as you can connect to multiple sites, also good to see if you have problems before anyone else calls you to say there is a problem. It also helps in cases like this, if your return quality is good then the possible fault lies with the ZAP interface. Process of elimination, works for me every time. Regards Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Ousmane Doukara Sent: 06 February 2005 08:46 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] inter asterisk Hi, I am trying to forward calls to another * server with IAX Here is What I want to Do 1- Call SERVER1, let say at 51412345678 2- SERVER1 should transfer the call to SERVER2 in a remote location 3- SERVER2 Receive the call and transfer it to the PSTN number. I have one X100P card on each machine. What is happening is that when the remote party picks up the phone, all he can hear is a weird sound. CONFIGS: SERVER1: zaptel.conf - ~ [channels] ~ language=fr ~ context=montréal ~ signalling=fxs_ks ~ usercallerid=yes ~ callwaiting=yes ~ threewaycalling=yes ~ transfer=yes ~ cancellforward=yes ~ echocancel=yes ~ echocancelwhenbridged=yes ~ echotraining=yes ~ relaxdtmf=yes ~ busydetect=yes ~ busycount=4 ~ callprogress=yes ~ group=1 ~ channel=>1 -- (same for SERVER2) IAX.conf ~ [general] ~ bindport=4569 ~ delayreject=yes ~ language=fr ~ allow=all ~ jutterbuffer=no ~ register => username:[EMAIL PROTECTED] ~ tos=lowdelay ~ autokill=yes ~ ~ [quebec] ~ type=friends ~ username = username ~ password=password ~ context=montréal ~ host=Dynamic ~ secret = password ~ disallow = all ~ allow=ulaw ~ allow=gsm extensions.conf --(Same for SERVER2 but no registration) ~ [general] ~ static=yes ~ writeprotect=yes ~ autofallthrough=yes ~ [montréal] ~ exten=>s,1,Answer ~ exten=>s,2,Playback(message-transfer) ~ exten=>s,3,Dial(IAX2/username:[EMAIL PROTECTED]/[EMAIL PROTECTED] al) ; always the same number ~ exten=>s,4,Hangup My remote server receive the call, answer the line and then Dial(ZAP/1/51412345678). So far so good. But when 51412345678 pickup the phone, all she can hear is a weird sound. What am I doing wrong ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iax2-jitter-trunking?
Two cvs-head asterisk boxes with iax2 working fine (without register
statements).
When two calls are placed simultanously from system A -> B and the packets
are sniffed on the wire, I see the two calls using two different udp
packets. At the top of iax.conf I have trunk=yes and jitterbuffer=yes
(at both ends).
I was expecting to see both calls handled within a single udp packet,
but that's not happening. Each iax2 packet is 79 bytes using ethereal.
I've tried the trunk=yes both within the inbound context and at the top
of the iax.conf file (assuming the one at the top would be used for all
outbound iax calls that don't reference a context). Calls are placed
with:
exten => _2.,1,Dial(IAX2/user:[EMAIL PROTECTED]/${EXTEN:1})
Is trunking dependent upon the use of 'register'? Or, dependent on the
above exten=>_2., referencing a context (instead of the IP directly)?
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Re: [Asterisk-Users] Call Waiting on X100P
*0 doesn't work.. tried that already... On Sun, 2005-02-06 at 08:24, Lyle Giese wrote: > http://voip-info.org/tiki-index.php?page=Asterisk%20vertical%20service%20activation%20codes > > I have never done this, so YMMV > > But searching the wiki can be quite usefull and enlightening. > > Lyle > > - Original Message - > From: "Derek Whitten" <[EMAIL PROTECTED]> > To: "asterisk-users" > Sent: Saturday, February 05, 2005 9:59 PM > Subject: Re: [Asterisk-Users] Call Waiting on X100P > > > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users -- S***I***G***N***A***T***U***R***E*** ,(), ,(,. >>huh-huh<< ,---,,,_ ()))//((\ Check it out, ( )) (\\( \))( \(/)Beavis...we're, () /( \\ like, in "ASS-kee." () // _ \>>huh-huh-huh<< (_(_ ) // \ /\ / (, \) \ (. .\ / | / ) ) (, |,) Yeah. >>heh-heh<< |\ /( ) \ ^\/^ / That's COOL! Hey, (.(.)S ) \ / Butt-Head...you're/_ \ ) \ (-<>-) / an "ASS-kee." \ /__) ^ \/ \ -- / >>heh-heh<< //| \ __ / )__| | | //\/\\/\//\/\//\/\\\/\\ | __-|__|-__ \ / __-\__|-__ ( ) > BEAVIS AND BUTT-HEAD <( ) |_|AC//DC|_| / \|_| MTVu |_| | | | | \/\//\/\\/\\/\//\//\/\ TM | | | | signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 12SP+ firware anyone?
Anyone know where I can lay my hands on some Skinny firmware for some Cisco 12SP+ phone I got at a yard sale this morning? Thanks Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FIX YOUR AUTO-RESPONDERS!!!
Yah .. or getting "message refused because of password protected attachment" .. aka. GPG signature.. On Sun, 2005-02-06 at 07:57, Jeremiah Chapman wrote: > That's brilliant. So you want everyone to fix the symptom instead of the > problem. I guess that makes sense to some people. > > > Jeremiah C. > > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Keith Burns > Sent: Sunday, February 06, 2005 9:34 AM > To: 'Wilson Pickett'; 'Asterisk Users Mailing List - Non-Commercial > Discussion' > Subject: RE: [Asterisk-Users] FIX YOUR AUTO-RESPONDERS!!! > > I think most people have spent more time complaining about the > AUTO-RESPONDERS than it takes to hit the delete key. > > > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- S***I***G***N***A***T***U***R***E*** ,(), ,(,. >>huh-huh<< ,---,,,_ ()))//((\ Check it out, ( )) (\\( \))( \(/)Beavis...we're, () /( \\ like, in "ASS-kee." () // _ \>>huh-huh-huh<< (_(_ ) // \ /\ / (, \) \ (. .\ / | / ) ) (, |,) Yeah. >>heh-heh<< |\ /( ) \ ^\/^ / That's COOL! Hey, (.(.)S ) \ / Butt-Head...you're/_ \ ) \ (-<>-) / an "ASS-kee." \ /__) ^ \/ \ -- / >>heh-heh<< //| \ __ / )__| | | //\/\\/\//\/\//\/\\\/\\ | __-|__|-__ \ / __-\__|-__ ( ) > BEAVIS AND BUTT-HEAD <( ) |_|AC//DC|_| / \|_| MTVu |_| | | | | \/\//\/\\/\\/\//\//\/\ TM | | | | signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Proxied SIP
I have been playing with SER, but we need something to proxy the initial part of an origination (outbound) call, and then negotiate it's way out of the middle of the call. Like redirect appears to work with calls that are termination (inbound). Chris On Sun, 06 Feb 2005 08:13:02 -0500, Todd Lieberman <[EMAIL PROTECTED]> wrote: > Chris Tooley wrote: > > >I want to install Asterisk for an organization that wants it to do > >some call routing for them. They have a SIP provider that is going to > >provide one termination and one origination account. > > > >We are going to have to route a rather large number of calls > >(50-100,000 concurrent), but can't find any information on how to > >proxy calls adaquately. > >___ > >Asterisk-Users mailing list > >Asterisk-Users@lists.digium.com > >http://lists.digium.com/mailman/listinfo/asterisk-users > >To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > look to SER but 100,000 calls requires a tremendious amount of > bandwidth, make sure you have it! > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] blindxfer not in stable 1.0.5?
Am I missing something here or is the blindxfer option in features.conf not an option in 1.0.5? If not, is it going to be anytime soon? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Proxied SIP
I have solved this with IAX loop on asterisk. Internal SIP phone - SIP asterisk inbound - IAX call to the same Asterisk box - Outbound SIP call. Calls are bridged and outgoing calls are appear to be from asterisk IP. This solves also incoming calls. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Tooley Sent: Sunday, February 06, 2005 8:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Proxied SIP I have been playing with SER, but we need something to proxy the initial part of an origination (outbound) call, and then negotiate it's way out of the middle of the call. Like redirect appears to work with calls that are termination (inbound). Chris On Sun, 06 Feb 2005 08:13:02 -0500, Todd Lieberman <[EMAIL PROTECTED]> wrote: > Chris Tooley wrote: > > >I want to install Asterisk for an organization that wants it to do > >some call routing for them. They have a SIP provider that is going to > >provide one termination and one origination account. > > > >We are going to have to route a rather large number of calls > >(50-100,000 concurrent), but can't find any information on how to > >proxy calls adaquately. > >___ > >Asterisk-Users mailing list > >Asterisk-Users@lists.digium.com > >http://lists.digium.com/mailman/listinfo/asterisk-users > >To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > look to SER but 100,000 calls requires a tremendious amount of > bandwidth, make sure you have it! > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] First Call straight to my extension
Thanks anyway. But what files is this? that I have to play with -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Eric Wieling Sent: Friday, February 04, 2005 1:39 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] First Call straight to my extension [EMAIL PROTECTED] wrote: >>Anybody with an idea how will I set me Asterisk to send call straight to my >>extension with out playing demo-Congrat MENU. > > Comething like this > > exten => s,1,Answer > exten => s,2,Dial(IAX2/2001,20,tr) > exten => s,3,Voicemail(u2001) > exten => s,4,Voicemail(b2001) or exten => s,1,Dial(IAX2/2001,20) exten => s,2,Voicemail(u2001) exten => s,102,Voicemail(b2001) Advantages are no fake ringing, phone does transfers not Asterisk pound transfer, no answer so caller doesn't pay toll charges, busy voicemail works. Disadvantage: Asterisk can't tell if they hang up before going to voicemail. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] First Call straight to my extension
Thaniks anyway but where and what files is this to be motify. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Friday, February 04, 2005 1:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] First Call straight to my extension > Anybody with an idea how will I set me Asterisk to send call straight to my > extension with out playing demo-Congrat MENU. Comething like this exten => s,1,Answer exten => s,2,Dial(IAX2/2001,20,tr) exten => s,3,Voicemail(u2001) exten => s,4,Voicemail(b2001) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Autodetecting faxes
I have managed to get spandsp working, and if I dial a specific
extension I can receive faxes. WhooHoo.
However, I was wanting to use the "fax detect" option in order to allow
individuals to receive faxes, but can't get that to work.
Given the following extensions (mainly copied from examples on the
wiki), why is the call simply passed onto the sip device rather than
being detected as a fax ?
Asterisk CVS-HEAD-02/02/05-07:11:04 built by asterisk on a i686 running
Linux
Spandsp is 0.2pre9
Incoming lines are E1 line 30 channels PRI.
Many thanks.
Julian.
===
exten => 442781,1,Goto(fax,1,1) ; dialling this number works
exten => 442781,2,Hangup()
exten => _4427XX,1,Answer() ; dialling any number in here does not
exten => _4427XX,2,Macro(dialsip,${EXTEN:3})
exten => _4427XX,3,Hangup()
exten => fax,1,Goto(fax,1,1)
[macro-faxreceive]
exten => s,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten => s,2,DBGet(EMAILADDR=extensionemail/${MACRO_EXTEN})
exten => s,3,rxfax(${FAXFILE})
exten => s,103,SetVar([EMAIL PROTECTED])
exten => s,104,Goto(3)
[fax]
exten => 1,1,Macro(faxreceive)
exten => h,1,system(/usr/sbin/mailfax ${FAXFILE} ${EMAILADDR} \
"${CALLERIDNUM} ${CALLERIDNAME}")
exten => h,2,Hangup()
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[Asterisk-Users] Call status after Answer
Hi, I setup asterisk as an autoattendant. When I call using IAX I get the autoattendent okay, but when I dial one of the extensions, there is no ringing sound passed back to the caller. It happens when I use my DID number, but I also configured a context so I can get it to happen with Firefly (iax client) as the caller. It seems that once the Answer command is executed in the dialplan, status commands (RINGING, etc) aren't passed back through the IAX channel. My only workaround has been to use music on hold instead of making a ringing sound. Has anyone seen this, or a solution. It seems basic, but I have been working all day on it. I've tried this with 1-0-2, 1-0.5 and the head version, but all behave the same. The IAX trace shows that RINGING is getting sent back to the client. Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Autodetecting faxes
Asterisk wrote:
I have managed to get spandsp working, and if I dial a specific
extension I can receive faxes. WhooHoo.
However, I was wanting to use the "fax detect" option in order to
allow individuals to receive faxes, but can't get that to work.
Given the following extensions (mainly copied from examples on the
wiki), why is the call simply passed onto the sip device rather than
being detected as a fax ?
Asterisk CVS-HEAD-02/02/05-07:11:04 built by asterisk on a i686
running Linux
Spandsp is 0.2pre9
Incoming lines are E1 line 30 channels PRI.
Many thanks.
Julian.
===
exten => 442781,1,Goto(fax,1,1) ; dialling this number works
exten => 442781,2,Hangup()
exten => _4427XX,1,Answer() ; dialling any number in here does not
add this:
exten => _4427XX,2,Wait(3)
\/
exten => _4427XX,3,Macro(dialsip,${EXTEN:3})
exten => _4427XX,4,Hangup()
exten => fax,1,Goto(fax,1,1)
[macro-faxreceive]
exten => s,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten => s,2,DBGet(EMAILADDR=extensionemail/${MACRO_EXTEN})
exten => s,3,rxfax(${FAXFILE})
exten => s,103,SetVar([EMAIL PROTECTED])
exten => s,104,Goto(3)
[fax]
exten => 1,1,Macro(faxreceive)
exten => h,1,system(/usr/sbin/mailfax ${FAXFILE} ${EMAILADDR} \
"${CALLERIDNUM} ${CALLERIDNAME}")
exten => h,2,Hangup()
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Re: [Asterisk-Users] Autodetecting faxes
Hmm, a little more digging got me further : I found the option
faxdetect=both needed to be set in zapata.conf.
Put this in. Works like a charm. Well, almost - the phone rang for about
three rings before the autodetect kicked in and started the fax receive.
Getting there ..
Julian.
Asterisk wrote:
I have managed to get spandsp working, and if I dial a specific
extension I can receive faxes. WhooHoo.
However, I was wanting to use the "fax detect" option in order to allow
individuals to receive faxes, but can't get that to work.
Given the following extensions (mainly copied from examples on the
wiki), why is the call simply passed onto the sip device rather than
being detected as a fax ?
Asterisk CVS-HEAD-02/02/05-07:11:04 built by asterisk on a i686 running
Linux
Spandsp is 0.2pre9
Incoming lines are E1 line 30 channels PRI.
Many thanks.
Julian.
===
exten => 442781,1,Goto(fax,1,1) ; dialling this number works
exten => 442781,2,Hangup()
exten => _4427XX,1,Answer() ; dialling any number in here does not
exten => _4427XX,2,Macro(dialsip,${EXTEN:3})
exten => _4427XX,3,Hangup()
exten => fax,1,Goto(fax,1,1)
[macro-faxreceive]
exten => s,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten => s,2,DBGet(EMAILADDR=extensionemail/${MACRO_EXTEN})
exten => s,3,rxfax(${FAXFILE})
exten => s,103,SetVar([EMAIL PROTECTED])
exten => s,104,Goto(3)
[fax]
exten => 1,1,Macro(faxreceive)
exten => h,1,system(/usr/sbin/mailfax ${FAXFILE} ${EMAILADDR} \
"${CALLERIDNUM} ${CALLERIDNAME}")
exten => h,2,Hangup()
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[Asterisk-Users] Call forwarding of IAX inbound call
I am trying to do the following: 1. Call comes in to my * box over IAX (VP Connect DID) 2. Check to see if call should be forwarded to my cell 3. Forward the call to my cell phone and take * out of the media path. I am able to do all of the above except * is not able to natively bridge the call. I am using sixtel and for the call forward portion, but the calls don't connect before sixtel hangs up. -- Attempting native bridge of IAX2/[EMAIL PROTECTED]:4569/1 and IAX2/sixtel/3 -- Channel 'IAX2/sixtel/3' ready to transfer -- Releasing IAX2/sixtel/3 and IAX2/[EMAIL PROTECTED]:4569/1 -- Hungup 'IAX2/sixtel/3' Any thoughts on getting a native bridge between the two providers? In iax.conf I DO NOT have notransfer=yes anywhere and am using ULAW for both legs. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Autodetecting faxes
Ariel Batista wrote:
Asterisk wrote:
I have managed to get spandsp working, and if I dial a specific
extension I can receive faxes. WhooHoo.
However, I was wanting to use the "fax detect" option in order to
allow individuals to receive faxes, but can't get that to work.
Given the following extensions (mainly copied from examples on the
wiki), why is the call simply passed onto the sip device rather than
being detected as a fax ?
Asterisk CVS-HEAD-02/02/05-07:11:04 built by asterisk on a i686
running Linux
Spandsp is 0.2pre9
Incoming lines are E1 line 30 channels PRI.
Many thanks.
Julian.
===
exten => 442781,1,Goto(fax,1,1) ; dialling this number works
exten => 442781,2,Hangup()
exten => _4427XX,1,Answer() ; dialling any number in here does not
add this:
exten => _4427XX,2,Wait(3)
\/
Thanks for that - However, it still rings the extensions before
transfering to the fax function. I'm going to try increasing the number
of seconds and see what happens.
Julian
exten => _4427XX,3,Macro(dialsip,${EXTEN:3})
exten => _4427XX,4,Hangup()
exten => fax,1,Goto(fax,1,1)
[macro-faxreceive]
exten => s,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten => s,2,DBGet(EMAILADDR=extensionemail/${MACRO_EXTEN})
exten => s,3,rxfax(${FAXFILE})
exten => s,103,SetVar([EMAIL PROTECTED])
exten => s,104,Goto(3)
[fax]
exten => 1,1,Macro(faxreceive)
exten => h,1,system(/usr/sbin/mailfax ${FAXFILE} ${EMAILADDR} \
"${CALLERIDNUM} ${CALLERIDNAME}")
exten => h,2,Hangup()
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Re: [Asterisk-Users] Autodetecting faxes
Asterisk wrote: Hmm, a little more digging got me further : I found the option faxdetect=both needed to be set in zapata.conf. Put this in. Works like a charm. Well, almost - the phone rang for about three rings before the autodetect kicked in and started the fax receive. Getting there .. Julian. I have managed to get spandsp working, and if I dial a specific extension I can receive faxes. WhooHoo. However, I was wanting to use the "fax detect" option in order to allow individuals to receive faxes, but can't get that to work. Given the following extensions (mainly copied from examples on the wiki), why is the call simply passed onto the sip device rather than being detected as a fax ? What we found was that the fax/voice decision was being made before the intermittent "beep--beep--beep" fax tone was being generated, so it wasn't being detected. Changing the order of things in extensions.conf around a smidge got it all working nicely :- [inbound-from-pstn] include => default exten => s,1,Answer exten => s,2,Wait,1 exten => s,3,Playback(thank-you-for-calling-please-wait-a-moment) exten => fax,1,Macro(faxreceive) exten => s,4, The wait allows the start of the Playback to be heard by the caller - without it, we were finding the first word clipped. That second plus the duration of the "Thank you for calling" message gives enough time for the roughly 2.5sec duration between fax beeps to repeat, no matter when it last fell compared to the answer. We've not checked more into the three rings before answer, but there's been discussion (here? elsewhere?) that it's down to the wait for caller ID. Try turning that off. TBH, I *like* the three rings - as a caller, it psychologically gives you time to get your head in gear before the call's answered. Besides - If you're ringing from a mobile, it also gives you time to physically put the phone to your ear... -- Adrian Chapman Director Trivas Ltd Business on the Move Mobility - Messaging - Infrastructure - Security - Remote Access 07796 690210 - 01582 626552 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] snom soft phone
Some of you might already know that we are releasing a new phone, snom 360. To make the phone well-known and stable, we have made a soft phone version out of it and offer it for trial or private use for free (for more details, see the license conditions). There are only few limitations to the phone. First of all, the audio subsystem will work only work with an acceptable quality if you are using a headset. We only offer G.711 and GSM in this build (G729A can be enabled by an additional license). Other features like transfer, Security and LEDs should work just like in the "real" phone. It is available at http://snom.com/download/snom360-3.57j.exe (Windows only). To use the free version, just leave the license code field empty. Please write your comments to mailto:[EMAIL PROTECTED] Please treat this version as a beta version. We are looking forward to a close integration with Asterisk! Enjoy! Christian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicepulse DNID is blank - Any other options?
I just signed up for a second voicepulse number. I assumed that I would be able to differentiate which number the caller dialed. But DNID is empty and DIALEDPEERNUMBER is empty and CHANNEL contains the same info (almost, with the exception of a randomly assigned suffix) for both numbers. Does anyone know how I might determine which number was called? Note, this is not CALLERID. I need the number that the caller CALLED. As a last resort, I guess I could use a different provider for the second number. Can anyone shed any light? Thanks in advance! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] snom soft phone
Congrats! I will make sure we take a look at it right away. Mike R. Jenson LiveVoip LLC http://www.livevoip.com [EMAIL PROTECTED] Some of you might already know that we are releasing a new phone, snom 360. To make the phone well-known and stable, we have made a soft phone version out of it and offer it for trial or private use for free (for more details, see the license conditions). There are only few limitations to the phone. First of all, the audio subsystem will work only work with an acceptable quality if you are using a headset. We only offer G.711 and GSM in this build (G729A can be enabled by an additional license). Other features like transfer, Security and LEDs should work just like in the "real" phone. It is available at http://snom.com/download/snom360-3.57j.exe (Windows only). To use the free version, just leave the license code field empty. Please write your comments to mailto:[EMAIL PROTECTED] Please treat this version as a beta version. We are looking forward to a close integration with Asterisk! Enjoy! Christian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP URI modified unexpectedly! Is that a router problem?
Hi, I set up an asterisk server at my home computer, and the * box is configured as the DMZ of my ADSL modem/router. I found the SIP URI in an INVITE message has been changed before it reaches the * server. In my setup, I have a SJPhone software installed in with a public IP yy.yy.yy.yy. (ph no. 10930) I have a unknown brand SIP phone in the same network as the * sever is in. The public IP is xx.xx.xx.xx, phone no =10916. Here is the SJphone Log of an INVITE message sent from 10930 to 10916. 10:42:05 INFO Initiating SIP call to sip:[EMAIL PROTECTED]:5060 10:42:05 DEBUG 2005-02-05 21:42:05.937 UDP LOCAL->xx.xx.xx.xx:5060 INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Content-Length: 341 Contact: Call-ID: [EMAIL PROTECTED] Content-Type: application/sdp From: "Dan";tag=588746825471 CSeq: 1 INVITE Max-Forwards: 70 To: Via: SIP/2.0/UDP yy.yy.yy.yy;rport;branch=z9hG4bKcb617ac20131c9b142053dad79830032 User-Agent: SJLabs-SJphone/1.40.258 --snip--- Here is the corresponding message I received in my server, the output of "ngrep 10930 port 5060 -d eth0" . ### U yy.yy.yy.yy:5060 -> 192.168.1.2:5060 INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0..Content-Length: 341..Co ntact: ..Call-ID: C64037DF-A3BA-4F1D-8FE9-84 [EMAIL PROTECTED]: application/sdp..From: "Dan";tag=588746825471..CSeq: 1 INVITE..Max-Forwards: 7 0..To: ..Via: SIP/2.0/UDP 203.97.122.1 94;rport;branch=z9hG4bKcb617ac20131c9b142053dad79830032..User-Agent : SJLabs-SJphone/1.40.258v=0..o=- 3316628525 3316628525 IN IP4 203.97.1 22.194..s=SJphone..c=IN IP4 yy.yy.yy.yy..t=0 0..a=direction:active..m=au dio 16386 RTP/AVP 0 8 3 97 98 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/80 00..a=rtpmap:3 GSM/8000..a=rtpmap:97 iLBC/8000..a=rtpmap:98 iLBC/8000..a=fm tp:98 mode=20..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-11,16.. # Obviously, they are two places in the received messages changed (the INVITE tag and the "To" header), from to sip:[EMAIL PROTECTED]:5060 As a result, I got a 404 not found error. Before the error, there is one line in the CLI output saying: Looking for 192.169.1.2 in local ... I think it should have looked for 10916 in local (the context defined in extension.conf). Has anyone here experienced similar problem? Can I say my router is not VoIP friendly? Cheers, Dan _ Dont just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicepulse DNID is blank - Any other options?
Maybe I am missing your exact point, but what about handling this in your extensions.conf [voicepulse-incoming] exten => 2124007999,1,Goto(nyc,s,1) exten => 2124007998,1,Goto(nyc2,s,1) That will put calls to 2124007999 into context nyc and calls to 2124007998 into context nyc2. I guess the real questions is what is your ultimate goal? On Sun, 6 Feb 2005 12:52:21 -0800 (PST), Tim Burt <[EMAIL PROTECTED]> wrote: > I just signed up for a second voicepulse number. > > I assumed that I would be able to differentiate which number the caller > dialed. > > But DNID is empty and DIALEDPEERNUMBER is empty and CHANNEL contains the > same info (almost, with the exception of a randomly assigned suffix) for > both numbers. > > Does anyone know how I might determine which number was called? > > Note, this is not CALLERID. I need the number that the caller CALLED. > > As a last resort, I guess I could use a different provider for the second > number. > > Can anyone shed any light? > > Thanks in advance! > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call status after Answer
Who is your DID provider? On Sun, 6 Feb 2005 15:03:19 -0500, Scott Simpson <[EMAIL PROTECTED]> wrote: > Hi, > I setup asterisk as an autoattendant. When I call using IAX I get the > autoattendent okay, but when I dial one of the extensions, there is no > ringing sound passed back to the caller. > > It happens when I use my DID number, but I also configured a context so I > can get it to happen with Firefly (iax client) as the caller. It seems that > once the Answer command is executed in the dialplan, status commands > (RINGING, etc) aren't passed back through the IAX channel. > > My only workaround has been to use music on hold instead of making a ringing > sound. > > Has anyone seen this, or a solution. It seems basic, but I have been > working all day on it. > > I've tried this with 1-0-2, 1-0.5 and the head version, but all behave the > same. The IAX trace shows that RINGING is getting sent back to the client. > > Thanks > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Using Asterisk to monitor in/out calls (single line)
Hello group, Please don't shoot the new guy. I have a linux workstation with a phone-in and a phone-out jack and a single phone line. I would like to use Asterisk (if possible) to monitor/log in and out calls, including logging the phone number -- pretty basic. I would also like to play with possible call filtering (sending unwanted calls to voice mail). Are these things that I can do with Asterisk or am I under-using it? If so, do I need to build any of the ISDN or Telephony support driovers into my kernel to support Asterisk? I ask because I build my own kernels and I usually leave that stuff out. Thanks. Job___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] First Call straight to my extension
> Thanks anyway. But what files is this? that I have to play with extensions.conf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax-modem
Does any body recommend a fax-modem that works well with Asterisk? Does any modem works fine with Asterisk? Thanks to any info __ Do you Yahoo!? The all-new My Yahoo! - What will yours do? http://my.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicepulse DNID is blank - Any other options?
Ah.. the obvious. I don't know why I missed it. I am just a newbie at this PBX stuff. Thanks for the pointer. It worked. First off. Hopefully, someday soon, I will contribute more than silly questions to this list! Thanks again! > Maybe I am missing your exact point, but what about handling this in > your extensions.conf > > [voicepulse-incoming] > exten => 2124007999,1,Goto(nyc,s,1) > exten => 2124007998,1,Goto(nyc2,s,1) > > That will put calls to 2124007999 into context nyc and calls to > 2124007998 into context nyc2. > > I guess the real questions is what is your ultimate goal? > > On Sun, 6 Feb 2005 12:52:21 -0800 (PST), Tim Burt > <[EMAIL PROTECTED]> wrote: >> I just signed up for a second voicepulse number. >> >> I assumed that I would be able to differentiate which number the caller >> dialed. >> >> But DNID is empty and DIALEDPEERNUMBER is empty and CHANNEL contains the >> same info (almost, with the exception of a randomly assigned suffix) for >> both numbers. >> >> Does anyone know how I might determine which number was called? >> >> Note, this is not CALLERID. I need the number that the caller CALLED. >> >> As a last resort, I guess I could use a different provider for the >> second >> number. >> >> Can anyone shed any light? >> >> Thanks in advance! >> >> ___ >> Asterisk-Users mailing list >> Asterisk-Users@lists.digium.com >> http://lists.digium.com/mailman/listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users >> > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicepulse DNID is blank - Any other options?
A lot of times we all overlook the obvious or easiest way to do things :) On Sun, 6 Feb 2005 13:26:25 -0800 (PST), Tim Burt <[EMAIL PROTECTED]> wrote: > Ah.. the obvious. I don't know why I missed it. > > I am just a newbie at this PBX stuff. > > Thanks for the pointer. It worked. First off. > > Hopefully, someday soon, I will contribute more than silly questions to > this list! > > Thanks again! > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] no caller ID presented from 12SP+
Hi folks, I have got a Cisco 12SP+ working (thanks Derek!) but I'm having a minor issue with it. When I use it to call another desk I get no CallerID. The receiving phone diplsays "asterisk" as the CID. Below is the skinny.conf stanza. [2207] device=SEP00308062B006 version=P002L2J2 context=intern callerid="Luke's Room" <2207> mailbox=2207 transfer=1 callwaiting=1 threewaycalling=1 line => 2207 Thanks Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] snom soft phone
> It is available at http://snom.com/download/snom360-3.57j.exe (Windows > only). To use the free version, just leave the license code field empty. Unfortunately it does not run under wine/crossover office/cedega. Will there be a Linux version or at least a version that works with wine? I'd love to test it but I will not install Windows just for testing a phone. -- Maik Schmitthttp://graphics.cs.uni-sb.de/VoIP signature.asc Description: Digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iax hardphone
Is there such a beast yet available? I want to equip our exhibition staff with one to plug into their local net feed at whatever exhibition they pitch up at this week. It must be IAX as that's all I allow in through the firewall and it must be a hardphone cos they'll loose an ATA and a hone and the wires etc. Analogue phone costs are far too much money for the stand and cell phone rarely work in the exhibition halls for some odd reason. Thanks de Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Understanding the "Hint" priority.
Hi all, I'm trying to get a better understanding of the 'Hint' priority for use with Snom phones, etc. >From the Wiki, I understand that the following will work: exten => 200,hint,sip/200 exten => 200,1,Dial(sip/200) exten => 201,hint,sip/300 exten => 201,1,Dial(sip/201) When exactly is 'Hint' executed? After someone dials '200' or '201'? Or does the 'hint' priority apply it's magic at dialplan load time? Suppose I want to dial multiple extensions at once: exten => 999,1,Dial(sip/200&sip/201) .. can hint work in this case? If so, would I simply pass multiple channels to hint? ie: exten => 999,hint,sip/200&sip/201 exten => 999,1,Dial(sip/200&sip/201) ? Thanks in advance, Confused ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need help with perl script/agi for ringback
taf taffey a écrit :
Hi,
I'm trying to write a simple perl script that will run
the following:
Action: Originate
Channel: local/[EMAIL PROTECTED]/r/n
Exten: 1234
Context: callback
Priority: 1
Extensions.conf
exten => 500,1,agi,callback.pl
callback perl script:
use Net::Telnet ();
$mgrUSERNAME='fred';
$mgrSECRET='bloggs';
$server_ip='127.0.0.1';
$tn->print("Action: originate\nExten: 1234\nContext:
user\nChannel: local/[EMAIL PROTECTED]/r/n\nPriority:
1\nCallerid: 1234\n\n");
$tn->waitfor('/Event: Newchannel.*/') or die "Unable
to determine call status", $tn->lastline;
# wait for asterisk to process
$tn->print("Action: Logoff\n\n");
I'm not a programmer (as u can probably tell) so any
pointers would be much appreciated.
Cheers,
Taff.
You should take a look at the perl modules from
http://asterisk.gnuinter.net/
Thanks,
Jean-Denis
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Re: [Asterisk-Users] iax hardphone
Hi! > Is there such a beast yet available? - Digium IAXy - PA168 chipset: http://www.voip-info.org/tiki-index.php?page=PA168 - farfon (only test devices yet) - several products: http://www.iaxtalk.com/ Cheers, Philipp ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk@home basic
Thanks, timezone sorted but still having issues with e-mail, will work on it. Steve - Original Message - From: "Andrew Niemantsverdriet" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Saturday, February 05, 2005 6:21 PM Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] basic I too had a hard time with email notfications. I knew that the * box needed a smtp sever. I belive @home has send mail installed so you just need to add a smart host. Sendmail.org has infomation on setting that all up. Also try copying the correct /usr/share/zoneinfo file to /etc/localtime that should clear up any problems. On Sat, 5 Feb 2005 13:14:43 -, Steve Rawlings <[EMAIL PROTECTED]> wrote: Apologies for asking something that must have been asked many times. I'm running [EMAIL PROTECTED] v0.4 and can't get the * time to be local GMT. Tried tzselect etc etc and added ntp server addresses to ntp.conf, * still uses system time of EST so call logs are 5 hours behind. Also, e-mail notifications of vm don't appear to be getting sent, I've set voicemail.conf to include a valid e-mail address but I never receive it. Any pointers to subject posts would be appreciated, I'm not after detailed replies just a pointer to where this might already be documented, I've tried all the usual and obvious voip forums etc. Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Encrypted VOIP?
Wolfgang S. Rupprecht wrote: In theory, the Sipura line supports SRTP. I've got both a spa-841 and a spa-3000 that have config areas for loading the srtp rsa keys. Unfortunately there isn't enough information given by sipura as to how to generate these rsa keys. (eg. can one use an openssl generated key?) In the Sipura support area (authentication required), there is a tool to generate Mini Certificates for this. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] passing "*" into a dial plan
I'm trying to dial for example *67 to block caller ID but is it not
working.
I have SPA-3000 an dial plan:
exten => _9.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,tr)
If I try
exten => _3.,2,Dial(SIP/*67${EXTEN:[EMAIL PROTECTED],60,tr)
it gives me congestion.
How to pass "*" in a context when dialing a number?
--
#Joseph
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Re: [Asterisk-Users] iax2-jitter-trunking?
AFAIK, trunk=yes is not a global option. You set it within a context.
Also, using the jitter buffer with trunk=yes is not recommended since
its broken right now.
-mark
On Feb 6, 2005, at 12:45 PM, Rich Adamson wrote:
Two cvs-head asterisk boxes with iax2 working fine (without register
statements).
When two calls are placed simultanously from system A -> B and the
packets
are sniffed on the wire, I see the two calls using two different udp
packets. At the top of iax.conf I have trunk=yes and jitterbuffer=yes
(at both ends).
I was expecting to see both calls handled within a single udp packet,
but that's not happening. Each iax2 packet is 79 bytes using ethereal.
I've tried the trunk=yes both within the inbound context and at the top
of the iax.conf file (assuming the one at the top would be used for all
outbound iax calls that don't reference a context). Calls are placed
with:
exten => _2.,1,Dial(IAX2/user:[EMAIL PROTECTED]/${EXTEN:1})
Is trunking dependent upon the use of 'register'? Or, dependent on the
above exten=>_2., referencing a context (instead of the IP directly)?
--
Mark Eissler, [EMAIL PROTECTED]
Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com
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[Asterisk-Users] Soft keys and transfer problem on Sayson 480i
I have a strange problem with my Sayson 480i IP phone. If I press the Transfer button and then dial extension 200 to try to transfer the call, the Sayson apparently is treating the 200 as the last part of an IP address, and the call fails. As soon as I enter 200 and press Dial, I see an IP address of the form x.y.z.200 show up on the display. How do I get the phone to treat the dial string as an extension to be processed by the PBX rather than an IP address? If I simply pick up the phone and dial a 3-digit extension it works, but not when I try to Transfer to a 3-digit extension. Secondly, I have no documentation at all on how to program the 6 soft keys. It's not explained in the User Manual. Anyone know how? Thanks. -Ron ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] HP ProLiant server for Asterisk
Digium support are trailing some new firmware with the TE410P for machines with the Intel E75xx Chipsets that are having issues (such as the DL380 G4). I believe they are confident they have resolved the issue that prevents the cards working, but you may need to specifically mention that you are running this type of machine when acquiring the cards to ensure you get the 'in-testing' firmware. I'm sure as soon as someone gets and tests the new firmware on one of these machines they will post their results to the list. Regards, Peter -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Dana Olson Sent: Saturday, 5 February 2005 1:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] HP ProLiant server for Asterisk I'm looking at ordering a server from HP. I checked around on Google and found in the Wiki that the ProLiant DL380 is supposed to be known to work with *. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Soft keys and transfer problem on Sayson 480i
Asterisk Users wrote: I have a strange problem with my Sayson 480i IP phone. If I press the Transfer button and then dial extension 200 to try to transfer the call, the Sayson apparently is treating the 200 as the last part of an IP address, and the call fails. As soon as I enter 200 and press Dial, I see an IP address of the form x.y.z.200 show up on the display. How do I get the phone to treat the dial string as an extension to be processed by the PBX rather than an IP address? If I simply pick up the phone and dial a 3-digit extension it works, but not when I try to Transfer to a 3-digit extension. I haven't seen that problem specifically, but I have found numerous other bugs/annoyances that prevent me from using this phone as anything more than a toy. Secondly, I have no documentation at all on how to program the 6 soft keys. It's not explained in the User Manual. Anyone know how? Thanks. The only way to do this is when you provision the phone via TFTP. The Manual should have an example of an aastra.cfg file that includes soft keys (there are actually multiple pages of soft keys that you can create). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server
Contact Digium Support. They have been very helpful with this issue (mention your using the G4 server with the Intel E7520 Chipset..) Cheers, Peter -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Erick Perez Sent: Saturday, 5 February 2005 5:51 AM To: Eric Bishop; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server Has anyone on this list have a way to contact ServerWorks? they make the mobos for the G4. I dont have a G4 but i do know HP in the G line uses ServerWorks I have to make a full stop ordering on 2 G4 monsters because of this thread...However one friend is using a sangoma card without problems TE410P/ServerWork motherboard combo not working because of bus problems my less than 1 cent On Mon, 31 Jan 2005 20:42:47 +1100, Eric Bishop <[EMAIL PROTECTED]> wrote: > Did anyone get anywhere with this thread? Any HP G4 series servers working? > > > On Wed, 26 Jan 2005 09:46:31 +1100, Eric Bishop <[EMAIL PROTECTED]> wrote: > > Has anyone had any luck with this issue and new Asterisk/Zaptel > > releases (1.05/1.04)? I am still searching for a solution and waiting > > for that Eureka! moment.. > > > > > > On Thu, 20 Jan 2005 09:20:09 +0100, Tais M. Hansen <[EMAIL PROTECTED]> > > wrote: > > > On Wednesday 19 January 2005 23:15, Eric Bishop wrote: > > > > Well guys this is truly bizarre. I managed to get a DL360 G3 to show > > > > interrupts with FC2 but not FC3. Exact same config and setup > > > > proceedure. Ofcourse neither FC2 or FC3 show interrupts with the DL360 > > > > G4. I think TE410P is just a flakey card. > > > > Anyone got a DL360 G3 going with a TE410P and FC3? > > > > > > I did manage to get a TE110P running on the DL380 G4. Still can't get the > > > TE410P working in the G4 though. Supports your theory. > > > > > > Sadly we're now being forced to look elsewhere for PRI cards. > > > > > > -- > > > Regards, > > > Tais M. Hansen > > > ComX Networks A/S > > > Tel: +45-70257474 > > > Fax: +45-70257374 > > > > > > > > > > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 Bandwidth Study
Can someone verify that these figures are correct? http://www.voip-info.org/wiki-Asterisk+bandwidth+iax2 According to this you should be able to route 11 G.729 calls using an IAX2 trunk over a 128Kbps line. It seems high, what sort of figures are some of you getting? Alen ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HP ProLiant server for Asterisk
I have the new card on order via an RMA for my old one. I'll let you know. Mark Peter Childs wrote: Digium support are trailing some new firmware with the TE410P for machines with the Intel E75xx Chipsets that are having issues (such as the DL380 G4). I believe they are confident they have resolved the issue that prevents the cards working, but you may need to specifically mention that you are running this type of machine when acquiring the cards to ensure you get the 'in-testing' firmware. I'm sure as soon as someone gets and tests the new firmware on one of these machines they will post their results to the list. Regards, Peter -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Dana Olson Sent: Saturday, 5 February 2005 1:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] HP ProLiant server for Asterisk I'm looking at ordering a server from HP. I checked around on Google and found in the Wiki that the ProLiant DL380 is supposed to be known to work with *. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 Bandwidth Study
> Can someone verify that these figures are correct? > > > > http://www.voip-info.org/wiki-Asterisk+bandwidth+iax2 > > > > According to this you should be able to route 11 G.729 calls using an IAX2 > trunk over a 128Kbps line. It seems high, what sort of figures are > some of you getting? > The 30.0 kbps for a single call is about right. I don't use trunking, so not sure about the remainder of those figures. I would seriously doubt that you could get anywhere near 11 calls out of a 128kb circuit (of any type). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] "whispering" mode in Meetme?
Dear all:
I want to setup a "whispering" (or "coach") meetme room. the goal is
make a channel with the capability of barge in a Agent call with a
customer, but only the operator can ear the voice of the supervisor.
This with the purpose agent monitoring.
Anybody out there can set-up this function?
The only parameters in Meetme app ('m' and 't') don't are usesfully
enough for my purposes.
Thanks in advanced... Fernando Romo
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[Asterisk-Users] Intel 537EP is NOT the MD3200 aka X100P [Re: Intel 537EP chipset, revisited]
It seems the information out there on this topic was confusing enough to
cause me to waste a lot of time, and that this topic is rehashed
periodically. So now it's my turn
I had a modem with an Intel 537EP chip on it and thought it might work
with the zaptel drivers. Initially, it seemed the only issue would be
the PCI device & vendor codes. Then, I ran into the problem that the
serial driver claimed the device, disallowing the zaptel wcfxo driver to
grab it. The following patches fix those problems, BUT, further
investigation reveals this card is not compatible. It's not clear how
different the 537EP is from teh MD3200 / X100P; might be worth checking
into.
http://linmodems.technion.ac.il/packages/Intel/537/ is Intel's site
containing linux modem drivers for the 537 line, and these appear to
include voice features. Not sure if duplex and if compatible with
Asterisk voicemodem driver. Again, might be worth checking into.
In the meantime, make sure you purchase the X100P or MD3200 !
*** wcfxo.c 2005-01-15 17:59:18.0 -0500
--- wcfxo.c.new 2005-02-06 22:47:06.343556005 -0500
***
*** 980,985
--- 980,986
{ 0xe159, 0x0001, 0x8086, PCI_ANY_ID, 0, 0, (unsigned long)
&generic },
{ 0xe159, 0x0001, 0x8087, PCI_ANY_ID, 0, 0, (unsigned long)
&generic },
{ 0x1057, 0x5608, PCI_ANY_ID, PCI_ANY_ID, 0, 0, (unsigned long)
&wcx100p },
+ { 0x8086, 0x1080, PCI_ANY_ID, PCI_ANY_ID, 0, 0, (unsigned long)
&generic },
{ 0 }
};
***
*** 1000,1005
--- 1001,1048
{
int res;
int x;
+
+
+ struct pci_dev *pdev;
+ int i;
+
+ /* workaround for serial driver falsely claiming to handle our
devices. */
+ for(i=0; wcfxo_pci_tbl[i].vendor; i++) {
+ pdev = NULL;
+ while( (pdev = pci_find_device(wcfxo_pci_tbl[i].vendor,
wcfxo_pci_tbl[i].device, pdev)) ) {
+ if pdev->class >> 8) != PCI_CLASS_COMMUNICATION_SERIAL) &&
+ ((pdev->class >> 8) !=
PCI_CLASS_COMMUNICATION_MODEM)) ||
+ (pdev->class & 0xff) > 6)
+ continue;
+
+ if(pdev->driver) {
+ if(pdev->driver->name && !strcmp(pdev->driver->name,
"serial")) {
+ printk(KERN_WARNING"wcfxo: %s driver grabbed our
device (%s), reclaiming it..\n", pdev->driver->name, pdev->slot_name);
+ #if ( LINUX_VERSION_CODE >= KERNEL_VERSION(2,6,0) )
+ {
+ struct device *dev = get_device(&pdev->dev);
+
+ if(dev) {
+ device_release_driver(dev);
+ put_device(dev);
+ }
+ }
+ #else
+ if(pdev->driver) {
+ if(pdev->driver->remove) {
+ pdev->driver->remove(pdev);
+ }
+ pdev->driver = NULL;
+ }
+ #endif
+
+ }
+ }
+ }
+ }
+
+
+
if ((opermode >= sizeof(fxo_modes) / sizeof(fxo_modes[0])) ||
(opermode < 0)) {
printk("Invalid/unknown operating mode specified.
Please choose one of:\n");
for (x=0;x
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Re: [Asterisk-Users] iax2-jitter-trunking?
There is a new patch in the mantis for jitter buffering together with
trunking.
On 6 Feb 2005 at 18:45, Mark Eissler wrote:
> AFAIK, trunk=yes is not a global option. You set it within a context.
> Also, using the jitter buffer with trunk=yes is not recommended since
> its broken right now.
>
> -mark
>
> On Feb 6, 2005, at 12:45 PM, Rich Adamson wrote:
>
> >
> > Two cvs-head asterisk boxes with iax2 working fine (without register
> > statements).
> >
> > When two calls are placed simultanously from system A -> B and the
> > packets
> > are sniffed on the wire, I see the two calls using two different udp
> > packets. At the top of iax.conf I have trunk=yes and jitterbuffer=yes
> > (at both ends).
> >
> > I was expecting to see both calls handled within a single udp packet,
> > but that's not happening. Each iax2 packet is 79 bytes using ethereal.
> >
> > I've tried the trunk=yes both within the inbound context and at the top
> > of the iax.conf file (assuming the one at the top would be used for all
> > outbound iax calls that don't reference a context). Calls are placed
> > with:
> > exten => _2.,1,Dial(IAX2/user:[EMAIL PROTECTED]/${EXTEN:1})
> >
> > Is trunking dependent upon the use of 'register'? Or, dependent on the
> > above exten=>_2., referencing a context (instead of the IP directly)?
> --
> Mark Eissler, [EMAIL PROTECTED]
> Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com
>
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[Asterisk-Users] Which version of asterisk-oh323 should i use with asterisk v1-0-5.
Hi list, I have successfully upgrade my Asterisk V1-0-RC2 to V1-0-5, but I have a problem. The Asterisk box crashes now every time. I’m using asterisk-oh323. is there a stable version of asterisk-oh323 that can work with the v1-0-5 of Asterisk. Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re: difference between STUN servers and far-end solutions
Hi asterisk list, this is a bit off topic, but can anyone explain the point of the commercial far-end solutions floating around (jasomi, for example)? or are the far-end things just hyped up media proxies? They claim to be b2bua devices but that's a very wide category and only implies that the media stream passes through it - exactly what can be done with fairly simple OSS stuff. In short, what advantage does such a setup have over, for example, an all-IAX setup, or STUN, or a setup with SER/mediaproxy as a SIP server and asterisk behind it? thanks, yair ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] wanted: sample config' using GOTOIF's for all features for a roll-out
Bon soir, * People, You guys are fantastic. This list is hyper-helpful; thanks for that. I have an organization that needs to roll-out * for each of its 30+ locations across the country & we want all features in a DB where they can be easily toggled on & off, as needed. I have done some research in the usual places & got a bit of it, but not all of it, & am hoping someone else out there might be generous enough to share with me (on or off list) a skeleton type config' that would achieve this. I have a few notes to add: + that the internal DB can handle up to approx 10 calls per second, beyond which I might start investigating using an external DB. I dont think we'll have that level of calling. + that a DB via ODBC might not be required until we have serious strain on the DB + need sample DB tables & fields + sample skeleton config' (so we dont flood the list, either offlist or www.pastebin.ca) + that AGI might be overly-complicated + that it might make sense to use GOTOIF's everywhere possible in the dialplan to be able to flexibly enable features for each caller, each destination, each client, each + that there are approx' 20 main features (ie: call waiting, moh, intl dialing, local dialing, n-digit dialing, voicemail, forwarding, etc, etc, etc) any client might want. (I havent really counted. Any one know of a good list of core features that a given client might want.) + does any one have a list or URL that lists the core features that a given client might want in a typical intl business type setting. I count 65 features on asterisk.org features page. Any and all replies & pointers & sample config's & URL's are so very much appreciated. I hope I can return the favor to the list soon. Spread peace. Bobby Do you Yahoo!? Yahoo! Search presents - Jib Jab's 'Second Term'___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7902 Phone
Hi Has anyone got the 7902 phone work with asterisk , the only thing I was able to do with it, is to dial from it.. It doesn't ring ,and if you pick the handset for 30 secs , asterisk crashes. I know cisco is not planning on releasing a SIP image for it , so we are stuck with SCCP. Regards -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Trevor Peirce Sent: Monday, February 07, 2005 1:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Encrypted VOIP? Wolfgang S. Rupprecht wrote: >In theory, the Sipura line supports SRTP. I've got both a spa-841 and >a spa-3000 that have config areas for loading the srtp rsa keys. >Unfortunately there isn't enough information given by sipura as to how >to generate these rsa keys. (eg. can one use an openssl generated >key?) > > In the Sipura support area (authentication required), there is a tool to generate Mini Certificates for this. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 Trunk Problems with NAT
Hi, I have successfully configured an IAX trunk between 2 asterisks, calls can go through both ways without any problems, NAT in the middle of course (iptables) Now , leave them for a while , and make a call from the external server , it doesn’t go through, Dial from the internal one, everything works fine again.. Now , it is clearly a problem in the NAT engine, although IAX shouldn’t have this problem is there something I can do to prevent asterisk from tearing down the trunk while it is not used or , modify the iptables so that to allow the incoming UDP connections Cheers h ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P FXS works only if two lines are off hook?
I have a TDM400P with one FXO module and two FXS modules in it. I also have a Wildcard X101P. After trying hard to get things working on various Intel computers, but having echo problems that made it not really usable, I decided to try it on some older PowerPC (Macintosh) hardware running Yellow Dog Linux. Things started off smoothly. Both zaptel and asterisk seemed to compile okay, and both cards are detected: kernel: Found a Wildcard FXO: Wildcard X101P kernel: PCI: Enabling device 00:0f.0 (0004 -> 0007) kernel: Freshmaker version: 63 kernel: Freshmaker passed register test kernel: Module 0: Installed -- AUTO FXO (FCC mode) kernel: Module 1: Installed -- AUTO FXS/DPO kernel: Module 2: Installed -- AUTO FXS/DPO kernel: Module 3: Not installed I can dial out from a SIP phone through the FXO ports on the X101P or the TDM400P (with almost no echo), so some things are basically working. However, the FXS lines didn't work properly: there is no audio in either direction. If I call these channels from a SIP phone, they do ring properly, and if I pick up a phone connected to them, the console correctly shows, for example: -- Starting simple switch on 'Zap/3-1' But there is no dialtone and no voice audible in either direction when they are called. Outgoing calls from these FXS channels don't work; pressing numbers on the keypad beeps but has no other effect. Then by accident I picked up both FXS lines at the same time, and both of them work perfectly! I get dialtones, I can dial and make calls with them, audio works in both directions -- nothing wrong at all. So as long as they're both off the hook at the same time, everything is fine. But as soon as I hang up either one of the lines, the sound on the other line will *also* go dead again within a second. A little more experimentation: having just one FXS module on the TDM400P (removing the other) doesn't work at all. The only way the FXS lines work is with both FXS modules installed and off hook simultaneously. This problem occurs with the released versions (zaptel 1.0.4 with the wcfxs driver and asterisk 1.0.5), and with cvs head (using the wctdm driver). Does anyone have any idea why this would happen, and how I could fix it? -- Robert L Mathews, Tiger Technologies http://www.tigertech.net/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
