[Asterisk-Users] Pound to Hangup an ongoing call
Hi Folks, Is it possible to setup some parameter on Dial command to hangup a call if the customer press # ? Thanks, Isamar ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Configuring Meeting Room from Asterisk Manager API
Somesh, On 2/3/06, Somesh S Shanbhag [EMAIL PROTECTED] wrote: I want to do a three-party conferencing using manager api. But I found out from the asterisk-users list that I *MUST* use the meeting room concept. I wanted to know wheather meeting room can be configured dynamically? on the fly? Otherwise, configuring meeting room statically is not scalable. First search for 'dynamic conferences' on voip-info.org. There you'll find macro to create dynamic conferences on the fly. Main idea is to enable dynamic creation of meetme rooms and create them according to user phone number. See also Originate command in manager actions reference. You may use command similar to this: Action: Originate Channel: SIP/4 Application: MeetMe Data: 41|adEpq ActionID: MeetMe-id CallerID: MeetMe-caller-id Use 'Channel' to specify user you want to add, and you may use 'CallerID' to track following events. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Directed Call Pickup
Remember that the *8 in your features.conf has nothing to do with direct pickup. So in your case try replacing _86. with _*8. but I don't know if that will cause problems. Yes!!! I thought that this was a feature too instead it's a dialplan application. Asterisk is a bottomless sea. Thanks Mimmus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE : [Asterisk-Users] OT O'Reilly Asterisk TFOT
Have you seen that 3 Asterisk servers were running during this show ? François, I was there (had a coffee with Dave in fact) but was wondering, there was no official asterisk presence, was there? Maybe we should have helped organize this as * is a Linux Solution ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Directed Call Pickup
Ok, I think I am getting somewhere. When I am ringing extension 200 I
do a show channel SIP/200 and this is what I get:
-- General --
Name: SIP/200-b699
Type: SIP
UniqueID: asterisk-2177-1138957721.175
Caller ID: s
Caller ID Name: (N/A)
DNID Digits: (N/A)
State: Ringing (5)
Rings: 0
NativeFormat: 8
WriteFormat: 8
ReadFormat: 8
1st File Descriptor: 28
Frames in: 1
Frames out: 354
Time to Hangup: 0
Elapsed Time: N/A
Direct Bridge: none
Indirect Bridge: none
-- PBX --
Context: internal
Extension:
Priority: 1
Call Group: 2
Pickup Group: 2
Application: AppDial
Data: (Outgoing Line)
Blocking in: ast_waitfor_nandfds
Variables:
DIALEDPEERNUMBER=200
[EMAIL PROTECTED]
There is no value for Extension:. Is this normal? If not, how is it set?
My sip.conf:
[general]
port = 5060
bindaddr = 0.0.0.0
canreinvite=no
disallow=all
allow=alaw
context=internal
[200]
callerid=Reception 200
type=friend
host=dynamic
dtmfmode=rfc2833
username=200
secret=pbx
Kind Regards
Garth
Garth van Sittert wrote:
Show Features produces:
Builtin Feature Default Current
--- --- ---
Pickup*8 *8
Blind Transfer# #
Attended Transfer *2
One Touch Monitor
Disconnect Call * *
Dynamic Feature Default Current
--- --- ---
(none)
Call parking
Parking extension : 700
Parking context : parkedcalls
Parked call extensions: 701-720
in extension.conf I have:
exten = _8.,1,Pickup(${EXTEN:1})
When I dial 812, in the CLI I can see:
Executing Pickup(SIP/29-707f, 12) in new stack
Any thoughts?
Kind Regards
Garth
Bob Goddard wrote:
On Thursday 02 Feb 2006 16:46, Garth van Sittert wrote:
Hi All
I am having problems with Directed Call Pickup in Asterisk 1.2.1
If extension 100 is ringing, a user at another extension is supposed to
be able to dial *8100 and pickup the call to 100. It isn't working for
me and I cannot figure out why.
I have in features.conf:
pickupexten = *8
At the CLI, show features should tell you if it is configured.
If so, you need to tell us what happens on the console.
If not, then you are liable to get asked my car does not work,
does anyone know why?.
B
--
Garth van Sittert
BSc (Physics Computer Science)
-
Mobile: +27 (0)83 791 6662
Email: [EMAIL PROTECTED]
Phone: 08600 BITCO
Web:www.bitco.co.za
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Re: [Asterisk-Users] Directed Call Pickup
Hi Alex
I tried your exact example below and still the same thing. I am getting
403 Denied after I see the Pickup cmd in the CLI. If you do a show
channel SIP/XXX when the phone is ringing, do you get a value for
Extension:??
Kind Regards
Garth
Alex Barnes wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Garth van Sittert
Sent: 02 February 2006 16:47
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Directed Call Pickup
Hi All
I am having problems with Directed Call Pickup in Asterisk 1.2.1
If extension 100 is ringing, a user at another extension is supposed
to
be able to dial *8100 and pickup the call to 100. It isn't working
for
me and I cannot figure out why.
I have in features.conf:
pickupexten = *8
I am running 1.2.1 and works for me.
exten = _86.,1,Macro(directedPickup) ; Direct Pickup
[macro-directedPickup]
exten = s,1,Pickup(${MACRO_EXTEN:2});
Remember that the *8 in your features.conf has nothing to do with direct
pickup. So in your case try replacing _86. with _*8. but I don't know
if that will cause problems.
HTH
Alex
---
Alex Barnes
Engineering Support
Ubiquity Software
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--
Garth van Sittert
BSc (Physics Computer Science)
-
Mobile: +27 (0)83 791 6662
Email: [EMAIL PROTECTED]
Phone: 08600 BITCO
Web:www.bitco.co.za
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[Asterisk-Users] international calling via POTS in Russia
Hi,
I'm having a problem calling international numbers with debian's
Asterisk 1.0.7 w/ zaptel 1.0.9 in Moscow. Russia doesn't seem to have
touchtone dialing, so pulsedial is enabled on my TDM400P interface.
Local numbers work fine, but when it comes to long distance or
international, I'm lost.
The prefix for these should be 8 (wait for dialtone) 10 (country code)
(city code) (phone number). I've tried with 8w10, 8p10 and even
Dial(Zap/g1/8||D(10${PHONENUM})), but nothing seems to work, I get
either dead air (first 2 methods) or a plain dialtone (for the last).
The Asterisk console shows that exactly the desired number has been
dialed.
Any help would be much appreciated. Thanks for reading this mail.
P.S. Sorry if this turns out to be a double post, my provider's smtp server has
sometimes serveral days' delays.
--
Regards,
Balint Kovacs
System Administrator
AES Cargo - MoveOne Relocations
-
This mail was sent through IMP: http://horde.org/imp/
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[Asterisk-Users] CallerID popup
Hi, I'm trying to write a small Visual Basic app to throw a popup with CallerIDNum when a call center agent answers a queue call. Does anyone know what is the right manager event to intercept? Thanks Mimmus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallerID popup
Link event2006/2/3, Mimmus [EMAIL PROTECTED]: Hi,I'm trying to write a small Visual Basic app to throw a popup withCallerIDNum when a call center agent answers a queue call.Does anyone know what is the right manager event to intercept?ThanksMimmus ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Giovanni Miano ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CallerID popup
It works. Thanks a lot. With 15/20 users, is it better to use a manager proxy or to connect directly to the Asterisk server? Thanks From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giovanni MianoSent: Friday, February 03, 2006 11:42 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] CallerID popup Link event ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Receiving faxes with spandsp - strange problem
Bartosz Jozwiak wrote: Check if rxfax actually receives anything... How? -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallerID popup
Dial event, in Asterisk 1.2:203-201Event: Dial Privilege: call,all Source: SIP/203-8467 Destination: SIP/201-45d9 CallerID: 203 CallerIDName: 203 SrcUniqueID: asterisk-1912-1138197095.3769 DestUniqueID: asterisk-1912-1138197095.3771Mimmus [EMAIL PROTECTED] ha scritto: Hi,I'm trying to write a small Visual Basic app to throw a popup withCallerIDNum when a call center agent answers a queue call.Does anyone know what is the right manager event to intercept?ThanksMimmus___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Mail: gratis 1GB per i messaggi, antispam, antivirus, POP3___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: delaying answer for a number of ringsor an amount of time
Right. My original question was about making Asterisk wait a number or rings (or amount of time) before picking up a Zap line. If the rings/time were not reached while the line is still ringing, do nothing. As someone must have already said, it's not a good idea to share lines with asterisk. This said, when we first began to use asterisk, I needed a way to have it NOT pickup when we temporarily had phones in parellel on the same 2 lines. When a line was flagged in the database as /pickup1=0 the dialpan did a goto(do-nothing,s,1) [do-nothing] exten = s,1,AbsoluteTimeout(0) exten = s,2,NoOp(doing nothing: call came if with flag set to noanswer) exten = s,3,Wait(70) exten = t,1,Hangup The above code made asterisk *never* pickup, so it must be possible, though not good practice, to do what you want, unless things have changed in more recent versions. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to handle provider UNREACHABLE in the dialplan?
Florian Overkamp wrote: Hi Ronald, Ronald Wiplinger wrote: voipbuster/ 194.221.62.201 5060 UNREACHABLE voipstunt/x 194.120.0.200 5060 a reload shows than: voipbuster/ 80.239.235.200 5060 UNREACHABLE voipstunt/x 194.120.0.200 5060 UNREACHABLE Seems like voipbuster is doing round-robin DNS for redundancy. Bad choice with asterisk, since asterisk only looks up DNS on startup or reloads. You could read out all the entries in the DNS zone and create your own list of entries in /etc/hosts, and then create multiple asterisk peers: voipbuster1, voipbuster2, etc... Then you can use regular dialplan logic to cycle through all of them. Florian, that is exactly the point what I am looking for. How can I use the next peer in the dial logic? I was trying DIALSTATUS, ... but I could not make it. bye Ronald ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Receiving faxes with spandsp - strange problem
Pierre Burton wrote: What's your cisco conf ? how did you transfert between Cisco and asterisk ? A-law, U-law ?? This is part of my Cisco config: voice-card 0 no dspfarm ! ! ! voice service voip sip ! ! voice class codec 1 codec preference 1 g711ulaw codec preference 2 g711alaw codec preference 3 g729r8 bytes 40 codec preference 4 g723r63 bytes 96 codec preference 5 g726r16 bytes 80 codec preference 6 g726r24 codec preference 7 g726r32 codec preference 8 g728 codec preference 9 gsmefr codec preference 10 gsmfr voice vad-time 65536 ! voice translation-rule 1 rule 1 /^0?/ // ! voice translation-rule 2 rule 2 /^1?2?/ // voice translation-profile CutTwelve translate called 2 ! voice translation-profile CutZero translate calling 3 translate called 1 voice-port 0/1/0:15 echo-cancel coverage 32 no comfort-noise music-threshold -70 dial-peer voice 1 pots translation-profile outgoing CutZero destination-pattern ^0 direct-inward-dial port 0/1/0:15 ! dial-peer voice 2 voip description Route calls starting with 293 to centile translation-profile outgoing CutTwelve application session destination-pattern 1229339[60-79] voice-class codec 1 session protocol sipv2 session target ipv4:62.111.174.79 dtmf-relay rtp-nte h245-signal h245-alphanumeric ! sip-ua retry invite 3 retry response 3 retry bye 3 retry cancel 3 timers trying 1000 sip-server ipv4:asterisk.ip.add.ress As I understand, the preferred codec is ulaw. Should I change something in this configuration? -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400 and Phone does not 'ring'
There is an issue here in France with our Siemens DECT phones that
required a patch to change the ring _frequency_. It was given here
ages ago, but now I can't find it.
Shame on me for not coming back!
//{20,7,RING_OSC,0x7EF0}, // changed to
{20,7,RING_OSC,0x7E6C}, // new value for 25hz
for all our euro phones
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RE: [Asterisk-Users] Re: Web interface
I have been playing with the Signate switch. Official training starts soon but just playing with it leaves me with the impression that it is powerful but very complex. You need to RTFM to get anything working. They have also used IonCube to encode all PHP and HTML files so customization is impossible without reverse engineering :-( I will reserve final judgement until I go through the official Signate training, but for several thousand dollars, I could have wrote a GUI that accesses and writes to the realtime database with a much more intuitive interface. Another question, If Signate is not using ABE, what are their requirements for releasing source as far as the GUI? Thanks Steve -Original Message- From: Shidan [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 31, 2006 11:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: Web interface Yes DRUID is a much better interface than AMP, and I would never buy it because its not that good. Destar is something I have also discovered recently and even though its still in early development it has the best design and its plug in architecture makes it potentially quite powerful. it's very easy to customize your pbx interface with Destar, and that can translate to $$$ if you are selling pbx's or if you are a service provider, adding an interface for functionalities specific to your offerings. Hopefully by this time next year more people will be talking of it. --- Shidan Gouran On 1/31/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Tue, Jan 31, 2006 at 11:24:54AM +0100, Vikram Rangnekar wrote: +++ Strain Jer [30/01/06 01:29 +]: I was searching thru the internet and I found a wide variety of different web interfaces for asterisks I was curious which one is best suited for asterisks. Thanks Check out www.voiceroute.net DRUID is much better than AMP or any of the other interfaces out there. Also its under active development so expect a lot from it. And unlike AMP, it is non-free. BTW: there is also DeStar: http://destar.berlios.de/ . Version 0.1.1 was recently released. Nice and clean. Generally runs its own daemon, though can run under apache. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.375 / Virus Database: 267.15.0/248 - Release Date: 2/1/2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.375 / Virus Database: 267.15.0/248 - Release Date: 2/1/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Receiving faxes with spandsp - strange problem
Bartosz Piec wrote: Hello, I'm trying to receive faxes with asterisk. My configuration is like this: Codec? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Receiving faxes with spandsp - strange problem
Matt Riddell (IT) wrote: I'm trying to receive faxes with asterisk. My configuration is like this: Codec? In Asterisk or in Cisco? -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cmd set with multiple values
hello!
has this made it into 1.2.3 already:
http://bugs.digium.com/view.php?id=6128 ?
i'm trying to set a variable that should be used as a dialstring in the
dial-command, including parameters seperated with the respective
delimiter, e.g. like:
exten = 907,1,Set(DESTINATION1=Zap/G1/4989123456789|10|gh)
exten = 907,n,Set(DIALSTRING=${DESTINATION1})
exten = 907,n,Dial(${DIALSTRING})
asterisk complains:
Feb 3 12:39:40 WARNING[26200]: pbx.c:6010 pbx_builtin_setvar: Ignoring
entry '10' with no = (and not last 'options' entry)
i've tried several of the resolution-proposals mentioned in the
bugnotices, but none of them seems to work yet.
the best fit was exten =
907,1,Set(DESTINATION1='Zap/G1/4989123456789,10,gh') but then the value
included in the quotes seems to be set as a string that is not parsed when
dialing ${DIALSTRING}, resulting in
Called G1/4989123456789,10,gh
is there any workaround?
thanks
christian
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[Asterisk-Users] How can I configure to call from the console by means of a sip phone,
How can I configure to call from the console by means of a sip phone, any docs on this. Regards, Anthony. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Musiconhold in zapata.conf
Title: Musiconhold in zapata.conf I've been trying to change the musiconhold= in the zapata.conf to use something other than default. However it doesn't seem to do it. I know the other musiconhold source works but whatever I set it to in the zapata.conf file it always plays whatever is [default] in musiconhold.conf. Also random=yes doesn't work. [default] mode=files directory=/var/lib/asterisk/mohmp3 random=yes [livestream1] mode=custom application=/usr/local/bin/mpg123 -s --mono -y -f 8192 -r 8000 -@ /etc/asterisk/playlist Lee ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] international calling via POTS in Russia
Hello,
There're few POTS supporting touchtone, others - just pulse. In
Russia you need to dial 8, wait for tone and only then continue
dialing 10 (for intl. plan), country code, area code and number.
bkmc Hi,
bkmc I'm having a problem calling international numbers with debian's
bkmc Asterisk 1.0.7 w/ zaptel 1.0.9 in Moscow. Russia doesn't seem to have
bkmc touchtone dialing, so pulsedial is enabled on my TDM400P interface.
bkmc Local numbers work fine, but when it comes to long distance or
bkmc international, I'm lost.
bkmc The prefix for these should be 8 (wait for dialtone) 10 (country code)
bkmc (city code) (phone number). I've tried with 8w10, 8p10 and even
bkmc Dial(Zap/g1/8||D(10${PHONENUM})), but nothing seems to work, I get
bkmc either dead air (first 2 methods) or a plain dialtone (for the last).
bkmc The Asterisk console shows that exactly the desired number has been
bkmc dialed.
bkmc Any help would be much appreciated. Thanks for reading this mail.
bkmc P.S. Sorry if this turns out to be a double post, my provider's smtp
server has
bkmc sometimes serveral days' delays.
bkmc --
bkmc Regards,
bkmc Balint Kovacs
bkmc System Administrator
bkmc AES Cargo - MoveOne Relocations
bkmc -
bkmc This mail was sent through IMP: http://horde.org/imp/
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--
Grigoriy Puzankin
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Re: RE : [Asterisk-Users] OT O'Reilly Asterisk TFOT
On Fri, 2006-02-03 at 09:52 +0100, Wilson Pickett wrote: Have you seen that 3 Asterisk servers were running during this show ? François, I was there (had a coffee with Dave in fact) but was wondering, there was no official asterisk presence, was there? Maybe we should have helped organize this as * is a Linux Solution Good idea, and we've got 362 days to organise it. I'd be ready to do it. It could be in the village or even a proper stand, what do the rest of the French users think? -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Web interface
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 The GPL primarily deals with linking to the libraries of a GPL project. I am not aware of any changes they made directly to asterisk, php, mysql etc that would bind them to the GPL. However, if they are using/requiring mysql, then they may have to have purchase the mysql commercial license as mysql is not available under the GPL for any closed source application that requires mysql. ( IIRC ) Steve Totaro wrote: I have been playing with the Signate switch. Official training starts soon but just playing with it leaves me with the impression that it is powerful but very complex. You need to RTFM to get anything working. They have also used IonCube to encode all PHP and HTML files so customization is impossible without reverse engineering :-( I will reserve final judgement until I go through the official Signate training, but for several thousand dollars, I could have wrote a GUI that accesses and writes to the realtime database with a much more intuitive interface. Another question, If Signate is not using ABE, what are their requirements for releasing source as far as the GUI? Thanks Steve -Original Message- From: Shidan [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 31, 2006 11:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: Web interface Yes DRUID is a much better interface than AMP, and I would never buy it because its not that good. Destar is something I have also discovered recently and even though its still in early development it has the best design and its plug in architecture makes it potentially quite powerful. it's very easy to customize your pbx interface with Destar, and that can translate to $$$ if you are selling pbx's or if you are a service provider, adding an interface for functionalities specific to your offerings. Hopefully by this time next year more people will be talking of it. --- Shidan Gouran On 1/31/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Tue, Jan 31, 2006 at 11:24:54AM +0100, Vikram Rangnekar wrote: +++ Strain Jer [30/01/06 01:29 +]: I was searching thru the internet and I found a wide variety of different web interfaces for asterisks I was curious which one is best suited for asterisks. Thanks Check out www.voiceroute.net DRUID is much better than AMP or any of the other interfaces out there. Also its under active development so expect a lot from it. And unlike AMP, it is non-free. BTW: there is also DeStar: http://destar.berlios.de/ . Version 0.1.1 was recently released. Nice and clean. Generally runs its own daemon, though can run under apache. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.375 / Virus Database: 267.15.0/248 - Release Date: 2/1/2006 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFD41O6y9wPyZpnL2URAsq+AKCWGLq14yeHhV2vET+twWFyBtHsXACdF1fb VtFJLAKfqb7+ch2e7lbKrzQ= =cQx7 -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] inform the agent about the queue he is answering
i'm planning to migrate a call center to asterisk, i don't understand if i can launch a resident application on the agent's client in relation with the queue the agent's is answering. For example: I have - queue A - queue B - queue C Agent 100 (logged in A.B,C) Agent 101 (logged in C) When Agent 100 receives a call from the queue A i'd like to launch his browser and point it to http://myserver/clientA, when the agent receives a call from the queue B i'd link to launch his browser and point it to http://myserver/clientB Is it possible? With what soft-phone? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] hardware and network requirements
Hi i'm planning to migrate a callcenter to asterisk and VOIP, the call center can have up to 25 cuncurrents agents logged in. I'll have some simplty IVR business logic and the some queues. Can a normal server with 1 GB ram 100 GB HDD Pentium 4 3.6 Ghz CPU Ethernet 10/100/1000 Support this? Would you suggest me a particular products? The server and the agents will be in the same LAN, is enought a 100 Mbit LAN or shall i use a Gbit switch, Gbit LAN interface on server, 100Mbit on agents pc? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CallerID popup
I have been planning to do the same thing but never got around to it, I actually did write a nice class to wrap the interface to the manager but it isn't complete. Would you be willing to share your work? -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mimmus Sent: Friday, February 03, 2006 5:19 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] CallerID popup Hi, I'm trying to write a small Visual Basic app to throw a popup with CallerIDNum when a call center agent answers a queue call. Does anyone know what is the right manager event to intercept? Thanks Mimmus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Using *RT for HA purposes was: [Asterisk-Users]Realtime MultipleAsterisk boxes, iaxusers
Hi, ALL: Can anyone tell me what *RT is ? What is its full name? I think the * is asterisk but what is RT ? 2006/2/2, Rusty Shackleford [EMAIL PROTECTED]: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alistair Cunningham Sent: Wednesday, January 04, 2006 4:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: Using *RT for HA purposes was: [Asterisk-Users]Realtime MultipleAsterisk boxes, iaxusers load balacing isn't perfect, and it can give uneven loads at low capacity, but it gets better as load increases which is where it matters. What kind of loads are we talking about here, please? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Charles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Serusers] high-availibility setup using f5 bigip
I think that the range of this question is too large. You should tell us what your scenario is. And tell us more about your configurations. 2006/2/2, Jack Wei [EMAIL PROTECTED]: hi, I'm trying to set up 2 SER and 2 Asterisks boxes using a bigip switch to do load-balancing. I'm using Asterisk as a voicemail application only and have successfully integrated SER with Asterisk without the switch. But when I try to use the switch as a load-balancer, I get lots of NAT problems. Does anyone know how to setup the switch and SER/Asterisk properly? Thanks, Jack __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Serusers mailing list [EMAIL PROTECTED] http://mail.iptel.org/mailman/listinfo/serusers -- Best Regards Charles -- Best Regards Charles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT O'Reilly Asterisk TFOT
Le Vendredi 3 Février 2006 13:54, Dave Cotton a écrit : On Fri, 2006-02-03 at 09:52 +0100, Wilson Pickett wrote: Have you seen that 3 Asterisk servers were running during this show ? François, I was there (had a coffee with Dave in fact) but was wondering, there was no official asterisk presence, was there? Maybe we should have helped organize this as * is a Linux Solution Good idea, and we've got 362 days to organise it. I'd be ready to do it. It could be in the village or even a proper stand, what do the rest of the French users think? Hello, It's A Good Idea . We have allready made some Asterisk Presentation on OpenSource Day In Alsace It Fun to Discuss Open Source , Linux and Asterisk. fabrice ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] (newby) IAX Trunk on low bandwidth connection
My remote 64kbit connection would only be used for VoIP and NOTHING else! No email, no browsing. Besides, my remote 64kbit guaranteed-bandwidth connection changed into a 256/512 ADSL connection from the same telco provider (that's actualy wy cheeper then the other 64kbit guaranteed connection) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jolly M. Recto Sent: Friday, February 03, 2006 5:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] (newby) IAX Trunk on low bandwidth connection hi, I am just like yours a Satellite HUB operator that providing a voip. Before i am providing a ptp h323 with g723 codec boxes ranging 2 to 4 port at 64kbps upstream and shared 2mbps downstream and now i come up to put asterisk using a g723 and g729 of digium but not work for me because when the remote end used data like email, browsing the voice suffer the quality. I look back on my last experiment and i see that SER and OPENSER is much better solutions to provide just a voice and voicemail to call out. The 64kbps with Qos in the remote config will help a lot better to provide a 99 % satisfaction to the customer. //jollyr Cosmin Prund wrote: At my HQ I'm instaling a 128kbit leased line connection, with guaranteed bandwidth to the Internet; The telco promises less then 20 ms to the internet (to ronix.ro), no jitter and no packet loss. So I'm hoping for 40 ms times to net and small jitter J This is my hub. For my satelite instalations I'm planning on grabing a connection from a different provider (as this telco provider is expensive) but I'm also considering a 64kbit leased line from the same provider, just in case my VoIP doesn't work with the cheeper providers. My remote instalations will never have more then one conversation load, and this conversation would be ZAP to IAX or SIP. That is, the distant instalation will need to forward all calls coming in on the zap chanel to my HQ Asterisk. That's all it will ever do J. I'm not sure trunking woud provide anything in this case as there will never be more then one concurent conversation from the remote * to my HQ *. I'm expecting IAX to provide better performance over SIP but not by much. Considering my remote * instalations will never have more then one concurent conversation with my HQ and considering I can get a really good 64kbit line I guess I'm OK. As for my HQ, I'm sure I'm OK because I'll get a 128 kbit line and I'll be able to afford an upgrade to 256kbit. I can actually go all the way to 2048 kbit, but that would no longer be economically viable. So I'll see how it goes, and I hope I'll have the time to put in a comment on the low bandwidth wiki on voip-info.org. Thanks to everyone for your help. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *tim panton *Sent:* Thursday, February 02, 2006 11:05 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [Asterisk-Users] (newby) IAX Trunk on low bandwidth connection On 2 Feb 2006, at 08:09, Cosmin Prund wrote: Brrghhh: Bandwidth calculation is really foggy for me: Using the calculator I'm getting about 23 kbps for both incoming and outgoing. What does this mean: Is a 64kbit link used at 71% capacity ((23+23):64) or is it used at only 35% (23:64)? Will this vary over time (i.e: does the codec generate more then average data at times? How about less then average?) It depends on what sort of link you have. Most links are full duplex (leased lines etc) which would be 35% but some radio based links are half duplex which would be 71% So for a 64k link you will (just about) get 3 729 calls. If all the calls between are between the same two servers, you can use IAX trunking, which would push you up to 5 calls. (What that tells you is that for 729 and gsm, the headers are as big as the data). You talk about satellite stations, if you are going for a hub and spoke, you should put the hub on the highest bandwidth link. Thanks. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Rob Lith *Sent:* Wednesday, February 01, 2006 11:40 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [Asterisk-Users] (newby) IAX Trunk on low bandwidth connection What codec is that using. G.729 will give you 10 calls at best over 256k unless you're trunking with IAX2? I don't know anyone using lpc10... Remember a G.729 8k codec turns into 23.63 Kbps with all the overheads... Regards Rob On 2/1/06, *Garth van Sittert* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi Cosmin You should be able to get
RE: [Asterisk-Users] CallerID popup
I would be happy to share everything but actually I'm working only at a feasibility study. In addition, I'm a system admin and development job is made by someone else! In principle, it's simple: open a socket to manager port, login and wait for right event. Ideal target is a small, traybar application giving chance to login/logoff/pause to the agent (in my opinion, it's better than doing it by phone) and pointing the CRM application to the right caller card. In addition, peraphs, if number of agents is more than a few, it's better to use a manager proxy. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan k. Creasy Sent: Friday, February 03, 2006 2:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] CallerID popup I have been planning to do the same thing but never got around to it, I actually did write a nice class to wrap the interface to the manager but it isn't complete. Would you be willing to share your work? -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mimmus Sent: Friday, February 03, 2006 5:19 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] CallerID popup Hi, I'm trying to write a small Visual Basic app to throw a popup with CallerIDNum when a call center agent answers a queue call. Does anyone know what is the right manager event to intercept? Thanks Mimmus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Waiting x100P and Cisco IP Phone
OK with that being said how can you modify the phone to use the second line button as a speed dial? Then you can label it has flash. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nabeel Jafferali Sent: Thursday, February 02, 2006 11:28 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Call Waiting x100P and Cisco IP Phone How can I send the hook flash to the x100P card to switch to the call coming in from the PSTN? http://www.voip-info.org/wiki-Asterisk+cmd+Flash Scroll down to Re: X100P + Call-Waiting how-to Enjoy. Nabeel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] routing question: multipath routing for SIP
I have been doing multipath between 2 cable modems for over 2 years now. e-mail me off list and I will get you my configs for this. Cheers, /Zac [EMAIL PROTECTED] wrote: Yes, and, you will probably need a different method. Are these t1's to the same provider? Have you considered bonding the channels? Greg *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Script Head *Sent:* Thursday, February 02, 2006 6:32 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [Asterisk-Users] routing question: multipath routing for SIP I have two T1s and I'd like to split my SIP traffic over the two. I am looking at this: http://lartc.org/howto/lartc.rpdb.multiple-links.html what bothers me about it is the note Note that balancing will not be perfect, as it is route based, and routes are cached. This means that routes to often-used sites will always be over the same provider.. If all my traffic goes to the same IP, which is a remote SER proxy, will my second T1 be utilized at all? Does anyone have any experiece with this? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] varion card
Hi, Has anyone used the varion v400p card? what's its performance?? Rgds, _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] callback script?
On Thursday 02 February 2006 11:40, Arne Morten Johansen wrote: How do I setup a Callback script? This script does what I want to do. But how do I set it up? http://www.junghanns.net/en/callback.html I see it uses PHP for scriptlanguage. So where do I place it (the .agi)? /var/lib/asterisk/agi-bin and should be 755 benchev ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] meetme and dtmf
Imran Ahmed wrote:
may or may not work, try at your own risk:
1) Use a sip soft phone and set the dtmf mode = inband.
2) In asterisk set the dtmf mode for that soft phone to be rfc2833 or
info. (this is done so that asterisk ignores the inband dtmf on the
sip channel).
3) Design your dialplan such that asterisk should not depend on dtmf
from the sip call.
ex:
exten xxx, 1, dial(zap/g/client_number) //on answer directed to conference room
exten xxx, 2, dial(zap/g/ivr_number) //on answer directed to conference room.
exten xxx, 3, meetme(conference room)
Thank you very much.
I tried sjphone setting clinet and asterisk as above and it seems to
work. I will test it better in the next hours.
I had a look at meetme.c and i found a portion of code that manage dtmf
if ((f-frametype == AST_FRAME_DTMF) (confflags
CONFFLAG_EXIT_CONTEXT)) {
..
..
-
I think this part manage the case of meetme application is called with
p, X or s option,
but maybe also (i'm not sure, i had not the time to study well enough
the source, and over all i'm not a so good c programmer)
that this part of code prevents asterisk to broadcast the sound to other
channels when it is not inband.
Sorry if my bad english make me not very clear.
Anyway, thank you very much to all for your help.
Accursio Avona
___
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[Asterisk-Users] Events when the target of the call answer
Hi Group, I am sending my question again why I dont have answer yet: I am developing a application, this use Manager API to connect with Asterisk. But when I call to an external number (over a zap channel), I dont receive any event when the target answer, Who can help me?, Which event notify me that the phone call was answered? Thank you. Ezequiel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Queue() with timeout=0
Hi,
do you require more information about this behaviour, I'd be more
than glad to provide it.
thanks,
kr,
Bart
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bart van Daal
Sent: woensdag 1 februari 2006 11:03
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Queue() with timeout=0
On 01/31/06 20:49 Bart van Daal said the following:
exten = 654,1,Answer
exten = 654,2,SetCIDName(${CALLERIDNAME}) exten =
654,3,SetVar(MONITOR_FILENAME=/var/spool/asterisk/monitor/q${EXTEN}-$
{
TIMEST
AMP}-${UNIQUEID})
exten = 654,4,Queue(654|t|||0)
exten = 654,5,Goto(ext-queues,654,1)
what does the variable QUEUESTATUS say when it drops out of the queue ?
thanks for your response
The queuestatus returns: TIMEOUT.
-- Executing Queue(SIP/6900-ee4b, 666|t|||0) in new stack
-- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/7000|60|tr)
in new stack
-- Called 7000
-- Local/[EMAIL PROTECTED],1 is ringing
-- SIP/7000-1b31 is ringing
-- SIP/7000-1b31 is ringing
-- SIP/7000-1b31 is ringing
-- SIP/7000-1b31 is ringing
-- SIP/7000-1b31 is ringing
-- Nobody picked up in 15000 ms
-- Stopped music on hold on SIP/6900-ee4b
-- Executing NoOp(SIP/6900-ee4b, QUEUESTATUS,TIMEOUT) in new stack
this is an exerpt form the CLI logging:
-- Executing Answer(SIP/6900-ee4b, ) in new stack
-- Executing SetCIDName(SIP/6900-ee4b, device) in new stack
-- Executing SetVar(SIP/6900-ee4b,
MONITOR_FILENAME=/var/spool/asterisk/monitor/q666-20060201-105559-113878775
9.45) in new stack
-- Executing Playback(SIP/6900-ee4b, custom/None) in new stack
-- Executing NoOp(SIP/6900-ee4b, before queue|) in new stack
-- Executing Queue(SIP/6900-ee4b, 666|t|||0) in new stack
-- Started music on hold, class 'default', on channel 'SIP/6900-ee4b'
-- Called Local/[EMAIL PROTECTED]
-- Executing Macro(Local/[EMAIL PROTECTED],2,
exten-vm|novm|7000) in new stack
-- Executing Macro(Local/[EMAIL PROTECTED],2, user-callerid)
in new stack
-- Executing DBget(Local/[EMAIL PROTECTED],2,
AMPUSER=DEVICE/6900/user) in new stack
-- DBget: varname=AMPUSER, family=DEVICE, key=6900/user
-- DBget: set variable AMPUSER to 6900
-- Executing DBget(Local/[EMAIL PROTECTED],2,
AMPUSERCIDNAME=AMPUSER/6900/cidname) in new stack
-- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=6900/cidname
-- DBget: set variable AMPUSERCIDNAME to snz
-- Executing GotoIf(Local/[EMAIL PROTECTED],2, 0?5) in new
stack
-- Executing SetCallerID(Local/[EMAIL PROTECTED],2, snz
6900) in new stack
-- Executing NoOp(Local/[EMAIL PROTECTED],2, Using CallerID
snz 6900) in new stack
-- Executing SetVar(Local/[EMAIL PROTECTED],2,
FROMCONTEXT=exten-vm) in new stack
-- Executing Macro(Local/[EMAIL PROTECTED],2,
record-enable|7000|IN) in new stack
-- Executing GotoIf(Local/[EMAIL PROTECTED],2, 0 0?2:4) in
new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI(Local/[EMAIL PROTECTED],2,
recordingcheck|20060201-105600|1138787759.47) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20060201-105600|1138787759.47: Inbound recording not
enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp(Local/[EMAIL PROTECTED],2, No recording
needed) in new stack
-- Executing Macro(Local/[EMAIL PROTECTED],2, dial|60|tr|7000)
in new stack
-- Executing GotoIf(Local/[EMAIL PROTECTED],2, 0?4:2) in new
stack
-- Goto (macro-dial,s,2)
-- Executing GotoIf(Local/[EMAIL PROTECTED],2, 0?5:4) in new
stack
-- Goto (macro-dial,s,4)
-- Executing AGI(Local/[EMAIL PROTECTED],2, dialparties.agi)
in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
-- dialparties.agi: priority = 4
-- dialparties.agi: callingani2 = 0
-- dialparties.agi: accountcode =
-- dialparties.agi: channel = Local/[EMAIL PROTECTED],2
-- dialparties.agi: callerid = 6900
-- dialparties.agi: context = macro-dial
-- dialparties.agi: callington = 0
-- dialparties.agi: dnid = unknown
-- dialparties.agi: request = dialparties.agi
-- dialparties.agi: calleridname = snz
-- dialparties.agi: extension = s
-- dialparties.agi: language = en
-- dialparties.agi: uniqueid = 1138787759.47
-- dialparties.agi: callingpres = 0
-- dialparties.agi: type = Local
-- dialparties.agi: rdnis = unknown
-- dialparties.agi: callingtns = 0
-- dialparties.agi: enhanced = 0.0
dialparties.agi: Caller ID name and number are '6900'
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 7000 to extension map
-- dialparties.agi: Extension 7000 cf is disabled
-- dialparties.agi: Extension 7000 do not disturb is disabled
-- dialparties.agi: Checking CW and CFB status for extension 7000
== Parsing
[Asterisk-Users] Re: delaying answer for a number of rings or an amount
Well in my setup I have a few IP phones connected to Asterisk as well as POTS phones on my analog line. When a call for my daughter comes in on the analog line (determined from callerID) I send it to her own voicemail after 20 seconds of ringing. It all works quite well. Here's a step-by-step of what happens below: 1 - a call comes in and Asterisk rings SIP/Brian and SIP/joe for 30 seconds. 2 - After 30 seconds if the line is still ringing (nobody picked up POTS phone or SIP phones) * answers the line and sends to Voicemail. Asterisk never picks up the call until the 30 seconds are up. [from-pots] exten = s,1,Dial(SIP/brianSIP/joe,30) exten = s,2,Voicemail(u2001) exten = s,3,Hangup Right. My original question was about making Asterisk wait a number or rings (or amount of time) before picking up a Zap line. If the rings/time were not reached while the line is still ringing, do nothing. This allows a handset *on the same POTS line* as Asterisk to pick up and Asterisk does nothing. But if nobody picks up the POTS line (that asterisk is on too) then it picks up. I essentially want Asterisk to be an answering machine on the line. b. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT O'Reilly Asterisk TFOT
Fabrice a écrit : Le Vendredi 3 Février 2006 13:54, Dave Cotton a écrit : On Fri, 2006-02-03 at 09:52 +0100, Wilson Pickett wrote: Have you seen that 3 Asterisk servers were running during this show ? François, I was there (had a coffee with Dave in fact) but was wondering, there was no official asterisk presence, was there? Maybe we should have helped organize this as * is a Linux Solution Good idea, and we've got 362 days to organise it. I'd be ready to do it. It could be in the village or even a proper stand, what do the rest of the French users think? Hello, It's A Good Idea . We have allready made some Asterisk Presentation on OpenSource Day In Alsace We organize in june 23-25 the JL4 (4ème journées du libre) in Strasbourg and we already contact Mark Spencer to invite him. It's under discussion. See http://strasbourg.linuxfr.org/jl4/conferences -- Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Using *RT for HA purposes was: [Asterisk-Users]RealtimeMultipleAsterisk boxes, iaxusers
Realtime.. As in pulling configs from a realtime database.. Or he's trying to link Asterisk to www.bestpracticals.com version of Request Tracker (also known as RT) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Charles Wang Sent: February 3, 2006 8:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: Using *RT for HA purposes was: [Asterisk-Users]RealtimeMultipleAsterisk boxes, iaxusers Hi, ALL: Can anyone tell me what *RT is ? What is its full name? I think the * is asterisk but what is RT ? 2006/2/2, Rusty Shackleford [EMAIL PROTECTED]: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alistair Cunningham Sent: Wednesday, January 04, 2006 4:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: Using *RT for HA purposes was: [Asterisk-Users]Realtime MultipleAsterisk boxes, iaxusers load balacing isn't perfect, and it can give uneven loads at low capacity, but it gets better as load increases which is where it matters. What kind of loads are we talking about here, please? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Charles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] varion card
Akpome Akpoguma wrote: Hi, Has anyone used the varion v400p card? what's its performance?? Rgds, Its the Tormenta 2 card, just like the old T400P and E400P cards from Digium. See www.zapatatelephony.org for details. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallerID popup
You should write a proxy and not connect directly, the reasons are as follows: 1. You don't want asterisk to crash because of problems with the manager app over the network, which Asterisk is known not to handle very well (as per the wiki). 2. Security, if you have every computer connecting to asterisk manager over the network, then you are giving the users a way to login to the system to do much more than they need, with a proxy however, you can always validate (and you should make sure to do that) everything before its submitted to asterisk. On 2/3/06, Mimmus [EMAIL PROTECTED] wrote: It works. Thanks a lot. With 15/20 users, is it better to use a manager proxy or to connect directly to the Asterisk server? Thanks From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giovanni Miano Sent: Friday, February 03, 2006 11:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] CallerID popup Link event ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to handle provider UNREACHABLE in the dialplan?
Hi Ronald,
Ronald Wiplinger wrote:
You could read out all the entries in the DNS zone and create your own
list of entries in /etc/hosts, and then create multiple asterisk
peers: voipbuster1, voipbuster2, etc... Then you can use regular
dialplan logic to cycle through all of them.
that is exactly the point what I am looking for. How can I use the next
peer in the dial logic? I was trying DIALSTATUS, ... but I could not
make it.
Should be easy; we use:
[macro-safedial]
;exten = s,1,Dial(${ARG1},${ARG2},g,${ARG4})
exten = s,1,Dial(${ARG1},${ARG2},${ARG3},${ARG4})
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-CANCEL,1,Hangup
exten = s-NOANSWER,1,GotoIf($[${DIALEDTIME} = 0]?3)
exten = s-NOANSWER,2,Hangup
exten = s-NOANSWER,3,Verbose(1,Need failover for ${ARG1})
exten = s-BUSY,1,Busy
exten = s-CHANUNAVAIL,1,Verbose(1,Need failover for ${ARG1})
exten = s-CONGESTION,1,Congestion
exten = _s-.,1,Congestion
exten = s-,1,Congestion
Florian
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Re: [Asterisk-Users] limit sip sessions
You should create a secret dialing prefix like if you wanted to dial 1555333222 the user would actually have to dial 548261555333222. This way, even if they snatch the username/password but do not know the prefix, they won't be able to dial. On 2/2/06, Miguel [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote:Shouldn't all sip users have different usernames?(or am I missing some vital detail here?)PaulHYes Paul, Im in El Salvador and my users like to share their usernames/passwords and the original owner doesnt like to pay for callshe hasnt made.---Miguel___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Pattern Match - 0 or 1 digit
Does anyone know of a special character used in pattern matching that would match 0 or 1 digit? I would just like to cut down on the number of extensions I have. Current example: exten = 2125551234,1,Dial(SIP/2125551234,15,rt) exten = 12125551234,1,Dial(SIP/2125551234,15,rt) I would like to do something like this (where + matches 0 or 1 digit): exten = _+2125551234,1,Dial(SIP/2125551234,15,rt) | Or in an area that has 7 digit dialing... exten = 5551234,1,Dial(SIP/2125551234,15,rt) exten = 2125551234,1,Dial(SIP/2125551234,15,rt) exten = 12125551234,1,Dial(SIP/2125551234,15,rt) exten = _5551234,1,Dial(SIP/2125551234,15,rt) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] inform the agent about the queue he is answering
Yes, it is possible. You need to track the queue log and channels via manager console or by tailing logs in real time and then match the destination of the caller by the callerid. Then make the decision which URL to redirect the caller too. None of this comes with Asterisk but it is possible to build. On 2/3/06, nik600 [EMAIL PROTECTED] wrote: i'm planning to migrate a call center to asterisk, i don't understandif i can launch a resident application on the agent's client inrelation with the queue the agent's is answering.For example:I have - queue A- queue B- queue CAgent 100 (logged in A.B,C)Agent 101 (logged in C)When Agent 100 receives a call from the queue A i'd like to launch hisbrowser and point it to http://myserver/clientA, when the agentreceives a call from the queue B i'd link to launch his browser andpoint it to http://myserver/clientBIs it possible?With what soft-phone? ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallerID popup
I 'm developing something similar. It a perl script which tells you who is calling but it do it by sendind a jabber message. it's my first perl script so it's not finished yet. i'll share it so you can contribute if you want... 2006/2/3, C F [EMAIL PROTECTED]: You should write a proxy and not connect directly, the reasons are as follows: 1. You don't want asterisk to crash because of problems with the manager app over the network, which Asterisk is known not to handle very well (as per the wiki). 2. Security, if you have every computer connecting to asterisk manager over the network, then you are giving the users a way to login to the system to do much more than they need, with a proxy however, you can always validate (and you should make sure to do that) everything before its submitted to asterisk. On 2/3/06, Mimmus [EMAIL PROTECTED] wrote: It works. Thanks a lot. With 15/20 users, is it better to use a manager proxy or to connect directly to the Asterisk server? Thanks From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giovanni Miano Sent: Friday, February 03, 2006 11:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] CallerID popup Link event ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 FWD: 741664 MSN: asadoatlamorcilladotcomdotar ICQ: 74005793 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] limit sip sessions
I think I have the same issue... In case usershave an IP Phone on their desks and Softphones on their PCs and are configured with the same username extensions, which phone will ring? The one that last sent the REGISTER... This can be conflicting... - Original Message - From: Script Head To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, February 03, 2006 1:10 PM Subject: Re: [Asterisk-Users] limit sip sessions You should create a secret dialing prefix like if you wanted to dial 1555333222 the user would actually have to dial 548261555333222. This way, even if they snatch the username/password but do not know the prefix, they won't be able to dial. On 2/2/06, Miguel [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote:Shouldn't all sip users have different usernames?(or am I missing some vital detail here?)PaulHYes Paul, Im in El Salvador and my users like to "share" their usernames/passwords and the original owner doesnt like to pay for callshe hasnt made.---Miguel___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallerID popup
On Friday 03 February 2006 10:21, Facundo Ameal wrote: I 'm developing something similar. It a perl script which tells you who is calling but it do it by sendind a jabber message. it's my first perl script so it's not finished yet. i'll share it so you can contribute if you want... http://www.mixdown.ca/~andrew/astbot -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] return code from AGI
not sure what you want, but for multiple returns i use Set(AGI_STATUS=mystatus), so in the dialplan i just check for the variable AGI_STATUS and do whatever i need depending on the status. regardsOn 2/2/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:Hello friends,Asterisk applications like Dial and other commands return codes. When AGI script is executed, it returns -1 on hangup and 0 on non hangup exit. How do I check these return codes from the extensions.conf . I want to check these return codes and control the dialplan.Please help me how do I track this.Thanks all for reading this mail.With warm regards.Vivek J. Joshi. [EMAIL PROTECTED]Trikon electronics Pvt. Ltd.--Sweat saves blood, blood saves lives, and brains saves both.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SV: SV: [Asterisk-Users] delaying answer for a number of ringsor anamount of time
On Fri, 2006-02-03 at 08:49 +0100, [EMAIL PROTECTED] wrote: From what I understand it means that the *hardware* in your computer *acknowledges* the call as soon as it is recieved and then sends it to asterisk dialplan for processing. Hrm. Yes, that is what I got from it. But in my case the hardware is an internal, PCI (Zap) Wildcard. I am pressuming that since I can use functions like Wait(), then Answer() in dialplan to actually delay answering (for the Wait() time) that Asterisk actually acknowledges the call. You would essentially need to put the delay before the call ever reaches asterisk. So this problem isn't asterisk related... if I've understood your question and the answer I found correctly. Hrm. Yeah. Perhaps. I guess perhaps Asterisk isn't currently able to handle deciding if the call has been hung up before it even picks it up (i.e. no more rings). Maybe a peek at the source to Asterisk and the zaptel drivers might tell me more. I just find it strange that I am the first person to want this feature. Indeed tomshardware.com has an article describing how to make an answering machine out of Asterisk by doing exactly what I tried. In my experiments though, it just don't work the way they describe it. Thanx, b. -- My other computer is your Microsoft Windows server. Brian J. Murrell signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Server Wildcard TE110P
Hi, I have an IBM xSeries 206 and now looking at the Wildcard TE110P to connect to our ISDN30. Has anyone any experience with this combination? Would the TE110P work in this server? I've listed the PCI slots the machine has: 2 ( 2 ) x PCI-X / 66 MHz - full-length ¦ 3 ( 3 ) x PCI - full-length Any response is appreciated. Phil. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pattern Match - 0 or 1 digit
_N On 2/3/06, Sum Ding Wong [EMAIL PROTECTED] wrote: Does anyone know of a special character used in pattern matching that would match 0 or 1 digit? I would just like to cut down on the number of extensions I have. Current example: exten = 2125551234,1,Dial(SIP/2125551234,15,rt) exten = 12125551234,1,Dial(SIP/2125551234,15,rt) I would like to do something like this (where + matches 0 or 1 digit): exten = _+2125551234,1,Dial(SIP/2125551234,15,rt) | Or in an area that has 7 digit dialing... exten = 5551234,1,Dial(SIP/2125551234,15,rt) exten = 2125551234,1,Dial(SIP/2125551234,15,rt) exten = 12125551234,1,Dial(SIP/2125551234,15,rt) exten = _5551234,1,Dial(SIP/2125551234,15,rt) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ddi???
Hi, We are ordering a bank of numbers from our provider BT. We will have an ISDN30 with 8 channels enabled. Is it possible to do this? Is this known as DDI? Can anyone give tips on how to configure the Asterisk server so that users are available on the extensions. Hope this explains this better ... 01925 838381 Switchboard 01925 838382 User 1 01925 838383 User 2 01925 838384 User 3 01925 838385 User 4 01925 838386 User 5 01925 838387 User 6 01925 838388 User 7 01925 838389 User 8 01925 838390 User 9 01925 838391 User 10 01925 838392 User 11 01925 838393 User 12 01925 838394 User 13 01925 838395 User 14 etc ... Thank you in advance! Phil___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Events when the target of the call answer
Ezequiel A. Sculli wrote: Hi Group, I am sending my question again why I don’t have answer yet: I am developing a application, this use “Manager API” to connect with Asterisk. But when I call to an external number (over a zap channel), I don’t receive any event when the target answer, Who can help me?, Which event notify me that the phone call was answered? Which type of channel? If analogue (tdm400 etc) then the call is answered as soon as it connects to the pstn, not when the other end answers. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hardware and network requirements
Hi, The usual bottleneck is cpu. i'm planning to migrate a callcenter to asterisk and VOIP, the call center can have up to 25 cuncurrents agents logged in. Ie. max 25 concurrent calls. I'll have some simplty IVR business logic and the some queues. Unknown number of concurrent calls (with prompts and hold-music). Can you provide an estimate on the maximum number of calls in queue ? The second major question is how are you going to recieve the calls ? VoIP (g.729, g.711, ???) PRI card on server ? Question three: I assume you're going to run voip for the agents. What codec are you going to use on the agent side ? (g.729, g.711, gsm) The real question is how much transcoding are you going to do ? Because that's where your clock-cycles are going to be spent. Can a normal server with Pentium 4 3.6 Ghz CPU ... do it. Most likely. It'll do 40-50 concurrent 711 to 729 transcodings. John VoIP Doctor Føroya Tele ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cmd set with multiple values
Christian Benke wrote:
hello!
has this made it into 1.2.3 already:
http://bugs.digium.com/view.php?id=6128 ?
i'm trying to set a variable that should be used as a dialstring in the
dial-command, including parameters seperated with the respective
delimiter, e.g. like:
exten = 907,1,Set(DESTINATION1=Zap/G1/4989123456789|10|gh)
exten = 907,n,Set(DIALSTRING=${DESTINATION1})
exten = 907,n,Dial(${DIALSTRING})
Set multiple variables? One for each option maybe?
--
Cheers,
Matt Riddell
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Re: [Asterisk-Users] Events when the target of the call answer
You recive Link event when Channel Caller and Channel Called bridged, match it and good luck.cheers,Giovanni2006/2/3, Ezequiel A. Sculli [EMAIL PROTECTED]: Hi Group, I am sending my question again why I don't have answer yet: I am developing a application, this use "Manager API" to connect with Asterisk. But when I call to an external number (over a zap channel), I don't receive any event when the target answer, Who can help me?, Which event notify me that the phone call was answered? Thank you. Ezequiel ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Giovanni Miano ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Rewind MusicOnHold?
Dan Journo wrote: Ok, i feel like im getting somewhere but i need a little help. Asterisk displays this when its loading:- [res_musiconhold.so] = (Music On Hold Resource) == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' == Registered application 'StartMusicOnHold' == Registered application 'StopMusicOnHold' == Parsing '/etc/asterisk/musiconhold.conf': Found Use a custom musiconhold class playing ulaw files or whatever - they will start from the beginning each time. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] varion card
I've been using it in a test environment with no problems. However, I haven't used it in production yet. I'm doing some voice broadcasting with a PRI and so far I'm content with the performance. -MC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Akpome Akpoguma Sent: Friday, February 03, 2006 5:47 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] varion card Hi, Has anyone used the varion v400p card? what's its performance?? Rgds, _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pattern Match - 0 or 1 digit
N matches any digit from 2-9. Are there any other wildcards outside of the ones listed below http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns X matches any digit from 0-9 Z matches any digit form 1-9 N matches any digit from 2-9 [1237-9] matches any digit or letter in the brackets (in this example, 1,2,3,7,8,9) . wildcard, matches one or more characters ! wildcard, matches zero or more characters immediately (only Asterisk 1.2 and later, see note) On 2/3/06, C F [EMAIL PROTECTED] wrote: _N On 2/3/06, Sum Ding Wong [EMAIL PROTECTED] wrote: Does anyone know of a special character used in pattern matching that would match 0 or 1 digit? I would just like to cut down on the number of extensions I have. Current example: exten = 2125551234,1,Dial(SIP/2125551234,15,rt) exten = 12125551234,1,Dial(SIP/2125551234,15,rt) I would like to do something like this (where + matches 0 or 1 digit): exten = _+2125551234,1,Dial(SIP/2125551234,15,rt) | Or in an area that has 7 digit dialing... exten = 5551234,1,Dial(SIP/2125551234,15,rt) exten = 2125551234,1,Dial(SIP/2125551234,15,rt) exten = 12125551234,1,Dial(SIP/2125551234,15,rt) exten = _5551234,1,Dial(SIP/2125551234,15,rt) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] inform the agent about the queue he is answering
On 2/3/06, Script Head [EMAIL PROTECTED] wrote: Yes, it is possible. You need to track the queue log and channels via manager console or by tailing logs in real time and then match the destination of the caller by the callerid. Then make the decision which URL to redirect the caller too. None of this comes with Asterisk but it is possible to build. i'd like sto start a project about it, an maybe share my works with some other people, do you think that the solutions regarding tailing logs is stable and affidable? maybe i can build a php utility that returns a xml of the call in the queue the agents logs hiself in this applications and when he receives a call the php applications return a match from his ID and the last result in the XML response... what do you think about that? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FW: Web Interface
I've also purchased their GUI and hoped it would work for us, but the lack of proper documentation, horribly garbled tech support lines (support seems to come from Australia, and they apparently use very low quality voip trunks),broken installer, and cryptic interface forced me to reconsider. After hours and hours of wasted time, I chucked this product in the garbage in favor of AMP... wasted $1K thinking their commercial product would be significantly better than free options, I was wrong. I have been playing with the Signate switch. Official training starts soon but just playing with it leaves me with the impression that it is powerful but very complex. You need to RTFM to get anything working. They have also used IonCube to encode all PHP and HTML files so customization is impossible without reverse engineering :-( ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] click to talk
hi to all i have a website powered by a c# CMS i have an asterisk in our office my need is that my customers could surf on my website, click the phone button, a sip call is established between the website (sip client) and my phone allowing me to talk with them any idea where i can find the sip client to embed in mywebsite ? (c# - java or whatever) thnx in advance Graziano ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura SPA-2002 rings randomly
About once an hour, my Sipura 2002 rings just once. I thought it might be faulty, so I configured a second one, and it does the same thing. I updated the firmware to 3.1.5 and still have the same problem. Anybody able to shed some light on the random ringing problem? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] meetme and dtmf
Accursio Avona wrote: but maybe also (i'm not sure, i had not the time to study well enough the source, and over all i'm not a so good c programmer) that this part of code prevents asterisk to broadcast the sound to other channels when it is not inband. MeetMe is not designed to pass DTMF through between the parties in any case. It may happen if you use inband DTMF and don't have Asterisk actually paying attention to DTMF for any reason, but it's not intended to work that way. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ddi???
[EMAIL PROTECTED] wrote: Hi, We are ordering a bank of numbers from our provider BT. We will have an ISDN30 with 8 channels enabled. Is it possible to do this? Is this known as DDI? Can anyone give tips on how to configure the Asterisk server so that users are available on the extensions. Hope this explains this better ... 01925 838381Switchboard 01925 838382User 1 01925 838383User 2 01925 838384User 3 01925 838385User 4 01925 838386User 5 01925 838387User 6 01925 838388User 7 01925 838389User 8 01925 838390User 9 01925 838391User 10 01925 838392User 11 01925 838393User 12 01925 838394User 13 01925 838395User 14 etc ... Thank you in advance! Yes, no problem to do. I've done this for our manchester office (although with fewer people and on NTL's ISDN30). You need to get BT to agree and allocate or port the numbers. You need to agree how many digits BT will pass on to you (probably 1925838395 but possibly just the last 2) If you want your users Direct dial numbers to be presented when the make calls then you need to tell BT that too, or they will just use the 'main' number for everything. Once you have that agreed, there is some work to do in extensions.conf to make it all join up, but it wasn't hard. Drop me a mail if you need a hand or some examples. (It gets easier if you make your internal extensions match up in some way to the external numbers eg 382 as user 1's internal extension number, it isn't essential, but it lets you write a simpler dialplan based on pattern matching) Tim. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: delaying answer for a number of ringsor an amount of time
On Fri, 2006-02-03 at 12:19 +0100, Wilson Pickett wrote: As someone must have already said, it's not a good idea to share lines with asterisk. Well, yeah, ideally I have the phones on an FSX, but a) I don't have one yet and b) I want to make sure I am happy with running a PBX before I invest in a. :-) The above code made asterisk *never* pickup, so it must be possible, I hope so. I'm going to dig into source to see. :-/ b. -- My other computer is your Microsoft Windows server. Brian J. Murrell signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Web interface
Steve Totaro wrote: Another question, If Signate is not using ABE, what are their requirements for releasing source as far as the GUI? The Asterisk GPL has no bearing on the external tools used to manage/configure it, unless those tools require changes in Asterisk itself or loadable modules that are used in the Asterisk instance. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP question
Hi, I have a provider sending me data through SIP, but with no registration. (there are constraints that forces us to work like this). And, as far as I am concerned, that's fine. Here is the relevant portion of my SIP.conf file. [514907]context=514907-inboundtype=friendhost=11.222.222.23language=frdisallow=allallow=ulawdtmf=rcf2833 Basically, I understand that I am saying everything coming in from 11.222.222.23 should be sent to the context514907-inbound. Right? If it is, how do I ask this provider for another DID, let`s say 555-555-, and send those calls ina different context (let`s say 55-inbound)??? There doesn't seem to be a way of differenciating between calls meant for 514907 and 55, since they both come in from the same provider (hence same IP address). What am I missing to treat those calls differently? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: delaying answer for a number of rings or an amount
On Fri, 2006-02-03 at 07:37 -0700, Bromont Quebec wrote: Well in my setup I have a few IP phones connected to Asterisk as well as POTS phones on my analog line. Ahhh. So we share the latter at least. When a call for my daughter comes in on the analog line (determined from callerID) I send it to her own voicemail after 20 seconds of ringing. It all works quite well. Hrm. Yeah, this is what I'm trying to do. Here's a step-by-step of what happens below: 1 - a call comes in and Asterisk rings SIP/Brian and SIP/joe for 30 seconds. So you don't want Asterisk to wait and see if the POTS line is picked up before ringing the SIP phones? Interesting. 2 - After 30 seconds if the line is still ringing (nobody picked up POTS phone or SIP phones) * answers the line and sends to Voicemail. Asterisk never picks up the call until the 30 seconds are up. What seems to be happening here is that even if somebody picks up the POTS line within a few seconds, after the 30 seconds (Wait() in my case, but I'd imagine the same will happen after ringing the SIP lines for 30s) is up Asterisk is also on the POTS line (with the callee who picked up the POTS phone) doing the voicemail intro and recording the conversation. [from-pots] exten = s,1,Dial(SIP/brianSIP/joe,30) exten = s,2,Voicemail(u2001) exten = s,3,Hangup I will try this exactly and see if it works any better. b. -- My other computer is your Microsoft Windows server. Brian J. Murrell signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_sccp availability?
hi, I'm trying to get the latest chan_sccp. The links from http://chan-sccp.berlios.de are all dead. Is it just me? Does anyone know an alternate source to get chan_sccp? Seems like chan_sccp is the way to go if I have a bunch of cisco phones I don't want to use SIP on? Anyone else have an opinion on the built in asterisk skinny vs the chan_sccp? thanks, Andy ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura SPA-2002 rings randomly
On 2/3/06, Jeremy Koski [EMAIL PROTECTED] wrote: About once an hour, my Sipura 2002 rings just once. I thought it might be faulty, so I configured a second one, and it does the same thing. I updated the firmware to 3.1.5 and still have the same problem. Anybody able to shed some light on the random ringing problem? Do you have audible MWI turned on? -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Events when the target of the call
Matt Riddell wrote: Ezequiel A. Sculli wrote: Hi Group, I am sending my question again why I don_t have answer yet: I am developing a application, this use _Manager API_ to connect with Asterisk. But when I call to an external number (over a zap channel), I don_t receive any event when the target answer, Who can help me?, Which event notify me that the phone call was answered? Which type of channel? If analogue (tdm400 etc) then the call is answered as soon as it connects to the pstn, not when the other end answers. Yes, I am use a tdm400, and I need receive an event when the target of the call answer, not when the analog channel connects to the pstn, Is possible? Thank you ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] php+agi
Hello, I want to know if someone made a script in php(with agi) to call some voip number, and when the user answer the call, he hears a message with an advertisement. I want to input the number directly from cli or read the numbers from a file(ex.8021,8022,8023). Thanks in advantage Ever Zalazar ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: 5, 000 concurrent calls system rolloutquestion
One of our Telephony Server 5000 modules will throughput between 2,000 and 2,500 SIP calls with streams if it is doing no other work. One of these days we will again announce the details of the ongoing benchmarks that we perform with the help of system engineers from a major computer manufacturer. The key statement is if it is doing no other work. If a server is playing IVR or hosting conferences, throughput declines in unpredictable ways depending on the actual mix of work. So when we spec a system for a particular call volume we use relatively conservative engineering to ensure that the system can handle the peak load. In real applications, we rate a box at less than half of its peak call throughput. So for 5,000 calls, we'd probably use five servers plus an extra one for failover. Someone trying to do that same amount of work with PC servers might need up to four dozen of them in a complex configuration with a central voicemail store. The load balancing and system management problems are considerable. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wai Wu Sent: Thursday, February 02, 2006 9:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] RE: 5,000 concurrent calls system rolloutquestion I don't think they are doing it with one Asterisk box. They did say one rack of servers. Well, that might mean up to 50 computers if they are using blade servers. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Todd Sent: Thursday, February 02, 2006 10:21 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] RE: 5,000 concurrent calls system rolloutquestion [top-posting continued due to formatting sloth on my part] So, then let me follow up with a few more comments: 1) I will make some assumptions from your note: a) Asterisk is currently capable (unless something has broken recently) of handling 2500 SIP-SIP calls with no transcoding, including RTP sessions, if on an operating system and hardware that is appropriately configured. This puts to rest some who have claimed that 5000 channels is impossible with Asterisk regardless of platform, at least according to Signate. b) It is unclear if other channel drivers (IAX and Zaptel, specifically) have had any testing with significant numbers of channels. c) It is unclear if anything other than pure RTP passthrough is viable in these configurations. Maybe IVR causes collapse. ? 2) Still no claims or comments on the specific testing methods, or on methodology. I'm left still scratching my head as to if this is actually possible, since there is no specific claim that can be verified. While I hope that your system can do those numbers (it would help me greatly in the future!) I can't say that I'm confident yet. I'll follow up in private email for further discussion. 3) Nobody else has thus far taken the bait and made any comments about their systems. I appreciate Signate's comments; they seem to be the only ones to publicly claim large-scale throughput using Asterisk in a public forum. Most other people who claim thousands or even high hundreds of connections do so offhand, without responding to second questions when I raise my figurative eyebrows. 4) There are still no notes on other problems with scale here. I've had systems with several hundred simultaneous SIP connections, but sip show channels sure does start to take a while. What _other_ problems crop up, but don't necessarily cause a failure condition? 5) I will agree that most SIP testing systems are currently too pricey. I would love to find a well-connected network that rents out a few of the better-known SIP testing tools to beat on Asterisk installations in remote places for short periods of time. But this has always been the case... test gear is a small market, and expensive. Just look at the MSRP of new high-end HP Oscilloscopes if you want to get a picture of price-gouging. JT At 11:21 AM -0800 2/2/06, William Boehlke wrote: Signate has claimed 5,000 streams, or 2,500 calls, on a single Telephony Server 5000. The throughput has little to do with Asterisk and a lot to do with hardware design and operating system tuning. Our very minor code changes were returned to the project last year. The benchmark we used to make that initial claim was flawed, however we have since replicated the throughput in a different way to save our marketing bacon. How we actually achieve the throughput is our intellectual property but we have a number of customers who are scaling towards and past that traffic level. One of these days we hope to be able to justify the very large fee Hammer wants to extract from us to produce a third party verification. In production environments, of course, systems do more than switch calls. We think high volume system design using 32-bit systems of any kind is complex,
RE: [Asterisk-Users] Sipura SPA-2002 rings randomly
Listen to your voicemail. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Koski Sent: Friday, February 03, 2006 9:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Sipura SPA-2002 rings randomly About once an hour, my Sipura 2002 rings just once. I thought it might be faulty, so I configured a second one, and it does the same thing. I updated the firmware to 3.1.5 and still have the same problem. Anybody able to shed some light on the random ringing problem? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_sccp availability?
On Fri, 3 Feb 2006, Andy Webster wrote: hi, I'm trying to get the latest chan_sccp. The links from http://chan-sccp.berlios.de are all dead. Is it just me? Does anyone know an alternate source to get chan_sccp? Seems like chan_sccp is the way to go if I have a bunch of cisco phones I don't want to use SIP on? Anyone else have an opinion on the built in asterisk skinny vs the chan_sccp? http://chan-sccp.org Armin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] meetme and dtmf
Kevin P. Fleming wrote: but maybe also (i'm not sure, i had not the time to study well enough the source, and over all i'm not a so good c programmer) that this part of code prevents asterisk to broadcast the sound to other channels when it is not inband. MeetMe is not designed to pass DTMF through between the parties in any case. It may happen if you use inband DTMF and don't have Asterisk actually paying attention to DTMF for any reason, but it's not intended to work that way. This means that if i'd like to use iax2 protocol (i need to integrate, into a propietary crm, calling features though asterisk, and i thougth to use iaxclient dll) i can't pass DTMF through between the parties? If so is it possible to modify meetme.c to avoid this behaviour? or i must use sip protocol. Thank's Accursio Avona ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_sccp availability?
Andy Webster ha scritto: hi, I'm trying to get the latest chan_sccp. The links from http://chan-sccp.berlios.de are all dead. Is it just me? Does anyone know an alternate source to get chan_sccp? Just tested, all the links work Sergio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sipura SPA-2002 rings randomly
As BJ mentionned, it could be your MWI of depending on your profiling, it might be scheduled to download it's profile every hour, and therefore might reboot and ring after each download -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Koski Sent: February 3, 2006 12:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Sipura SPA-2002 rings randomly About once an hour, my Sipura 2002 rings just once. I thought it might be faulty, so I configured a second one, and it does the same thing. I updated the firmware to 3.1.5 and still have the same problem. Anybody able to shed some light on the random ringing problem? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallerID popup
On Fri, 3 Feb 2006 11:41:53 +0100 Giovanni Miano [EMAIL PROTECTED] wrote: Link event For me, Link event only occurs when the called number pickup the call. I prefer 'Newchannel' event when the 'State' are equal to 'Ringing' -- Iuri Gomes Diniz adm.iuri (at) digi.com.br Network Admin and Programmer [http://clx.digi.com.br] DIGINET [http://www.digi.com.br] Natal - RN - Brazil. -- Iuri Gomes Diniz adm.iuri (at) digi.com.br Network Admin and Programmer [http://clx.digi.com.br] DIGINET [http://www.digi.com.br] Natal - RN - Brazil. -- Esta mensagem foi verificada pelo sistema de anti-virus e acredita-se estar livre de perigo. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP question
I'd change your definition to something like [providerX] context=providerX-inbound host=11.222.222.23 in your providerX-inbound context you can match the different extensions [providerX-inbound] exten = 514907,1,NoOp(514907) exten = 55,1,NoOp(55) Now a question I've always wondered, What if providerX uses multiple IPs. Is there any way to specify a range of IPs for the "host" in sip.conf ? So far I've had to make a sip entry for each IP my provider uses. Thanks Ben From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michaël GaudetteSent: February 3, 2006 12:47 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] SIP question Hi, I have a provider sending me data through SIP, but with no registration. (there are constraints that forces us to work like this). And, as far as I am concerned, that's fine. Here is the relevant portion of my SIP.conf file. [514907]context=514907-inboundtype=friendhost=11.222.222.23language=frdisallow=allallow=ulawdtmf=rcf2833 Basically, I understand that I am saying everything coming in from 11.222.222.23 should be sent to the context514907-inbound. Right? If it is, how do I ask this provider for another DID, let`s say 555-555-, and send those calls ina different context (let`s say 55-inbound)??? There doesn't seem to be a way of differenciating between calls meant for 514907 and 55, since they both come in from the same provider (hence same IP address). What am I missing to treat those calls differently? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SIP question
Benjamin, Thanks a lot for the answer. Sometimes the obvious escapes me, and this was the case here. Regards, Mike I'd change your definition to something like [providerX] context=providerX-inbound host=11.222.222.23 in your providerX-inbound context you can match the different extensions [providerX-inbound] exten = 514907,1,NoOp(514907) exten = 55,1,NoOp(55) Now a question I've always wondered, What if providerX uses multiple IPs. Is there any way to specify a range of IPs for the host in sip.conf ? So far I've had to make a sip entry for each IP my provider uses. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyway to do this?
Hi, Sorry to ask a slightly off topic question here, but I've been stuck on this for a while. My SIP ATA's are displaying callerID without problems. The problem is when a 2nd call comes in during a conversation, callwaiting callerID dosen't show up. I can only hear the callwaiting alert tones, but no callwaiting callerID. I have both callwating=yes and callwaitingcallerid=yes in my zapata.conf Can anyone please help me out here? Thanks. Andy On 2/1/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: If callerid is received, it will be displayed on the sip phones. My guess would be that it's not coming in on the analog line in the first place. PaulH Scott Geist [EMAIL PROTECTED] wrote: How do you retreive the caller id on incoming analog lines and display the id on the sip phones on the network? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SV: SV: [Asterisk-Users] delaying answer for a number of ringsor anamount of time
At 07:34 AM 02/03/2006, you wrote: I am pressuming that since I can use functions like Wait(), then Answer() in dialplan to actually delay answering (for the Wait() time) that Asterisk actually acknowledges the call. I think you need to use dial instead of answer. You can put a timeout in dial and if the call is hung up dial will exit. If it exited due to hangup the call will not be answered and the voicemail call will be ignores. How you set up something to dial to that can't answer is beyond me, but if you can figure it out, that should work. Ira ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sipura SPA-2002 rings randomly
Thanks, that was it. On Fri, 3 Feb 2006, Benjamin Lawetz wrote: As BJ mentionned, it could be your MWI of depending on your profiling, it might be scheduled to download it's profile every hour, and therefore might reboot and ring after each download -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Koski Sent: February 3, 2006 12:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Sipura SPA-2002 rings randomly About once an hour, my Sipura 2002 rings just once. I thought it might be faulty, so I configured a second one, and it does the same thing. I updated the firmware to 3.1.5 and still have the same problem. Anybody able to shed some light on the random ringing problem? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SV: SV: [Asterisk-Users] delaying answer for a number of ringsor anamount of time
On Fri, 2006-02-03 at 10:59 -0800, Ira wrote: I think you need to use dial instead of answer. You can put a timeout in dial and if the call is hung up dial will exit. Hung up? By whom? Assume this: while Dial() is working (and waiting for the timeout) somebody has picked up a phone that shares the POTS line with Asterisk. Will that second pick up of the POTS line look like a hangup on the POTS line to Asterisk while it is Dial()ing? b. -- My other computer is your Microsoft Windows server. Brian J. Murrell signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Sipura SPA-2002 rings randomly
My Sipura 1001 does that when I have a message waiting. You can turn the reminder half-ring off in the configuration settings. About once an hour, my Sipura 2002 rings just once. I thought it might be faulty, so I configured a second one, and it does the same thing. I updated the firmware to 3.1.5 and still have the same problem. Anybody able to shed some light on the random ringing problem? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyway to do this?
The way to first test the ATA is with a phone or caller ID display that supports caller ID with call waiting. Some devices work that way by default and some might require you to set the option. That test should have nothing to do with zapata.conf at all. I assume the reason you mention zxapata.conf is that you have the ATA FXS port connected to some FXO hardware that relates to zapata.conf? Andy Kuo wrote: Hi, Sorry to ask a slightly off topic question here, but I've been stuck on this for a while. My SIP ATA's are displaying callerID without problems. The problem is when a 2nd call comes in during a conversation, callwaiting callerID dosen't show up. I can only hear the callwaiting alert tones, but no callwaiting callerID. I have both callwating=yes and callwaitingcallerid=yes in my zapata.conf Can anyone please help me out here? Thanks. Andy On 2/1/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: If callerid is received, it will be displayed on the sip phones. My guess would be that it's not coming in on the analog line in the first place. PaulH Scott Geist [EMAIL PROTECTED] wrote: How do you retreive the caller id on incoming analog lines and display the id on the sip phones on the network? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] S100-FX v2.0
Curious about this device as well. Seems almost too good to be true? The built-in switch feature would be much handier than having to get a router for the extra network connection; Also, the 'life line passthru' thing seems interesting -- although I have no idea what a life line passthru is! haha It would be amazing if it was an FXO, but obviously isn't -- I'm assuming you can dial out on that line if necessary? 'life line' concept? Anyone had a good experience with one of these IN THE UK?? (or any country outside of North America for that matter) power issues etc Regards - Original Message - From: Mike Hammett To: asterisk-users@lists.digium.com Sent: Friday, January 27, 2006 1:47 AM Subject: [Asterisk-Users] S100-FX v2.0 I just saw the S100-FX v2.0 on eBay. I was wondering if anyone has tried it out and what their opinion of it was. Mike HammettIntelligent Computing Solutionshttp://www.ics-il.com -- This message has been scanned for viruses and dangerous content and is believed to be clean. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- This message has been scanned for viruses and dangerous content and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: delaying answer for a number of ring or an amount of time
Yeah I want both my POTS phones and SIP phones to ring at the same time, that way I have the choice to answer whatever one is most convenient. If a POTS phone picks up, the Zap channel closes and Asterisk does nothing more, if a SIP phone picks up, Asterisk connects the Zap channel to SIP/whatever. The only trouble I run into is if a POTS phone picks up right at the 30 second mark, then I get Asterisk passing the Zap channel to Voicemail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: delaying answer for a number of ring or an amount of time
On Fri, 2006-02-03 at 13:00 -0700, Bromont Quebec wrote: Yeah I want both my POTS phones and SIP phones to ring at the same time, that way I have the choice to answer whatever one is most convenient. If a POTS phone picks up, the Zap channel closes and Asterisk does nothing more, if a SIP phone picks up, Asterisk connects the Zap channel to SIP/whatever. The only trouble I run into is if a POTS phone picks up right at the 30 second mark, then I get Asterisk passing the Zap channel to Voicemail. Cool. That seems to work. It would seem, if the Zap driver sees another device on the line pick up, it sends a hangup to Asterisk. Very cool. Thanx for all the input. b. -- My other computer is your Microsoft Windows server. Brian J. Murrell signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: 5, 000 concurrent calls system rolloutquestion
There you go. if it is doing no other work is key phrase. A lot of PC can do that these days if all it has to do is re-route packets to different destinations, and guess what, if you make sure silence compression is turned on at the endpoints, you can claim even more streams can be passed through. The trict here is how * stores the mapping pair and how effiecent its lookup process is. I have not looked at this part of the code in *, but would be interesting to find out. On another topic. How many calls do you think one server can handle if every calls goes to a different IVR script of its own? Lets assume there is no trans-coding. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of William Boehlke Sent: Friday, February 03, 2006 1:29 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] RE: 5,000 concurrent calls system rolloutquestion One of our Telephony Server 5000 modules will throughput between 2,000 and 2,500 SIP calls with streams if it is doing no other work. One of these days we will again announce the details of the ongoing benchmarks that we perform with the help of system engineers from a major computer manufacturer. The key statement is if it is doing no other work. If a server is playing IVR or hosting conferences, throughput declines in unpredictable ways depending on the actual mix of work. So when we spec a system for a particular call volume we use relatively conservative engineering to ensure that the system can handle the peak load. In real applications, we rate a box at less than half of its peak call throughput. So for 5,000 calls, we'd probably use five servers plus an extra one for failover. Someone trying to do that same amount of work with PC servers might need up to four dozen of them in a complex configuration with a central voicemail store. The load balancing and system management problems are considerable. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cmd set with multiple values
At 08:07 AM 02/03/2006, you wrote:
exten = 907,1,Set(DESTINATION1=Zap/G1/4989123456789|10|gh)
exten = 907,n,Set(DIALSTRING=${DESTINATION1})
exten = 907,n,Dial(${DIALSTRING})
Set multiple variables? One for each option maybe?
Or call a macro instead and have the macro split it apart, then the
code stays the same and the macro can hide all the issues.
exten = 907,n,Macro(parse_dial,${DIALSTRING})
Ira
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Re: SV: SV: [Asterisk-Users] delaying answer for a number of ringsor anamount of time
At 11:38 AM 02/03/2006, you wrote: Hung up? By whom? Assume this: while Dial() is working (and waiting for the timeout) somebody has picked up a phone that shares the POTS line with Asterisk. Will that second pick up of the POTS line look like a hangup on the POTS line to Asterisk while it is Dial()ing? Sure seems to work that way here. I have a 4 line analog phone sharing the phones with * and if I grab it before a * goes to voicemail it never goes to voicemail. Both my analog and SIP phones are ringing at the same time. Ira ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] error cdr mysql addon
Hi, After installing mysql, mysql-devel mysql cdr add on, I get the following error when I start Asterisk: [res_config_mysql.so]2006-02-03 18:41:16 WARNING[24786]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/res_config_mysql.so: undefined symbol: _intel_fast_memcpy My server has the following MySQL rpms: rpm -qa | grep MySQLMySQL-server-4.0.20-0MySQL-shared-compat-4.0.18-0MySQL-devel-4.0.20-0perl-DBD-MySQL-2.1021-3MySQL-client-4.0.20-0 Any ideas? Thank you!Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
