Re: [Asterisk-Users] new AMPortal and Asterisk debs
On Sun, Jan 08, 2006 at 06:16:16PM -0800, Mike Fedyk wrote: Tzafrir Cohen wrote: Experimental: Asterisk 1.2: At the moment they are not that experimental anymore and should be ready for use, but are not well-tested yet. To use it, define both sources: deb http://rapid.dotsrc.org/ experimental/ How does this compare with Asterisk 1.2.1.dfsg-1 that is in etch/testing and 1.2.1.dfsg-3 that is in sid/unstable? Testing (Etch) is slightly behind. It is generally in line with the packages in Sid. Sort of. Actually ff you compare the changelog you'll find some striking similarities. In fact, it is based on the current version in the pkg-voip svn than to the current version. However it is built for Sarge (Stable). So if you have Sarge installed, you won't have to upgrade libc6/pgsql/pwdlib/whatever to use it. I try to commit most of the relevant changes and fixes to the main Debian package, so if you have Sid/Etch, you'll end up getting basically the same packages (only with a more changing base system...) . -- 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
RE: [Asterisk-Users] OT: SIP aware firewalls?
Weuse Juniper/Netscreen 5GT's with the latest 5.3 firmware.It is fully sip aware and in a NAT environment it modifies the addresses in the SIP frames according the NAT table.The netscreen also checks the sip frame for the udp ports to be opened for the audiochannels and openn them for the session only. Wehave clients and servers inside and outside, and everything talks SIP and works like a charm. Regards. Andre VinkVink Consultancy - Oorspronkelijk Bericht -Onderwerp:RE: [Asterisk-Users] OT: SIP aware firewalls?Afzender: Chris Bagnall [EMAIL PROTECTED]Aan:\'Asterisk Users Mailing List - Non-Commercial Discussion\' asterisk-users@lists.digium.comDatum:07-01-2006 1:25 I know that I can stay with m0n0. The question still stands; are there circumstances when something more is required? Would something be gained by such a migration.I would think the only real circumstances where true SIP-aware firewallswould be required would be in an environment where one had many SIP devicesbehind a NAT (and by many I mean more than it\'s reasonably practical toassign different port numbers to).I\'m no expert on firewalls, so hopefully someone\'ll correct me if I\'mmistaken.Regards,Chris-- C.M. Bagnall, Director, Minotaur I.T. LimitedThis email is made from 100% recycled electrons___--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] ISDN beronet: cannot send digits during outbound calls
Hi, we are trying a beronet ISDN card with asterisk 1.2 on debian sarge distro. Everything seems fine except for outbound calls: it seems we cannot send outbound digits so we cannot use phone digits to use ivr menus. I followed beronet dinstallation document. Is there some parameter missing to add to configuration files? TIA Giorgio Incantalupo ___ --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] JiveMessenger HOWTO
On 1/8/06, Chris Bagnall [EMAIL PROTECTED] wrote: Has anyone had experience using Asterisk-IM/Jive Messenger with any IMclients apart from Trillian and Spark? (Trillian costs money and I'm not that keen on Spark's lack of configurability) I've been looking as well. Unfortunately there's really not that much out there. For that matter I think most of the jabber clients suck.-- Is it something someone said, was it something someone said? ___ --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] Incoming PSTN Calls - Stumped
Hi,
The hostname that you used in your register directive ('provider.ie')
must have a corresponding section in sip.conf. In your example, you used
'[provider-in]'. If that is what you actually used, then this might
explain why your incoming goes to the default context because it
couldn't find its own section. Try renaming '[provider-in]' to
'[provider.ie]'.
-kokmeng.
Aisling O'Driscoll wrote:
Hi,
Yes InternalExtension is the context and 2093 the extension.
Just to explain something odd that’s happening (and I’m very stumped
with this)….I think my contexts are definately the reason that I
can’t interrupt the menu for incoming pstn calls to choose a submenu:
My users register with my sip proxy (SER). Therefore when I create an
entry for them in sip.conf I set only one context. Also to allow for
incoming calls from my provider it seems I must direct the calls
firstly to a ‘dummy’ extension.
sip.conf
register = username:[EMAIL PROTECTED]/2093
[provider-in]
type=peer
host=sip.provider.ie
context=onecontext
[2092]
type=peer
other stuff
context=onecontext
So the dummy extension here is ‘2093’ and 2092 is a phone who
registers with SER and when SER redirects to Asterisk uses the
‘onecontext’ context.
Now in my extensions.conf ‘onecontext’ includes other contexts. This
is how I get access to conference calls, creating IVR menus etc. Also
the main purpose of ‘onecontext’ is to allow outgoing access to the
PSTN.
[onecontext]
include = createmenu//creating an IVR menu
include = createconf//creating a conf call
etc
include = default //used for voicemail
[createmenu]
;does something
[createconf]
;does something
;outgoing calls – main purpose of onecontext
exten = _X.,1,Dial(SIP/[EMAIL PROTECTED])
exten = _X.,2,Hangup
[default]
;mailbox for 2092 and other users
Now this is where the problems start! For incoming calls I tried to
do “include = incomingpstn” in ‘onecontext’ which I thought would
call a new context called ‘incomingpstn’ which would have an entry
for the dummy user. i.e.
[incomingpstn]
exten = 2093,1,Wait(1)
exten = 2093,n,Background(MainMenu)
exten = 1,1,Goto(InternalExtension,2093,1)//directs to another
context called Internal Extension
I also changed the [provider-in] for context=incomingpstn in my
sip.conf. However this didn’t work and I kept getting directed to the
voicemail of my pstn provider. The ONLY way I could get the incoming
calls working was to add the contents of the ‘incomingpstn’ context
to the default context i.e.
[default]
exten = 2093,1,Wait(1)
exten = 2093,n,Background(MainMenu)
exten = 1,1,Goto(InternalExtension,2093,1)//directs to another
context called Internal Extension
With this I can hear the MainMenu when I dial my DDI but I can’t seem
to interrupt to divert to another submenu. In the testing that I have
done the user that is making the call is 2092 registered with SER. If
I change the context of 2092 directly in sip.conf to incomingpstn,
then I can hear the menu and interrupt to go to the submenu. But
obviously then I don’t have access to the other features in Asterisk.
The point is that I’m stumped as to why it only works in the default
context and if this is the case how do I get it to call the submenu.
This is what comes up on my asterisk console:
-- Executing Dial (“SIP/2092-2829”, “SIP/[EMAIL PROTECTED]) in
new stack
-- Called [EMAIL PROTECTED]
-- Playing ‘MainMenu’ (language ‘en’)
-- other messages (not relevant I think)
== Spawn extension (outgoing, 021123456, 1) exited non-zero on
‘SIP/2092-5837’
== Spawn extension (default, 2093, 2) exited non zero etc etc
I’m very stuck on this and can’t figure it out.
Any help appreciated.
Many thanks,
Aisling.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Giovanni Miano
Sent: 05 January 2006 21:09
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Incoming PSTN Calls
Is Exist InternalExtension context ? and 2093 exten ?
2006/1/5, Aisling [EMAIL PROTECTED]:
Hi all,
I am having difficulty getting incoming PSTN calls working. I have
set up an account with a third party provider. In my system, the user
register with SER and use Asterisk for PSTN access, voicemail etc
My provider told me to change my sip.conf as follows
register = username:[EMAIL PROTECTED]/2093
; To receive incoming calls specify this block and replace
yourcontext for your dial plan.
[blueface-in]
type=peer
host=sip.blueface.ie
context=incomingpstn
And then in my extensions.conf to have something similar to the
following (or however I wanted to handle my incoming calls)
[incomingpstn]
exten = 2093,1,Wait(1)
exten = 2093,n,Background(MainMenu)
exten = 1,1,Goto(InternalExtension,2093,1)
//press 1 for internal extensions.
This didn't work and I kept getting a 404 not found error saying the
user didn't exist. I tried creating the user in sip.conf and
[Asterisk-Users] Re: Asterisk CLI | more
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... If you're wanting to scroll through output from a CLI command, use: asterisk -rx command | less Thank to bouth of you. -- Tomislav Parcina [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
[Asterisk-Users] Re: Remotely reboot SIP Phones ?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... An example SIP friend is defined as [112], so we could now type, from the CLI: sip notify polycom-check-cfg 112 sip notify cisco-check-cfg 214 doesn't seam to do anything. I have sip_notify.conf in my /etc/asterisk/ directory. Cisco 7905 and 7940 phones don't react on that command. -- Tomislav Parcina [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] spandsp, rxfax, TDM400/zaptel, missed frames, any help?
Sorry in advance if this is a FAQ... I've got a working Asterisk setup based on [EMAIL PROTECTED] 2.2. I have a TDM400 card with 2 FXS and 2 FXO ports; PSTN connections come in via the TDM card. I haven't been able to get inbound fax with spandsp and rxfax to work. Occasionally an all-text fax will come in, though it's usually badly corrupted, but in most cases, it would appear that the call is terminated without successful transmission of the fax. I get logs that look what's included below. From reading the list, it looks like this is caused by the TDM card missing frames. Does that sound correct? If so, is there any relief in sight? Its been a problem since the card came out a couple of years ago. So, no it does not appear there is any relief in sight. ___ --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] Budge Tone-100 as a Ext in the LAN / please help
I am probably thinking that [EMAIL PROTECTED] might be a better way to start your journey PaulH - Original Message - From: luke devon To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, January 09, 2006 4:50 PM Subject: Re: [Asterisk-Users] Budge Tone-100 as a Ext in the LAN / please help Thanx for the reply and the help, but i want to tell u , i'm a newbe for asterick . as to use any opensouce , i gone through the docs and the installation guide , read the few faqs as well. So finally i got installed successfully. To talk with GrandStream Budge Tone- 100 , do i have to install and configure , whole packeges belongs to the project, ??? Asterisk Version 1.2.1Zaptel Version 1.2.1Libpri Version 1.2.1Addons Version 1.2.1Sounds Version 1.2.1 but in my case i installed Asterisk Version 1.2.1Zaptel Version 1.2.1Libpri Version 1.2.1 can some one help me to use that BT-100 phones as extentions via my LAN ?? Is there any giude can i find in the net for configure Asterick in fedora machine or redaht linux machine ? and how to configure for BT-100 in asterick also , Thanx in advance, Luke. Yair Hakak [EMAIL PROTECTED] wrote: lukeuse the wiki.(always wanted to do that)http://www.voip-info.org/wiki/view/Asterisk+phone+grandstream+budgetonehope this helps,yairOn 1/6/06, luke devon <[EMAIL PROTECTED]>wrote: HI , I installed asterisk in fedora core 3 machine perfectly. and i have 10 units of GrandStream IP phone ( Budge Tone-100 ) . I wanted to know how can i use it as extentions in my LAN ? Asterisk PBX alredy there. I didn't try to do any configurations of any files . What are the configurations has to be made with asterisk ? Thanx in advance, Luke. Send instant messages to your online friends http://uk.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Send instant messages to your online friends http://uk.messenger.yahoo.com ___--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] GradStream Budge Tone - 100 / PLease help
Hi , I wanted to connect GradStream Budge Tone - 100 phone with a Asterisk box for acc them as extentions on the LAN .1. After configure Asterisk in a Linux box with different ip network can i , use the other IP phones over the LAN ??2. Asterisk installed machine can wein the LAN as the PBX ? 3.Then the other IP phones , can we connect to the other remained LAN ports as extensions ? Thank you , Luke Send instant messages to your online friends http://uk.messenger.yahoo.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] Incoming PSTN Calls - Can't interrupt Main Menu
Hi,
Thanks to both Iqbal and Kokmeng for the replies.
Kokmeng I tried what you suggested however no luck...
What I have done which is currently working(kind of) is that in my
sip.conf in the [general] section I have set context=incomingpstn. My
register line looks like:
register = username:[EMAIL PROTECTED]/
In my extensions.conf I then have
[incomingpstn]
exten = s,1,Wait(1)
exten = s,n,Background(MainMenu)
exten = 1,1,Goto(internalExt,s,1)
exten = 2,1,Goto(mainconfmenu,s,1)
[internalExt]
exten = s,n,Background(InternalExtension)
[mainconfmenu]
exten = s,n,Background(MainConfMenu)
I can hear the MainMenu sound file being played. What's strange is that
when I press '1' to interrupt, which in my logic should invoke the
internalExt context, nothing happens. The MainMenu sound file continues
to play and finally I get the error:
Warning: pbx.c:2405 __ast_pbx_run: Timeout, but no rule 't' in context
'incomingpstn'
I used the 'Goto' as Iqbal suggested instead of includes...
Has anyone ever experienced this kind of behaviour before?
Many thanks,
Aisling.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of KokMeng
Loh
Sent: 09 January 2006 08:53
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Incoming PSTN Calls - Stumped
Hi,
The hostname that you used in your register directive ('provider.ie')
must have a corresponding section in sip.conf. In your example, you used
'[provider-in]'. If that is what you actually used, then this might
explain why your incoming goes to the default context because it
couldn't find its own section. Try renaming '[provider-in]' to
'[provider.ie]'.
-kokmeng.
Aisling O'Driscoll wrote:
Hi,
Yes InternalExtension is the context and 2093 the extension.
Just to explain something odd that's happening (and I'm very stumped
with this)..I think my contexts are definately the reason that I
can't interrupt the menu for incoming pstn calls to choose a submenu:
My users register with my sip proxy (SER). Therefore when I create an
entry for them in sip.conf I set only one context. Also to allow for
incoming calls from my provider it seems I must direct the calls
firstly to a 'dummy' extension.
sip.conf
register = username:[EMAIL PROTECTED]/2093
[provider-in]
type=peer
host=sip.provider.ie
context=onecontext
[2092]
type=peer
other stuff
context=onecontext
So the dummy extension here is '2093' and 2092 is a phone who
registers with SER and when SER redirects to Asterisk uses the
'onecontext' context.
Now in my extensions.conf 'onecontext' includes other contexts. This
is how I get access to conference calls, creating IVR menus etc. Also
the main purpose of 'onecontext' is to allow outgoing access to the
PSTN.
[onecontext]
include = createmenu //creating an IVR menu
include = createconf //creating a conf call
etc
include = default //used for voicemail
[createmenu]
;does something
[createconf]
;does something
;outgoing calls - main purpose of onecontext
exten = _X.,1,Dial(SIP/[EMAIL PROTECTED])
exten = _X.,2,Hangup
[default]
;mailbox for 2092 and other users
Now this is where the problems start! For incoming calls I tried to
do include = incomingpstn in 'onecontext' which I thought would
call a new context called 'incomingpstn' which would have an entry
for the dummy user. i.e.
[incomingpstn]
exten = 2093,1,Wait(1)
exten = 2093,n,Background(MainMenu)
exten = 1,1,Goto(InternalExtension,2093,1)//directs to another
context called Internal Extension
I also changed the [provider-in] for context=incomingpstn in my
sip.conf. However this didn't work and I kept getting directed to the
voicemail of my pstn provider. The ONLY way I could get the incoming
calls working was to add the contents of the 'incomingpstn' context
to the default context i.e.
[default]
exten = 2093,1,Wait(1)
exten = 2093,n,Background(MainMenu)
exten = 1,1,Goto(InternalExtension,2093,1)//directs to another
context called Internal Extension
With this I can hear the MainMenu when I dial my DDI but I can't seem
to interrupt to divert to another submenu. In the testing that I have
done the user that is making the call is 2092 registered with SER. If
I change the context of 2092 directly in sip.conf to incomingpstn,
then I can hear the menu and interrupt to go to the submenu. But
obviously then I don't have access to the other features in Asterisk.
The point is that I'm stumped as to why it only works in the default
context and if this is the case how do I get it to call the submenu.
This is what comes up on my asterisk console:
-- Executing Dial (SIP/2092-2829, SIP/[EMAIL PROTECTED]) in
new stack
-- Called [EMAIL PROTECTED]
-- Playing 'MainMenu' (language 'en')
-- other messages (not relevant I think)
== Spawn extension (outgoing, 021123456, 1) exited non-zero on
'SIP/2092-5837'
== Spawn extension (default, 2093, 2) exited non zero etc etc
I'm very stuck on this and can't figure it
RE: [Asterisk-Users] Cisco 801 and rcapi
On Mon, 9 Jan 2006, James Harper wrote: I would suggest extend the libcapi20. I already did such an extension to libcapi20 to support the bintec remote-capi. This means with that libcapi20, each program (including chan_capi) can do remote-capi without any change... The more I look, the more I think that the bintec protocol might be the one required to talk to the Cisco anyway. Do you have those patches somewhere? I have placed the patched libcapi20 sources (libcapi20.tgz) on the public ftp server ftp://isdn4linux.org/pub/capi4linux It works pretty good with the rcapid (bintec-router emulator) on the remote side. I never tested it with a real bintec router, because I don't have one. Maybe it will not work with the real hardware, because the authentification is not implemented yet. This libcapi20 support normal /dev/capi20 and the remote version, just create a file ~/.capi20rc to set the remote station. See README. You can also use one Linux Server running CAPI cards with rcapid and have your Asterisk/OpenPBX with chan_capi on another maschine... 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] Asterisk vs 3COM
Would anyone recommend a medium size company choosing Asterisk over 3COM - Original Message - From: Kerry Garrison [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Saturday, January 07, 2006 10:23 PM Subject: RE: [Asterisk-Users] Asterisk Jobs If you try to compare Asterisk to other PBX's TODAY, Asterisk is running somewhere close to 0%. Its simply too new still as most companies didn't even begin taking a look until version 1.0 and even more with 1.2. Of course this will change over time. We are selling several systems a month right now. So if you are looking at getting a job today, it may be a little rough, but if you spend the next year honing your Asterisk skills more and more positions will open up. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Saturday, January 07, 2006 3:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk Jobs I'm curious why the number of jobs out there requiring Asterisk seems to be pretty low. After looking around dice, monster, careerbuilder etc, I was surprised to find no more than 3-4 employment opportunities with Asterisk throughout the US. Is it really that low? There seems to be a job of opportunities for Cisco and other vendors solutions (duh... GUI's are good... duh). I wonder if demand will increase, or am I just looking in the wrong places? - Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] new AMPortal and Asterisk debs
Tzafrir Cohen wrote: On Sun, Jan 08, 2006 at 06:16:16PM -0800, Mike Fedyk wrote: Tzafrir Cohen wrote: Experimental: Asterisk 1.2: At the moment they are not that experimental anymore and should be ready for use, but are not well-tested yet. To use it, define both sources: deb http://rapid.dotsrc.org/ experimental/ How does this compare with Asterisk 1.2.1.dfsg-1 that is in etch/testing and 1.2.1.dfsg-3 that is in sid/unstable? Testing (Etch) is slightly behind. It is generally in line with the packages in Sid. Sort of. Actually ff you compare the changelog you'll find some striking similarities. In fact, it is based on the current version in the pkg-voip svn than to the current version. However it is built for Sarge (Stable). So if you have Sarge installed, you won't have to upgrade libc6/pgsql/pwdlib/whatever to use it. I try to commit most of the relevant changes and fixes to the main Debian package, so if you have Sid/Etch, you'll end up getting basically the same packages (only with a more changing base system...) . I didn't know you are a co-maintainer until now since I just checked. How much longer before this is on backports.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: [Asterisk-Users] Asterisk vs 3COM
Small, medium and large are relative. What do you want it to do, and why do you want to change your phone system? With the right talent, (consultant or in-house) Asterisk can be used in most situations. With that no more details, then a simple answer will have to suffice. Most likely yes. Dakota wrote: Would anyone recommend a medium size company choosing Asterisk over 3COM - Original Message - From: Kerry Garrison [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Saturday, January 07, 2006 10:23 PM Subject: RE: [Asterisk-Users] Asterisk Jobs If you try to compare Asterisk to other PBX's TODAY, Asterisk is running somewhere close to 0%. Its simply too new still as most companies didn't even begin taking a look until version 1.0 and even more with 1.2. Of course this will change over time. We are selling several systems a month right now. So if you are looking at getting a job today, it may be a little rough, but if you spend the next year honing your Asterisk skills more and more positions will open up. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Saturday, January 07, 2006 3:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk Jobs I'm curious why the number of jobs out there requiring Asterisk seems to be pretty low. After looking around dice, monster, careerbuilder etc, I was surprised to find no more than 3-4 employment opportunities with Asterisk throughout the US. Is it really that low? There seems to be a job of opportunities for Cisco and other vendors solutions (duh... GUI's are good... duh). I wonder if demand will increase, or am I just looking in the wrong places? - Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Re: using a Gigaset SX440isdn on a Diva 4BRI ?
On Fri, Dec 30, 2005 at 10:23:26PM +0100, Armin Schindler wrote: On Fri, 30 Dec 2005, Louis-David Mitterrand wrote: Hello, I just received a couple SX440isdn phones and was wondering if they can be plugged into a Diva 4BRI port in NT mode and work with asterisk+chan_capi? Yes, I don't know any reason why it shouldn't work. I know they probably work fine with mutliHFC cards with either bristuff of chan_misdn but since I have some spare Divas, I was curious about their potential as phone ports. The Diva's 3 and 4 ports are already set to NT mode at boot time: /sbin/divactrl load -SeparateConfig -c 1 -f ETSI -f1 ETSI -f2 ETSI -u2 -x2 -f3 ETSI -u3 -x3 looks good. And I think the capi.conf (using Armin's 0.6.1 version) looks OK: [DIVA2] ntmode=yes isdnmode=ptp incomingmsn=* controller=4 group=3 accountcode=diva context=international echosquelch=0 echocancel=no devices=1 isdnmode=ptp is wrong for chan_capi 0.6, use isdnmode=did But when I plug the phone into port 3 or 4 no led lights up, even with a Y plug and when dialing I get a busy. Before digging to deep, I am looking for some info on the feasability of that setup. What type of cable did you use? You need to use a crossed cable with 100 Ohm termination. Hello Armin, I am now using a cross cable and the green led lights up on the Diva port when plugging the phone in. When dialing from the phone I get no debug or trace at the asterisk console, only a not possible message on the phone display and busy tone. Is there some configuration to do on the phone itself? When dialing from asterisk I get this: CAPI Debugging Enabled -- Executing NoOp(SIP/0146472130-f4c2, ) in new stack -- Executing Queue(SIP/0146472130-f4c2, accueil|rnt|||5) in new stack data = DIVA2/2 parsed dialstring: 'DIVA2' 'NULL' '2' '' capi request for interface 'DIVA2' parsed dialstring: 'DIVA2' 'NULL' '2' '' == DIVA2: Call CAPI/DIVA2/2-5 (pres=0x00, ton=0x00) CONNECT_REQ ID=002 #0x0008 LEN=0054 Controller/PLCI/NCCI= 0x4 CIPValue= 0x1 CalledPartyNumber = 802 CallingPartyNumber = 00 800146472130 CalledPartySubaddress = default CallingPartySubaddress = default BProtocol B1protocol = 0x1 B2protocol = 0x1 B3protocol = 0x0 B1configuration= default B2configuration= default B3configuration= default GlobalConfiguration= default BC = default LLC = default HLC = default AdditionalInfo BChannelinformation= 00 00 Keypadfacility = default Useruserdata = default Facilitydataarray = default SendingComplete= default -- Called CAPI/DIVA2/2 -- Called SIP/0146472152 -- Called SIP/0146472131 CONNECT_CONF ID=002 #0x0008 LEN=0014 Controller/PLCI/NCCI= 0x304 Info= 0x0 -- DIVA2: received CONNECT_CONF PLCI = 0x304 CAPI devicestate requested for DIVA2/2 CAPI devicestate requested for DIVA2/2 -- SIP/0146472152-487d is ringing -- Incoming call: Got SIP response 500 Internal Server Error back from 10.0.3.138 -- SIP/0146472131-56e8 is ringing DISCONNECT_IND ID=002 #0x001f LEN=0014 Controller/PLCI/NCCI= 0x304 Reason = 0x3302 DISCONNECT_RESP ID=002 #0x001f LEN=0012 Controller/PLCI/NCCI= 0x304 CAPI INFO 0x3302: Protocol error layer 2 == DIVA2: CAPI Hangingup == DIVA2: Interface cleanup PLCI=0x304 CAPI devicestate requested for DIVA2/2 CAPI devicestate requested for DIVA2/2 -- Incoming call: Got SIP response 500 Internal Server Error back from 10.0.3.138 == Spawn extension (admin, 2131, 2) exited non-zero on 'SIP/0146472130-f4c2' This queue includes: member = CAPI/DIVA2/2 I'm not quite sure I got the syntax right for isdn phones in NT mode. By the way, when two phones are plugged in a Diva port with a Y-plug how does one dial each phone separately? Do they have an address? -- [EMAIL
[Asterisk-Users] Wake-Up Call
I have setup wake up call in * following those instructions http://www.voip-info.org/wiki/view/Asterisk+tips+Wake-Up+Call+PHP and it works fine. Now I have few questions. - When I arrange wake up call, does it call me only that day or I can set it up for whoole week? - Can I set it up for some other extension or only for one I'm calling? - Can this AM, PM be in 24h format? That is all (for now :)). -- Tomislav Parcina [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
[Asterisk-Users] SNOM Hotdesking...
Hi I have been trying to get SNOM (320,360) and hotdesking working with asterisk. I can get it working fine with SER but it fails with asterisk unless I have no SIP password/secret in sip.conf This is how it works with SER, 1. reset phone (removes accounts) 2. phone prompts for username and sip server 3. phone sends register to SER 4. SER sends a 401 unauthorized 5. phone sends register with Digest (but no password) 6. SER sends a 401 unauthorized 7. phone prompts for a password 8. phone sends register with Digest (with correct password) with asterisk, 1. reset phone (removes accounts) 2. phone prompts for username and sip server 3. phone sends register to ASTERISK 4. ASTERISK sends a 401 unauthorized 5. phone sends register with Digest (but no password) 6. ASTERISK sends 403 Forbidden 7. phone gives up.. I raised this with SNOM and they say it is purely an asterisk problem and it needs to be fixed (asterisk that is). If asterisk sent a 401 instead of a 403 the phone would work fine and we would all be happy. ___ --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: Call logging
In article 6A1C243A7E2E824293FABC3042045790930851 @dtw_localmail.strtrade.com, [EMAIL PROTECTED] says... Hello all, is anyone aware of any open source call accounting software for Asterisk? Something that can parse out Asterisk's call detail records and generate on-demand reports? Check out Asterisk-Stat: CDR Analyser http://areski.net/asterisk-stat-v2/about.php?s=0 -- Tomislav Parcina [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] SNOM Hotdesking...
Hi, I raised this with SNOM and they say it is purely an asterisk problem and it needs to be fixed (asterisk that is). If asterisk sent a 401 instead of a 403 the phone would work fine and we would all be happy. Here you can find a patch that will fix it: http://bugs.digium.com/bug_view_page.php?bug_id=6035 Maik Schmitt ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call files, fax
Hello, I have a couple of questions: 1) Before heading off for a bit of vacation, I was having a wierd problem where I was getting more than one call per callfile placed in the outgoing/ spool. I describe it here: http://forums.digium.com/viewtopic.php?t=3455 so far, so good - it's not doing it right now, but what might cause that? 2) app_txfax I need to know if a fax has gone through or not. My reading of txfax seems to indicate that it basically just fails, rather than giving me anything I can work with to try and fail gracefully (letting the user know that things didn't go well). Is that indeed correct? I don't know Asterisk that well, so I may be completely off base:-) What would be the best way to make it interact better with the dial plan so that one could detect if it fails and act accordingly? Set a variable? 3) I'm working on a small, simple email-fax system. Just out of curiosity, what else is out there for Asterisk? I found AsterFax, but it looks a little bit hairy to set up... Thanks! -- Webster srl Sede legale: Via del Seminario, 3 35122 Padova Sede operativa: Via S. Breda, 28 35010 Limena (PD) Tel. +39 049 652527 - Fax +39 049 655297 Email: [EMAIL PROTECTED] Visita www.libreriauniversitaria.it ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SNOM Hotdesking...
Ah cool, thanks ill look at it. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Maik Schmitt Sent: 09 January 2006 11:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SNOM Hotdesking... Hi, I raised this with SNOM and they say it is purely an asterisk problem and it needs to be fixed (asterisk that is). If asterisk sent a 401 instead of a 403 the phone would work fine and we would all be happy. Here you can find a patch that will fix it: http://bugs.digium.com/bug_view_page.php?bug_id=6035 Maik Schmitt ___ --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] Wake-Up Call
On Mon, 2006-01-09 at 12:07 +0100, Tomislav Parcina wrote: I have setup wake up call in * following those instructions http://www.voip-info.org/wiki/view/Asterisk+tips+Wake-Up+Call+PHP and it works fine. Now I have few questions. - When I arrange wake up call, does it call me only that day or I can set it up for whoole week? - Can I set it up for some other extension or only for one I'm calling? - Can this AM, PM be in 24h format? That is all (for now :)). That particular script appears to only schedule for the next 24 hours. It could do more but it doesnt. I was going to write a php one that doesnt work quite this way. Instead I was going to take advantage of features of the queue app so you dont need a seperate cron job, namely setting the time of the queue file to the time you want the wake up call. I was also going to add in features to record a custom message, and other such goodies. They arent complex features, but I think would make it nicer. But this is low priority for me right now. Between the Sac AUG and ETEL speaking engagements this month along with regular work I am unsure that I will have time until feburary. I hadnt thought about recurring ones, that would be better handled via a crontab type setup I think than creating a ton of queue files. You could easily do this, just a matter of storing who, when, how frequently, and then creating the queue files on time. If you are using the one I think you are then if you enter it in 24 hour format and the time is 12 it can tell that you mean 24 hour format, hwoever it cant tell teh difference bewteen 12hr and 24hr if the time is 12 so it asks. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group 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] dual IP connections
Hi all, I would like to know if there is a solution to this question. Scenario: Two asterisk servers connected across the Intenet ( in SIP or IAX mode, no matter) with both of them having static ip addresses Then I add a second link (with another provider), with another NIC at both side, and again both of them having static ip addresses. Is there a way to tell asterisk to use both of these link, i.e. doing a load balancing ? Or just better (in my case) to use only one link, and to use the second link as a backup link in the event the first link went down ? thanks in advance, Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dual IP connections
Hi, Two asterisk servers connected across the Intenet ( in SIP or IAX mode, no matter) with both of them having static ip addresses Then I add a second link (with another provider), with another NIC at both side, and again both of them having static ip addresses. Is there a way to tell asterisk to use both of these link, i.e. doing a load balancing ? Or just better (in my case) to use only one link, and to use the second link as a backup link in the event the first link went down ? this is a routing problem, not an asterisk one. you can do some ip policy based routing , but imho if you implement this is better to have another box between the * one and the 2 isp links that do the load balance, or the switch to the bkp isp if the first one goes down. my idea is: asterisk box(one nic) - router(3 nics) - isp1 - isp2 the on router you can play with ip policy based routing or simply failover routing. cya, Matteo. -- Come to visit us @ CeBit 2006 From 9 to 15 March 2006 Hall 13 Stand no. E25/1 Matteo Brancaleoni System Administrator Tel +39.02.70633354 Sip [EMAIL PROTECTED] Iax2 [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
[Asterisk-Users] Is it Wildcard 406
Hello! After many troubles, I have received my Wildcard 406. There is a label on antistatic bag stating that this is 406. The card itself is marked as 405. Kernel modules shows in dmesg that card is 405. Is 406 the same as 405 with additional board installed? ___ --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] rcapi quality (was: Cisco 801 and rcapi)
Hi Armin, You can also use one Linux Server running CAPI cards with rcapid and have your Asterisk/OpenPBX with chan_capi on another maschine... Did you ever try something like that? What kind of implication had the remote CAPI with regards to sound quality? -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] rcapi quality (was: Cisco 801 and rcapi)
On Mon, 9 Jan 2006, Peer Oliver Schmidt wrote: Hi Armin, You can also use one Linux Server running CAPI cards with rcapid and have your Asterisk/OpenPBX with chan_capi on another maschine... Did you ever try something like that? I just tried it. But I never really used it longer. What kind of implication had the remote CAPI with regards to sound quality? I think that depends on the connection between the rcapi-server and the libcapi20 client... 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] Asterisk initialization
That's great... I didn't know about the persistentagents features! I'll test it asap! Thank you Dov - Original Message - From: Alexander Lopez To: Dov Bigio ; Asterisk Users Mailing List - Non-Commercial Discussion Sent: Saturday, January 07, 2006 5:16 PM Subject: RE: [Asterisk-Users] Asterisk initialization Do not know what version you are running, But there are a few ways to do this. There is a persistant setting: from agents.conf ;; Define whether callbacklogins should be stored in astdb for; persistence. Persistent logins will be reloaded after; Asterisk restarts.;persistentagents=yes If you want to handle it outside of Asterisk via an AGI you can have your AGI execute: AgentCallbackLogin([AgentNo][|[options][|[EMAIL PROTECTED]): this is providing that you have the information saved in your DB. Personal Opinion: Use the builtin features with the persistentagents options and use the php script in the contribs directory to see who is on. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dov BigioSent: Friday, January 06, 2006 4:24 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Asterisk initialization Hi, I am doing an AGI that logs to a database every Agent login/logoff. My idea is to be able to go to this database and check which agents where logged so that I can force their login in case Asterisk goes down for some reason. The problem is that I would need to reload their status from this AGI when Asterisk initializes. Is there a way to do this? One idea I had was to make safe_asterisk to generate a .call file that calls and extension that would call the AGI to log all the agents back on. Is there another way of running an AGI on initialization? 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
Re: [Asterisk-Users] Polycom 501 netboot not working.
Are you sure you have the FTP server's IP address set correctly in the phone's configuration? On Thu, Jan 05, 2006 at 05:17:41PM -0500, Ken D'Ambrosio wrote: Anthony Rodgers wrote: Is the mac-address.cfg file name in lower case? Yeah, it is. Hell -- I've cut-and-pasted the filename from the below logfile, and been able to FTP it just fine. I've run an ethereal dump, and it never even -asks- the server for the file, so I'm kind of confused there. I've reset the phone with 4-6-8-* keys, but same thing. I'm tempted to try another phone, and see if I get anywhere. But before I -kill- another phone, I thought I'd ask if anyone else has seen this or anything like it... -Ken On Jan 5, 2006, at 1:37 PM, Ken D'Ambrosio wrote: When I try to boot my 501, it runs through the usual stuff, then stops with Config file error Error is 0x4020 and then reboots. The log on the FTP server shows: 0105164151|app1 |3|00|Bootline: ircaIP 0105164155|cfg |3|00|Image bootrom.ld has not changed. 0105164159|cfg |3|00|0004f202f803.cfg could not be downloaded, getting next file. 0105164206|cfg |3|00|Image sip.ld has not changed. 0105164237|app1 |4|00|Loaded application sip.ld successfully, errors 0x0. 0105164237|app1 |6|00|Uploading boot log, time is THU JAN 05 16:42:38 2006 I can't figure out why it can't download the cfg file -- the permissions are right, etc. I can FTP all the files as PlcmSpIp (with PlcmSpIp as the password) just fine. It -does- try to d/l the .cfg file, but appears to ignore it, even when I give it extension-specific config info (gives the same error). Any ideas? I'm afraid to try to provision my other phones, for fear of winding up in the same spot. Thanks, -Ken ___ --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 --- [This E-mail scanned for viruses by Declude Virus] -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --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] snom programmable buttons
Hi, I want to pick up a call with the snom's programmable buttons(snom190 -SIP 3.60x, snom360-SIP 4.1) with asterisk server (v 1.2.0), I tried with the option 'Destination' and when the incoming call arrive to another snom phone the button blinking. In this way I can only pick down it pressing the blinking button. The solution is call the *8 or parcking the call but my pbroblem is when the incoming call are 2 or 3 and I would press a programmable button to pick up the calls. Is possible have configured asterisk and the snom phone with the function shared line? Are there solutions ? Thanks Luca L. [cfh] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is it Wildcard 406
On Monday 09 January 2006 07:32, Dmitry Ivanov wrote: After many troubles, I have received my Wildcard 406. There is a label on antistatic bag stating that this is 406. The card itself is marked as 405. Kernel modules shows in dmesg that card is 405. Is 406 the same as 405 with additional board installed? The Digium TE406 is the TE405 with the optional VPM (voice processing module) installed. -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] Cisco 801 and rcapi
The more I look, the more I think that the bintec protocol might be the one required to talk to the Cisco anyway. Do you have those patches somewhere? I have placed the patched libcapi20 sources (libcapi20.tgz) on the public ftp server ftp://isdn4linux.org/pub/capi4linux Thanks! It works pretty good with the rcapid (bintec-router emulator) on the remote side. I never tested it with a real bintec router, because I don't have one. Maybe it will not work with the real hardware, because the authentification is not implemented yet. This libcapi20 support normal /dev/capi20 and the remote version, just create a file ~/.capi20rc to set the remote station. See README. I've had a quick look at it... can you use local (kernel) capi devices and remote devices on the one machine? You can also use one Linux Server running CAPI cards with rcapid and have your Asterisk/OpenPBX with chan_capi on another maschine... I assume you've tried that configuration then... can you comment on the performance and reliability? How does the system as a whole cope if the rcapi server goes down? Thanks James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spandsp, rxfax, TDM400/zaptel, missed frames, any help?
On 1/9/06, Rich Adamson [EMAIL PROTECTED] wrote: Sorry in advance if this is a FAQ... I've got a working Asterisk setup based on [EMAIL PROTECTED] 2.2. I have a TDM400 card with 2 FXS and 2 FXO ports; PSTN connections come in via the TDM card. I haven't been able to get inbound fax with spandsp and rxfax to work. Occasionally an all-text fax will come in, though it's usually badly corrupted, but in most cases, it would appear that the call is terminated without successful transmission of the fax. I get logs that look what's included below. From reading the list, it looks like this is caused by the TDM card missing frames. Does that sound correct? If so, is there any relief in sight? Its been a problem since the card came out a couple of years ago. So, no it does not appear there is any relief in sight. Sigh. What a disappointment! Are there any other options for home users to receive faxes over the PSTN through *? Is anyone working on an alternative to the zaptel driver that might fix this issue? Ben ___ --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] Lost my Zap's
Hi can anyone help. I just updated my CentOS and ran the command rebuild_zaptel and genzaptelconf with a Reboot in between each step. Now I have no Zaptel devices (I used to have 3 FXO X100P Cards) Summary of what happens below: (Zaptel.conf contains no card info after running this command.) Many thanks in advance, Richard STOPPING ASTERISK Disconnected from Asterisk server Asterisk Stopped STOPPING FOP SERVER FOP Server Stopped Generating '/etc/zaptel.conf' Generating '/etc/asterisk/zapata-auto.conf' Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected STOPPING ASTERISK STOPPING FOP SERVER Unloading zaptel hardware drivers:. Removing zaptel module:[ OK ] Loading zaptel framework: [ OK ] Waiting for zap to come online...Error: missing /dev/zap! SETTING FILE PERMISSIONS chown: cannot access `/dev/zap': No such file or directory Permissions OK STARTING ASTERISK Asterisk Started STARTING FOP SERVER FOP Server Started Chan Extension Context Language MusicOnHold pseudofrom-pstn en Verbosity is at least 3 attachment: winmail.dat___ --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] controlling SIP subscriptions from SNOM phones
On Saturday 07 January 2006 02:30, Philipp von Klitzing wrote: Hi! Now, one user, not the receptionist, has gone in and set his personal numbers to these function keys thinking that DESTINATION meant setting a number to dial out. So now I have a ton of SIP SUBSCRIBE messages for his numbers. Indeed this situation is not ideal. The first thing to do in my opinion is ask SNOM to provide a new type of DESTINATION option that does not issue subscribes. This is already available with firmware release 5 for snom320/360. This new type is named speed dial. Secondly you need to be aware that if Asterisk doesn't find a matching hint in the subscribecontext it will look check in the default context! This is, btw, one good reason to not have your local phones in the default context unless you want everyone out there to be able to subscribe to everyone else... Finally: Have you tried to create a new context, set the user's subscribecontext to this and do a _.,hint,SIP/DoesNotExist or smth similar within that context (and nothing else)? Cheers, Philipp Regards, Sven ___ --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 -- --- See our FAQs at: http://www.snom.com/faq0.html?L=1 Whitepapers at: http://www.snom.com/white_papers.html --- snom technology AG Gradestraße 46 D-12347 Berlin Sven Fischer fax +49 30 39833111 mailto:[EMAIL PROTECTED] http://www.snom.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] Re: Transfer
Tomislav Parcina schrieb: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I am aware of the possibility to add the option t or T to dial, so #33 transfers the call to extension 33. It needs to be deined in feautres.conf file. So when you dial #1 you'll hear transfer and than you enter extension. Is there any use of this command in the dialplan? If I want to redirekt a call because of the choices of a caller goto() or dial() does the job. In dialplan you need only to enter t and/or T. So, once again, why do we need the command Transfer ?? I still didn't got any suggestion of a useful situation where i can execute this command to do something useful. Greetings, Tobias ___ --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] Cisco 801 and rcapi
On Tue, 10 Jan 2006, James Harper wrote: The more I look, the more I think that the bintec protocol might be the one required to talk to the Cisco anyway. Do you have those patches somewhere? I have placed the patched libcapi20 sources (libcapi20.tgz) on the public ftp server ftp://isdn4linux.org/pub/capi4linux Thanks! It works pretty good with the rcapid (bintec-router emulator) on the remote side. I never tested it with a real bintec router, because I don't have one. Maybe it will not work with the real hardware, because the authentification is not implemented yet. This libcapi20 support normal /dev/capi20 and the remote version, just create a file ~/.capi20rc to set the remote station. See README. I've had a quick look at it... can you use local (kernel) capi devices and remote devices on the one machine? It is possible, but not with one and the same application. Currently the setting for remote-capi is done per-user. Of course, this can be changed. But if you want to merge local and remote CAPI, then it is not possible at the moment. This would need enhancements. You can also use one Linux Server running CAPI cards with rcapid and have your Asterisk/OpenPBX with chan_capi on another maschine... I assume you've tried that configuration then... I have tried it, but I didn't really test it with Asterisk yet. I use it, but with other applications like a standalone voicemailbox and capifax. can you comment on the performance and reliability? The connection is TCP, so no problem with reliability with that. But the perfomance depends on the IP connection. How does the system as a whole cope if the rcapi server goes down? I didn't test this yet. But I assume the client libcapi will signal error codes when the connection is lost. I'm sure, if someone really wants to use this, some enhancements must be done for the faulty cases. But it should be easy. 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
[Asterisk-Users] Snom Idleline XML
Anyone got the screen xml function to work yet? i've setup an URL in my line 1 (the only line I use) but i don't even see a GET request to my webserver. Kind regards, Erik ___ --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: dual IP connections
Have you checked http://www.voip-info.org/wiki/view/Asterisk+High+Availability+Solutions Regards, Evert [EMAIL PROTECTED] wrote: Hi all, I would like to know if there is a solution to this question. Scenario: Two asterisk servers connected across the Intenet ( in SIP or IAX mode, no matter) with both of them having static ip addresses Then I add a second link (with another provider), with another NIC at both side, and again both of them having static ip addresses. Is there a way to tell asterisk to use both of these link, i.e. doing a load balancing ? Or just better (in my case) to use only one link, and to use the second link as a backup link in the event the first link went down ? thanks in advance, Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] ISDN beronet: cannot send digits during outbound calls
Hi all, problem solved! The parameter /s at the end of Dial string command was necessary. Giorgio Incantalupo gincantalupo wrote: Hi, we are trying a beronet ISDN card with asterisk 1.2 on debian sarge distro. Everything seems fine except for outbound calls: it seems we cannot send outbound digits so we cannot use phone digits to use ivr menus. I followed beronet dinstallation document. Is there some parameter missing to add to configuration files? TIA Giorgio Incantalupo ___ --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] Lost my Zap's
Richard, This also happened to me over the weekend. What happened to me was yum updatd two files found in /etc/udev/permissions.d/ and the other in /etc/udev/rules.d/ Yum makes backup copies of each of these files. All you need to do is copy the missing lines from both files and paste them back into the new ones. Then try rebuilding zaptel and reconfiguring. This worked for me... - Jason -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mr Asterisk Sent: Monday, January 09, 2006 8:33 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Lost my Zap's Hi can anyone help. I just updated my CentOS and ran the command rebuild_zaptel and genzaptelconf with a Reboot in between each step. Now I have no Zaptel devices (I used to have 3 FXO X100P Cards) Summary of what happens below: (Zaptel.conf contains no card info after running this command.) Many thanks in advance, Richard STOPPING ASTERISK Disconnected from Asterisk server Asterisk Stopped STOPPING FOP SERVER FOP Server Stopped Generating '/etc/zaptel.conf' Generating '/etc/asterisk/zapata-auto.conf' Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected STOPPING ASTERISK STOPPING FOP SERVER Unloading zaptel hardware drivers:. Removing zaptel module:[ OK ] Loading zaptel framework: [ OK ] Waiting for zap to come online...Error: missing /dev/zap! SETTING FILE PERMISSIONS chown: cannot access `/dev/zap': No such file or directory Permissions OK STARTING ASTERISK Asterisk Started STARTING FOP SERVER FOP Server Started Chan Extension Context Language MusicOnHold pseudofrom-pstn en Verbosity is at least 3 ___ --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] Recording Calls at the phone
Starting and stopping the recording is based off of the message taking software which knows when I call is going on. They do make recording devices that go in between the headset and phone, but they take batteries. I can't really have a recording device running off batteries in a call center. I think I'm just going to get SIP to FXO adapters and run the recording control off the FXO port. Michael Sampson Information Systems Manager Customer Contact Services [EMAIL PROTECTED] 952-936-4000 Ioan Indreias wrote: A (too) simple sollution to your problem is to take the analog audio from your IP phone using a module atached between the curly handset cord and the base unit of the IP phone - like http://www.quasarelectronics.com/tre156.htm So, basically you need to change the old "RJ11 - 1/8 inch recording - RJ11" system you have used to a new one with "RJ10 - 1/8 inch recording - RJ10". Sure, this solution works only if the handeset it is attached through a RJ10 port to the handset. I do not know exactly how your software will deal with this change as there should be a mechnism to start stop recording based on the audio level injected into PC's audio card (mic port). Hope it helps. Ioan Indreias Modulo Consulting - http://www.modulo.ro I'm not really trying to monitor anything on the asterisk box at all. I guess this is more of an SIP phone question. Really all I need is to get the audio from an SIP phone, both the caller and callie, to a 1/8th inch stereo jack that I can plug into a mic input. Michael Sampson Information Systems Manager Customer Contact Services [EMAIL PROTECTED] 952-936-4000 Douglas Garstang wrote: On Demand-monitoring? If your referring to monitoring specific agents calls, I'm still trying to work out how to do that. You can either monitor all calls for a queue, or all calls for all agents, but not all calls for a specific agent. I tried to use the Monitor() command on it's own to start recording when an agent receives a call, but that does not appear to work. -Original Message- From: Francesco Peeters (Asterisk) [mailto:[EMAIL PROTECTED]] Sent: Friday, January 06, 2006 7:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Recording Calls at the phone On Fri, January 6, 2006 15:37, Michael Sampson said: I work for a call center and we are looking at using asterisk to have our operators take calls. Our message taking software records all the calls on the operators computers. Right now we use these recording controls from radio shack that plug in between the wall jack and the phone and plug in via a 1/8 inch stereo connector to the mic input on the computer. If I buy an IP phone I can't do that. I could get an FXO adapter and regular phones, but I'm looking to get as little equipment as possible. Radio shack makes a recording control that plugs in to a 2.5 mm headset jack, but it takes batteries so thats not going to work Does anyone else do something similar? Does anyone have any ideas about what producs/setup would work for this. Asterisk has a built in monitoring system. You can chose to do Always, Never or On Demand monitoring, depending on your setup and dialplan http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Monitor Good luck! ___ --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] call files, fax
David N. Welton [EMAIL PROTECTED] wrote: 2) app_txfax I need to know if a fax has gone through or not. My reading of txfax seems to indicate that it basically just fails, rather than giving me anything I can work with to try and fail gracefully (letting the user know that things didn't go well). Is that indeed correct? I don't know Asterisk that well, so I may be completely off base:-) What would be the best way to make it interact better with the dial plan so that one could detect if it fails and act accordingly? Set a variable? 3) I'm working on a small, simple email-fax system. Just out of curiosity, what else is out there for Asterisk? I found AsterFax, but it looks a little bit hairy to set up... You really should consider HylaFAX - www.hylafax.org. It has what you're missing - a fully featured queue manager / scheduler that takes care of retries for you, and notifies the sender of any failures encountered. It can be integrated with Asterisk via analog or digital lines, or by using a software-based modem such as IAXmodem. -Darren ___ --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] Agents in 1.2.1
Hi, I've used Agents + Queues before with success, but I can't figure
out why this trivial setup is not functioning...
stage*CLI show agents
1306 (gdh) available at '[EMAIL PROTECTED]' (musiconhold is 'default')
1 agents configured [1 online , 0 offline]
and the internal context is simply:
[internal]
exten = _13XX,1,Dial(SIP/${EXTEN},,h)
Now, taking this line...
exten = 123454,1,Dial(SIP/1306)
(Legacy PBX On Zaptel interface dials 123454)
-- Starting simple switch on 'Zap/66-1'
-- Accepting overlap call from '1010' to '123454' on channel 0/4, span 3
-- Executing Dial(Zap/66-1, SIP/1306) in new stack
-- Called 1306
-- SIP/1306-f498 is ringing
-- Channel 0/4, span 3 got hangup request
== Spawn extension (fromaxxess, 123454, 1) exited non-zero on 'Zap/66-1'
-- Hungup 'Zap/66-1'
Great - the phone rings - hurrah! BUT... :O
exten = *11,1,AgentCallbackLogin(${CALLERIDNUM}||[EMAIL PROTECTED])
exten = 123455,1,Dial(Agent/1306)
(SIP phone 1306 dials *11)
-- Executing AgentCallbackLogin(SIP/1306-d752,
1306||[EMAIL PROTECTED]) in new stack
== Setting global variable 'AGENTBYCALLERID_1306' to '1306'
-- Playing 'agent-loginok' (language 'en')
== Callback Agent '1306' logged in on [EMAIL PROTECTED]
-- Playing 'vm-goodbye' (language 'en')
== Spawn extension (fromip, *11, 1) exited non-zero on 'SIP/1306-d752'
(Legacy PBX On Zaptel interface dials 123455)
-- Starting simple switch on 'Zap/66-1'
-- Accepting overlap call from '1010' to '123455' on channel 0/4, span 3
-- Executing Dial(Zap/66-1, Agent/1306) in new stack
== Everyone is busy/congested at this time (1:1/0/0)
-- Hungup 'Zap/66-1'
Why am I being told that 'everyone is busy' on this Agent, when it is
clearly 'available', and calling the SIP device directly does work?
I'm assuming it's because of something I'm doing wrong, but I can't see
what :(
gdh
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[Asterisk-Users] Dialtone detection help needed
I would like to know if anyone out there has a known and working solution in Asterisk 1.2.1 for dialtone detection. We currently use the Chanisavail command on Zap channels and then need dialtone detection after that. Please respond on or off list. v o i p 3 a t t a n i b b l e d o t n e t ___ --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: using a Gigaset SX440isdn on a Diva 4BRI ?
On Mon, 9 Jan 2006, Louis-David Mitterrand wrote: On Fri, Dec 30, 2005 at 10:23:26PM +0100, Armin Schindler wrote: On Fri, 30 Dec 2005, Louis-David Mitterrand wrote: Hello, I just received a couple SX440isdn phones and was wondering if they can be plugged into a Diva 4BRI port in NT mode and work with asterisk+chan_capi? Yes, I don't know any reason why it shouldn't work. I know they probably work fine with mutliHFC cards with either bristuff of chan_misdn but since I have some spare Divas, I was curious about their potential as phone ports. The Diva's 3 and 4 ports are already set to NT mode at boot time: /sbin/divactrl load -SeparateConfig -c 1 -f ETSI -f1 ETSI -f2 ETSI -u2 -x2 -f3 ETSI -u3 -x3 looks good. And I think the capi.conf (using Armin's 0.6.1 version) looks OK: [DIVA2] ntmode=yes isdnmode=ptp incomingmsn=* controller=4 group=3 accountcode=diva context=international echosquelch=0 echocancel=no devices=1 isdnmode=ptp is wrong for chan_capi 0.6, use isdnmode=did But when I plug the phone into port 3 or 4 no led lights up, even with a Y plug and when dialing I get a busy. Before digging to deep, I am looking for some info on the feasability of that setup. What type of cable did you use? You need to use a crossed cable with 100 Ohm termination. Hello Armin, I am now using a cross cable and the green led lights up on the Diva port when plugging the phone in. When dialing from the phone I get no debug or trace at the asterisk console, only a not possible message on the phone display and busy tone. Is there some configuration to do on the phone itself? No, the phone does not need special settings. Does the phone have own power supply? If not, then it will not work, because devices on the NT-mode card must provide own power. When dialing from asterisk I get this: CAPI Debugging Enabled -- Executing NoOp(SIP/0146472130-f4c2, ) in new stack -- Executing Queue(SIP/0146472130-f4c2, accueil|rnt|||5) in new stack data = DIVA2/2 parsed dialstring: 'DIVA2' 'NULL' '2' '' capi request for interface 'DIVA2' parsed dialstring: 'DIVA2' 'NULL' '2' '' == DIVA2: Call CAPI/DIVA2/2-5 (pres=0x00, ton=0x00) CONNECT_REQ ID=002 #0x0008 LEN=0054 Controller/PLCI/NCCI= 0x4 CIPValue= 0x1 CalledPartyNumber = 802 CallingPartyNumber = 00 800146472130 CalledPartySubaddress = default CallingPartySubaddress = default BProtocol B1protocol = 0x1 B2protocol = 0x1 B3protocol = 0x0 B1configuration= default B2configuration= default B3configuration= default GlobalConfiguration= default BC = default LLC = default HLC = default AdditionalInfo BChannelinformation= 00 00 Keypadfacility = default Useruserdata = default Facilitydataarray = default SendingComplete= default -- Called CAPI/DIVA2/2 -- Called SIP/0146472152 -- Called SIP/0146472131 CONNECT_CONF ID=002 #0x0008 LEN=0014 Controller/PLCI/NCCI= 0x304 Info= 0x0 -- DIVA2: received CONNECT_CONF PLCI = 0x304 CAPI devicestate requested for DIVA2/2 CAPI devicestate requested for DIVA2/2 -- SIP/0146472152-487d is ringing -- Incoming call: Got SIP response 500 Internal Server Error back from 10.0.3.138 -- SIP/0146472131-56e8 is ringing DISCONNECT_IND ID=002 #0x001f LEN=0014 Controller/PLCI/NCCI= 0x304 Reason = 0x3302 DISCONNECT_RESP ID=002 #0x001f LEN=0012 Controller/PLCI/NCCI= 0x304 CAPI INFO 0x3302: Protocol error layer 2 The line/protocol still seem to be set up wrong. Can you provide a mlog for in/out calls? I'm not quite sure I got the syntax right for isdn phones in NT mode. The syntax for dial() is the same like for TE-mode. By the way, when two phones are plugged in a Diva port with a Y-plug how does one dial each phone separately? I never tested PtMP with NT-mode and I don't know if this will work. Do they have an address? They have
Re: [Asterisk-Users] Help Connecting server districts
Alexander Lopez wrote: I would incoparate dundi, After using it I have fallen in love with it for distributed applications such as this. It makes configuration at first a bit steeper but it picks up monentum as your deploy. With Dundi you basicaly broadcast what extensions or numbers are served by your machine and using a set of keys (which negats having to configure a perr for every machine to create a mesh netowrk) Thanks for bringing this up Alexander :-) He is right... a private DUNDi network is the perfect solution for this sort of thing. No central administration, no need to update servers to be aware of what routes the other servers offer, etc. ___ --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] spandsp, rxfax, TDM400/zaptel, missed frames, any help?
Sorry in advance if this is a FAQ... I've got a working Asterisk setup based on [EMAIL PROTECTED] 2.2. I have a TDM400 card with 2 FXS and 2 FXO ports; PSTN connections come in via the TDM card. I haven't been able to get inbound fax with spandsp and rxfax to work. Occasionally an all-text fax will come in, though it's usually badly corrupted, but in most cases, it would appear that the call is terminated without successful transmission of the fax. I get logs that look what's included below. From reading the list, it looks like this is caused by the TDM card missing frames. Does that sound correct? If so, is there any relief in sight? Its been a problem since the card came out a couple of years ago. So, no it does not appear there is any relief in sight. Sigh. What a disappointment! Are there any other options for home users to receive faxes over the PSTN through *? Is anyone working on an alternative to the zaptel driver that might fix this issue? I'm certainly not the expert on this topic, but I believe the issue has to do with the pci bus and probably relates to the TigerJet chip used on the card. Until that's addressed, any analog modem use through the card will be marginal at best. (Same issue as with the older x100p card.) One alternative for low volume faxing is to use an external service. I've found those to be very economical and I receive all faxes in the form of a pdf file (much better for me in terms of a distributed office environment). No more costs associated with junk faxes, toner, paper, etc, etc. Another alternative is to maintain a single pstn analog line for outbound faxing, E911, and other such services. All of the above has been discussed many times on the list and should be available from the archives. ___ --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] Chanspy options in Asterisk Manager API
The syntax for the options in chanspy are not well documented. How do I use multiple options? I am using the Asterisk Manager API and am using ChanSpy(|q) but would like to include volume ChanSpy(|q,v3) ? Any insight would be appreciated. Dan Littlejohn www.littlejohnconsulting.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] Lost my Zap's
Hi can anyone help. I just updated my CentOS and ran the command rebuild_zaptel and genzaptelconf with a Reboot in between each step. Now I have no Zaptel devices (I used to have 3 FXO X100P Cards) Summary of what happens below: (Zaptel.conf contains no card info after running this command.) Sounds like the CentOS update over-wrote the changes that you need in the udev stuff. Check the udev readme in the zaptel source directory for what's needed. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Jobs
Fonality just received an influx of capital, you can read about it here. http://gigaom.com/2006/01/09/fonality/ Cory Andrews Purchasing Manager ++ VOIPSupply.com A Division of b2 Technologies 454 Sonwil Drive Buffalo, NY 14225 direct - 716.250.3402 mobile - 716.907.4054 email - [EMAIL PROTECTED] AIM - b2Cory - Original Message - From: Douglas Garstang [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, January 09, 2006 12:17 AM Subject: RE: [Asterisk-Users] Asterisk Jobs Who? me? :) -Original Message- From: Steve Totaro [mailto:[EMAIL PROTECTED] Sent: Sun 1/8/2006 8:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: RE: [Asterisk-Users] Asterisk Jobs I am not sure why you are looking for jobs doing Asterisk work when less than two weeks ago you were publicly bashing on the list. Steve Consulting is fine, as long as I'm working for someone else. Setting up my own company etc isn't really what I'm looking for. I don't want the risk. If there aren't actual companies offering good paying positions, then there's really no opportunities for me. -Original Message- From: Steven Kalcevich [mailto:[EMAIL PROTECTED] Sent: Sat 1/7/2006 7:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Asterisk Jobs I think it would be biggest is in consulting. The people that refuse or cant to pay for call manager or Avaya's one. Example asterisk sugarcrm.com they work together. Thats really good to sell. They arent in monster.ca they are banging on doors making $. Make a buch of pre setup asterisk configs that would be most popular make marketing material, dump on website. go in trade shows. Demo and make $ Steve kalcevich Douglas Garstang wrote: I'm curious why the number of jobs out there requiring Asterisk seems to be pretty low. After looking around dice, monster, careerbuilder etc, I was surprised to find no more than 3-4 employment opportunities with Asterisk throughout the US. Is it really that low? There seems to be a job of opportunities for Cisco and other vendors solutions (duh... GUI's are good... duh). I wonder if demand will increase, or am I just looking in the wrong places? - Doug. --- - ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk over 3Com
I would if the tech that sets it up knows exactly what he or she is doing. Regards, Dovid : "Dakota" [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Asterisk vs 3COMTo: "Asterisk Users Mailing List - Non-Commercial Discussion"asterisk-users@lists.digium.comMessage-ID: [EMAIL PROTECTED]Content-Type: text/plain; format=flowed; charset="iso-8859-1";reply-type=originalWould anyone recommend a medium size company choosing Asterisk over 3COM ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Jobs
Thanks Cory. Awesome... and their in LA too. They'll be hearing from me. :) -Original Message- From: Cory Andrews [mailto:[EMAIL PROTECTED] Sent: Monday, January 09, 2006 8:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Jobs Fonality just received an influx of capital, you can read about it here. http://gigaom.com/2006/01/09/fonality/ Cory Andrews Purchasing Manager ++ VOIPSupply.com A Division of b2 Technologies 454 Sonwil Drive Buffalo, NY 14225 direct - 716.250.3402 mobile - 716.907.4054 email - [EMAIL PROTECTED] AIM - b2Cory - Original Message - From: Douglas Garstang [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, January 09, 2006 12:17 AM Subject: RE: [Asterisk-Users] Asterisk Jobs Who? me? :) -Original Message- From: Steve Totaro [mailto:[EMAIL PROTECTED] Sent: Sun 1/8/2006 8:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: RE: [Asterisk-Users] Asterisk Jobs I am not sure why you are looking for jobs doing Asterisk work when less than two weeks ago you were publicly bashing on the list. Steve Consulting is fine, as long as I'm working for someone else. Setting up my own company etc isn't really what I'm looking for. I don't want the risk. If there aren't actual companies offering good paying positions, then there's really no opportunities for me. -Original Message- From: Steven Kalcevich [mailto:[EMAIL PROTECTED] Sent: Sat 1/7/2006 7:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Asterisk Jobs I think it would be biggest is in consulting. The people that refuse or cant to pay for call manager or Avaya's one. Example asterisk sugarcrm.com they work together. Thats really good to sell. They arent in monster.ca they are banging on doors making $. Make a buch of pre setup asterisk configs that would be most popular make marketing material, dump on website. go in trade shows. Demo and make $ Steve kalcevich Douglas Garstang wrote: I'm curious why the number of jobs out there requiring Asterisk seems to be pretty low. After looking around dice, monster, careerbuilder etc, I was surprised to find no more than 3-4 employment opportunities with Asterisk throughout the US. Is it really that low? There seems to be a job of opportunities for Cisco and other vendors solutions (duh... GUI's are good... duh). I wonder if demand will increase, or am I just looking in the wrong places? - Doug. --- - ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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 ___ --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] PSTN line quality
I'm looking for some input from someone with real experience of telephony. I am having problems with the sound quality on our PSTN line calls. Our channel banks are Adtran 600 and 750 and I spent a lot of time on the phone with Adtran trying to work out the problem. We are getting hum and noise and very low volume on calls. I can increase the gain in zapata.conf but that increases the echo also. I acquired a Line Test Set and talked to the telco about access to a MilliWatt generator. They did not have one active, but they worked on it and activated the MilliWatt Generator on the exchange for me. I measured my lines and also the lines at another location I am having the same problem. The line levels were -20dbm at my office and -21.4dbm at the other location. My question is, how does this compare, what is the norm, and what is the recourse? Do I have the root of the problem or should the system be able to handle this loss of level? -- Chris Mason ___ --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 files, fax
Darren Nickerson wrote: 3) I'm working on a small, simple email-fax system. Just out of curiosity, what else is out there for Asterisk? I found AsterFax, but it looks a little bit hairy to set up... You really should consider HylaFAX - www.hylafax.org. It has what you're missing - a fully featured queue manager / scheduler that takes care of retries for you, and notifies the sender of any failures encountered. It can be integrated with Asterisk via analog or digital lines, or by using a software-based modem such as IAXmodem. Hi, I thought about using Hylafax, but after looking around a bit, I got the impression that it's not exactly trivial to integrate it with Asterisk, and that it will require a dedicated incoming line. Perhaps I'm mistaken? -- Webster srl Sede legale: Via del Seminario, 3 35122 Padova Sede operativa: Via S. Breda, 28 35010 Limena (PD) Tel. +39 049 8842188 Email: [EMAIL PROTECTED] Visita www.libreriauniversitaria.it ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Same Zap channel in multiple groups
Does anyone know if it would cause problems to have the same Zap channel in multiple goups? So, for example, if I have two PRIs would the following work or would it cause problems: channel = 1-23 group = 1 channel = 25-47 group = 2 channel = 1-23,25-47 group = 3 I am just curious if anyone has set some thing like this up and how it worked out. Thanks, Patrick ___ --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] PrivacyManager CallerID not passing
Hi all,
Please see the dialplan snippet below. Any hint why it does not pass the
correctly entered 10 digit number as calleridnum on to the SIP phone?
The SIP phone always shows Unknown.
exten = s,1,PrivacyManager(1,10)
exten = s,n,GotoIf($[${PRIVACYMGRSTATUS} =
SUCCESS]?privok:privfailed)
exten = s,n(privok),NoOp(CallerIDnum: ${CALLERIDNUM})
exten = s,n,Dial(SIP/9000,20)
exten = s,n,Voicemail(u9000)
exten = s,n(privfailed),Hangup()
Thanks and regards,
Patrick
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Re: [Asterisk-Users] Chanspy options in Asterisk Manager API
just as with any asterisk application, options are separated each by a pipe option1|option2|option3 regards On 1/9/06, Dan Littlejohn [EMAIL PROTECTED] wrote: The syntax for the options in chanspy are not well documented. How do I use multiple options? I am using the Asterisk Manager API and am using ChanSpy(|q) but would like to include volume ChanSpy(|q,v3) ? Any insight would be appreciated. Dan Littlejohn www.littlejohnconsulting.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 -- 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: [Asterisk-Users] Same Zap channel in multiple groups
Patrick Conroy wrote: Does anyone know if it would cause problems to have the same Zap channel in multiple goups? So, for example, if I have two PRIs would the following work or would it cause problems: The internal structures in chan_zap can only store one group association for each channel. If you try to configure it this way, only the last defined group will 'win'. ___ --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] Recording Calls at the phone
why dont use ChanSpy or Monitor? An AGI or MAGI script would let you monitor all the incoming and/or outgoing calls of anyone, taking the info from a database will make it flexible so you can add more monitored people, and then download the audio via web, or even email it to who it may concern. On 1/9/06, Michael Sampson [EMAIL PROTECTED] wrote: Starting and stopping the recording is based off of the message taking software which knows when I call is going on. They do make recording devices that go in between the headset and phone, but they take batteries. I can't really have a recording device running off batteries in a call center. I think I'm just going to get SIP to FXO adapters and run the recording control off the FXO port. Michael Sampson Information Systems Manager Customer Contact Services [EMAIL PROTECTED] 952-936-4000 Ioan Indreias wrote: A (too) simple sollution to your problem is to take the analog audio from your IP phone using a module atached between the curly handset cord and the base unit of the IP phone - like http://www.quasarelectronics.com/tre156.htm So, basically you need to change the old RJ11 - 1/8 inch recording - RJ11 system you have used to a new one with RJ10 - 1/8 inch recording - RJ10. Sure, this solution works only if the handeset it is attached through a RJ10 port to the handset. I do not know exactly how your software will deal with this change as there should be a mechnism to start stop recording based on the audio level injected into PC's audio card (mic port). Hope it helps. Ioan Indreias Modulo Consulting - http://www.modulo.ro I'm not really trying to monitor anything on the asterisk box at all. I guess this is more of an SIP phone question. Really all I need is to get the audio from an SIP phone, both the caller and callie, to a 1/8th inch stereo jack that I can plug into a mic input. Michael Sampson Information Systems Manager Customer Contact Services [EMAIL PROTECTED] 952-936-4000 Douglas Garstang wrote: On Demand-monitoring? If your referring to monitoring specific agents calls, I'm still trying to work out how to do that. You can either monitor all calls for a queue, or all calls for all agents, but not all calls for a specific agent. I tried to use the Monitor() command on it's own to start recording when an agent receives a call, but that does not appear to work. -Original Message- From: Francesco Peeters (Asterisk) [mailto:[EMAIL PROTECTED] Sent: Friday, January 06, 2006 7:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Recording Calls at the phone On Fri, January 6, 2006 15:37, Michael Sampson said: I work for a call center and we are looking at using asterisk to have our operators take calls. Our message taking software records all the calls on the operators computers. Right now we use these recording controls from radio shack that plug in between the wall jack and the phone and plug in via a 1/8 inch stereo connector to the mic input on the computer. If I buy an IP phone I can't do that. I could get an FXO adapter and regular phones, but I'm looking to get as little equipment as possible. Radio shack makes a recording control that plugs in to a 2.5 mm headset jack, but it takes batteries so thats not going to work Does anyone else do something similar? Does anyone have any ideas about what producs/setup would work for this. Asterisk has a built in monitoring system. You can chose to do Always, Never or On Demand monitoring, depending on your setup and dialplan http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Monitor Good luck! ___ --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 -- 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: [Asterisk-Users] call files, fax
On Mon, 2006-01-09 at 16:40 +0100, David N. Welton wrote: Hi, I thought about using Hylafax, but after looking around a bit, I got the impression that it's not exactly trivial to integrate it with Asterisk, and that it will require a dedicated incoming line. Perhaps I'm mistaken? http://sf.net/projects/iaxmodem iaxmodem connects to asterisk via iax2 (localhost interface prefered) and exposes a /dev entry suitable for use with hylafax, it even has a hylafax modem definition file to make it a little easier. all software no hardware -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group 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] Asterisk crashing system
ok, look for the file /etc/asterisk/modules.conf . There disable autoload. Then try loading as less modules as you can. This is a list of my modules. Im attaching you a copy of my modules.conf so you can use it as a start. From there start to disable modules, I dont think is a core problem. What distro are you using? On 1/9/06, Ivan Lopez [EMAIL PROTECTED] wrote: I have Asterisk 1.2.1 installed on FC4 box, a 2451E and 2440 TDM Digium cards on PCI slots 2 and 1 respectively. When the system boots up, it freezes when it reaches Asterisk, and if I go into interactive startup and reject Asterisk, it boots up. When I enter the following command service asterisk start The system crashes completely. Has anyone seens this happen? I tried re-installing Asterisk and getting the same results. I would really appreciate your idea for a solution. 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 -- 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: [Asterisk-Users] RE: Fax, txfax -bizarre thing
andrutto wrote: Yeah, but to traditional PBX central you can plug fax machine hassle free. Well, in theory you should be able to do the same with Asterisk: plug fax machines into FXS ports on the box. I say in theory because I've not done that myself, and I've heard rumors of past problems (for some reason) in doing it that way. 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] Same Zap channel in multiple groups
On Mon, Jan 09, 2006 at 10:44:58AM -0500, Patrick Conroy wrote: Does anyone know if it would cause problems to have the same Zap channel in multiple goups? So, for example, if I have two PRIs would the following work or would it cause problems: channel = 1-23 group = 1 channel = 25-47 group = 2 channel = 1-23,25-47 group = 3 BTW: I suppose you wanted to write: group = 1 channel = 1-23 group = 2 channel = 25-47 group = 3 channel = 1-23,25-47 I am just curious if anyone has set some thing like this up and how it worked out. I figure all would be in group 3, but I'm not sure. -- 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
[Asterisk-Users] asterisk stops unexpected, no crash, but clean exit
Hello, I have installed a brand new asterisk 1.2.1 server. OS is centos (RH enterprise kernel) 4.1. Asterisk suddenly stops working. It does not generate a core dump what so ever. I looks like a clean stop of asterisk, as if you where to enter stop now in asterisk CLI. Anybody experienced this before? And how did you resolve it? met vriendelijke groet, Joash Herbrink Technical Consultant Control the flow De Kahuna groep helpt organisaties met het zakelijk gebruik van Internet. Kahuna Network Solutions levert beheerde oplossingen die de beveiliging, performance en beschikbaarheid van netwerk- en Internetinfrastructuur verbeteren. Kahuna Business Solutions levert oplossingen voor het verbeteren van on-line Customer Relationship Management (eCRM). Specialisaties: E-mail management en Web Self Service. Kahuna Telecom is de service provider op het gebied van breedband Internet, point-to-point verbindingen en vaste telefonie oplossingen. Kahuna IP-communications richt zich op het verbeteren vanSpraak-, Data- en Beeldcommunicatie doorinnovatieve inzet van middelen op basis van IP. Maanlander 14a/bm: +31 6 53 80 28 20 3824 MP Amersfoort e: [EMAIL PROTECTED] t: +31 33 4500370ext 1006URL: www.kahuna.nl f: +31 33 4500371 Voor support e-mailt u naar [EMAIL PROTECTED] of belt u in dringende gevallen naar +31 33 4500373. Kahuna is winnaar van de ICT Company Award 2002, de ComputerPartner Award 2003, en staat 4e in de DeloitteTouche fast 50. ___ --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] ATA failover between datacenters
Hi Everyone, Does anyone know of any ATAs that can do proxy failover without using SRV. I don't want to rely on dns if at all possible. Basically, I have Asterisk boxes in two different data centers and I need ATAs to be able to uses the server at DC2 if DC1 goes down. The servers are already in a HA setup at each datacenter. I am looking for added protection if one of the datacenters becomes unreachable. The perfect solution I believe, would be an ATA that would failover to an alternate proxy if the first was unavailable, then failover to POTS if no proxies were available. Regards, David ___ --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] ATA failover between datacenters
I think cisco ATA can handle 2 proxies, This option is called altproxy in the web based management joash -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Thomas Sent: Monday, January 09, 2006 5:32 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ATA failover between datacenters Hi Everyone, Does anyone know of any ATAs that can do proxy failover without using SRV. I don't want to rely on dns if at all possible. Basically, I have Asterisk boxes in two different data centers and I need ATAs to be able to uses the server at DC2 if DC1 goes down. The servers are already in a HA setup at each datacenter. I am looking for added protection if one of the datacenters becomes unreachable. The perfect solution I believe, would be an ATA that would failover to an alternate proxy if the first was unavailable, then failover to POTS if no proxies were available. Regards, David ___ --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] Same Zap channel in multiple groups
On Mon, January 9, 2006 16:44, Patrick Conroy said: Does anyone know if it would cause problems to have the same Zap channel in multiple goups? So, for example, if I have two PRIs would the following work or would it cause problems: channel = 1-23 group = 1 channel = 25-47 group = 2 channel = 1-23,25-47 group = 3 I am just curious if anyone has set some thing like this up and how it worked out. Thanks, Patrick AFAIK group = 1,3 channel = 1-23 group = 2,3 channel = 25-47 should work... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] Voicemail emailed volume
We currently have most of our voicemail forwarded to user's email addresses, but the message is coming in at a way low volume. It sounds great when you listen on the phone, but it's very hard to hear when you listen on the computer. Does anyone know of a way to increase the gain on the file before sending it off? Aaron ___ --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] spandsp, rxfax, TDM400/zaptel, missed frames, any help?
Rich Adamson wrote: I'm certainly not the expert on this topic, but I believe the issue has to do with the pci bus and probably relates to the TigerJet chip used on the card. Until that's addressed, any analog modem use through the card will be marginal at best. (Same issue as with the older x100p card.) I can send and receive faxes just fine with my X100P (well, actually it was just a winmodem that was identical to the X100P). And I fax with ECM, so I notice every little glitch in the data stream. So I strongly doubt that your speculation is correct. 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] PSTN line quality
I'm looking for some input from someone with real experience of telephony. I am having problems with the sound quality on our PSTN line calls. Our channel banks are Adtran 600 and 750 and I spent a lot of time on the phone with Adtran trying to work out the problem. We are getting hum and noise and very low volume on calls. I can increase the gain in zapata.conf but that increases the echo also. I acquired a Line Test Set and talked to the telco about access to a MilliWatt generator. They did not have one active, but they worked on it and activated the MilliWatt Generator on the exchange for me. I measured my lines and also the lines at another location I am having the same problem. The line levels were -20dbm at my office and -21.4dbm at the other location. My question is, how does this compare, what is the norm, and what is the recourse? Do I have the root of the problem or should the system be able to handle this loss of level? I don't use the Adtran channel banks but do have 21+ years experience in all technical areas of telephony engineering, including transmission engineering. I'm assuming from the above description that you're using a T1 card in the asterisk box with the Adtran channel banks connected to that T1, and analog pstn lines attached to your channel banks. (Can't tell for sure if that assumption is correct or not from the above.) If that's correct, first ensure your fxo ports on the Adtrans are set to match the impedence of the pstn lines (600 ohms in the US). If that is not set correctly, you will almost always have issues with imbalance resulting in hum, noise, etc. Forgeting about echo cancellation for the moment, your objective in measuring the milliwatt generator is to get as close to 0 db of end-to- end loss as possible. If the above config assumption is correct, then adjust the transmit and receive gains on the Adtran fxo ports. To pick a starting point, simply use your new transmission test set to measure the loss on an ordinary analog pstn line to the milliwatt gen (no asterisk involvement). If that value really is -21 db, that seems like an awful lot of loss. I would expect that loss to be no more than about 10 db or so. Most telco's would find -21 db of loss unacceptable for any use, so if that value is correct, I'd suggest you have a telco problem (or we're not talking about the right config, above). The asterisk echo canceller will not function correctly with anything less then about a 5db to 7 db loss for long loops, therefore if your measured pstn loss is really -21 db, then start by setting your fxo ports to 21 - 7 = 14 db (of gain). Once you have something of reasonable volume and small (or no) echo, then try increasing the gains in 1 or 2 db steps to balance audio levels against minimal echo. ___ --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 files, fax
I thought about using Hylafax, but after looking around a bit, I got the impression that it's not exactly trivial to integrate it with Asterisk, and that it will require a dedicated incoming line. Perhaps I'm mistaken? It isn't that bad basically download compile and install the trick is to find the version of HylaFax that will compile clean under your kernel. You need a version that was released about the same time as the vintage of your kernel. IAXmodem works. It provides a virtual modem that interacts with Asterisk via IAX. Otherwise, you need a channel bank that will terminate to some POTS lines and regular modems + free serial ports, but then, any other fax package requires that as well. The big weakness in Hylafax is the client. 90% of the time the client will be under Windows, and your choices are Cypheus, which is pretty and user friendly but slow and crash-y or WHFC which is ugly and nasty but works 100% and has slick features like offline faxing. hth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call files, fax
Colin Anderson [EMAIL PROTECTED] wrote: The big weakness in Hylafax is the client. 90% of the time the client will be under Windows, and your choices are Cypheus, which is pretty and user friendly but slow and crash-y or WHFC which is ugly and nasty but works 100% and has slick features like offline faxing. There's a few more choices than those two ;-) See: http://www.hylafax.org/content/Client_Software -Darren ___ --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] spandsp, rxfax, TDM400/zaptel, missed frames, any help?
I'm certainly not the expert on this topic, but I believe the issue has to do with the pci bus and probably relates to the TigerJet chip used on the card. Until that's addressed, any analog modem use through the card will be marginal at best. (Same issue as with the older x100p card.) I can send and receive faxes just fine with my X100P (well, actually it was just a winmodem that was identical to the X100P). And I fax with ECM, so I notice every little glitch in the data stream. So I strongly doubt that your speculation is correct. It would be very interesting to know the real numbers that have it working. The archives (and about two/three years of attempting to help others with the exact same problem) suggests no better then maybe one in ten or twenty will ever get spandsp to work with the digium x100p or TDM card. Maybe the trick is for you to identify 'exactly' which winmodem card does work; others would be very happy to give it a try without a doubt! ___ --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] spandsp, rxfax, TDM400/zaptel, missed frames, any help?
On Mon, 2006-01-09 at 11:15 -0600, Rich Adamson wrote: It would be very interesting to know the real numbers that have it working. The archives (and about two/three years of attempting to help others with the exact same problem) suggests no better then maybe one in ten or twenty will ever get spandsp to work with the digium x100p or TDM card. Maybe the trick is for you to identify 'exactly' which winmodem card does work; others would be very happy to give it a try without a doubt! I bought a really cheap clone off ebay and it worked first time every time (so far). Even on longer many page faxes. Getting one with an md3200 chipset (which is not what I have) and will try it on that and see and report back).. this isnt a critical system especially for faxing, so it would be interesting for me to see if this problem you are talking about has any problems with either card. I can try to get the exact chip I have when I open it up to install the new card when it gets here) -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group 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] spandsp, rxfax, TDM400/zaptel, missed frames, any help?
Rich Adamson wrote: I'm certainly not the expert on this topic, but I believe the issue has to do with the pci bus and probably relates to the TigerJet chip used on the card. Until that's addressed, any analog modem use through the card will be marginal at best. (Same issue as with the older x100p card.) I can send and receive faxes just fine with my X100P (well, actually it was just a winmodem that was identical to the X100P). And I fax with ECM, so I notice every little glitch in the data stream. So I strongly doubt that your speculation is correct. It would be very interesting to know the real numbers that have it working. The archives (and about two/three years of attempting to help others with the exact same problem) suggests no better then maybe one in ten or twenty will ever get spandsp to work with the digium x100p or TDM card. I use IAXmodem... which uses spandsp, but it's not txfax/rxfax. Maybe the trick is for you to identify 'exactly' which winmodem card does work; others would be very happy to give it a try without a doubt! The silkscreened manufacturer model number is AMI-IA92/IE92. I actually bought two of these things over the course of time, and at one time they were sold by many various vendors. They were more popular in the 2001-2004 time-frame... so finding one of these nowadays may be rather difficult. Many vendors replaced the AMI-IA92/IE92 with another product without updating their databases, so it is possible to purchase the wrong thing very easily here. The chipset on the right card clearly indicates that it is an Ambient MD3200. If you have an Ambient MD3200 chipsetted modem and Zaptel does not pick it up, then it's most likely the case that the vendor's PCI ID is just not stored in the Zaptel driver. This only requires a very simple modification to the source code to fix. 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] spandsp, rxfax, TDM400/zaptel, missed frames, any help?
It would be very interesting to know the real numbers that have it working. The archives (and about two/three years of attempting to help others with the exact same problem) suggests no better then maybe one in ten or twenty will ever get spandsp to work with the digium x100p or TDM card. Maybe the trick is for you to identify 'exactly' which winmodem card does work; others would be very happy to give it a try without a doubt! I bought a really cheap clone off ebay and it worked first time every time (so far). Even on longer many page faxes. Getting one with an md3200 chipset (which is not what I have) and will try it on that and see and report back).. this isnt a critical system especially for faxing, so it would be interesting for me to see if this problem you are talking about has any problems with either card. I can try to get the exact chip I have when I open it up to install the new card when it gets here) It would _very_ interesting to see the data, so please do post it to the list. There are a fair number of people interested in selling small pbx's with fax/modem/pos support that actually works reliably. Have you tried the TDM card yet? ___ --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] SPA-841 spontaneous voicemail problem
Hello. A while back, I noticed an odd problem with our SPA-841 phones connected to Asterisk. Now we are having a different odd problem, and I'm not sure if they're related. I wonder if anyone else has experienced anything else like this, and/or if there is any reasonable explanation? Occasionally, one of our SPA-841's will spontaneously start up with Welcome to Comedian Mail! on the speaker phone. No one is near the phone or touching it. It is as if the Invisible Man walked up and pushed the dial voicemail button. I have obviously been unable to reproduce this problem, and I'd say it has happened maybe half a dozen times or so that I know of, on approximately 35 phones over the last 5 months. These phones use unroutable IP addresses, and are on a dedicated network which is not physically connected to the Internet. The only non-phone devices on the same physical network are the Asterisk server and a configuration server for the phones, so it seems unlikely to be rogue packets. The new problem, which may be related but I have no idea at this point: this weekend I got 2 reports of cases where an agent (AgentCallbackLogin) is on a call (with a customer, via queue()), and the call is suddenly interrupted by Allison's voice announcing something. In one case, it was the Comedian Mail login prompt. The other case was a prerecorded prompt we use before calls are sent to one of our queues. I have no idea how this audio stream could be merged with the agent/customer conversation. We do not have meetme turned on, so I can't imagine Asterisk would be doing the audio stream merge. The only thing I could think of was that the SPA-841's were spontaneously dialing voicemail and doing a conference at the phone itself. However, this doesn't explain the non-voicemail prerecorded prompt. We don't have any direct-dial extension which plays that prompt. You need to dial in from an outside line, and choose at least one menu option, before you can hear that prompt. So I still have no clue how the phone could be doing this, and no clue how Asterisk could be doing it. I am again, unable to reproduce the problem. In the message log, around the time of the other prompt issue, I saw: Jan 6 13:38:15 NOTICE[7627] channel.c: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format ulaw since our native format has changed to slin However, the logged channel Local/228 is unrelated to the SIP phones or the PRI where our calls come in, so I don't think this is likely to be related to this problem. I don't see any other log lines which are out of the ordinary. At the time of the voicemail prompt problem, I see: Jan 9 09:33:20 WARNING[7627] app_voicemail.c: Couldn't read username This makes sense, but isn't very helpful. I'd appreciate it if anyone could shed any light on this situation, though I admit I don't have very high hopes. Thanks, Alan Ferrency pair Networks, Inc. [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] SNOM Hotdesking...
Hi, I have now managed to get it working with asterisk 1.0.10 I had to modify the patch http://bugs.digium.com/bug_view_page.php?bug_id=6035 as its for the latest version of asterisk but it works very well now. Thanks for the pointer. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Maik Schmitt Sent: 09 January 2006 11:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SNOM Hotdesking... Hi, I raised this with SNOM and they say it is purely an asterisk problem and it needs to be fixed (asterisk that is). If asterisk sent a 401 instead of a 403 the phone would work fine and we would all be happy. Here you can find a patch that will fix it: http://bugs.digium.com/bug_view_page.php?bug_id=6035 Maik Schmitt ___ --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] ectoolkit
Anyone have any information regarding the ectoolkit on svn? ~ron ___ --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] snom programmable buttons
Unfortunately I asked the same question a day or two with no response... It appears the only way is to use a very beta patch, look on bugs.digium.com and search for snom pickup, you should find it. But I wouldn't recommend using it in a production environment just yet.. It's funny cause asterisk is awesome for large setups but when you want to do a small office, most people complain about lacking many features compared to their old avaya partner's, etc.. Such as line sharing, call pickup when on hold or ringing, intercom to a person using their blf button, etc I am still trying to figure out ways for my small business users to be happier, so again if anyone has any experience of ideas, I would appreciate it, and hopefully the patch on bugs will help you... Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of cfh Sent: Monday, January 09, 2006 8:07 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] snom programmable buttons Hi, I want to pick up a call with the snom's programmable buttons(snom190 -SIP 3.60x, snom360-SIP 4.1) with asterisk server (v 1.2.0), I tried with the option 'Destination' and when the incoming call arrive to another snom phone the button blinking. In this way I can only pick down it pressing the blinking button. The solution is call the *8 or parcking the call but my pbroblem is when the incoming call are 2 or 3 and I would press a programmable button to pick up the calls. Is possible have configured asterisk and the snom phone with the function shared line? Are there solutions ? Thanks Luca L. [cfh] ___ --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 This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --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] Answer call waiting / flash with Zaptel POTS and VOIP
Hello, hoping someone out there has some ideas - I have a VOIP line that has call waiting. It is terminated at a Sipura 3000 and the POTS side of that device connects to an FXO port in my * box. I also have a POTS/PSTN line that terminates in another FXO port on my * box. There are two FXS ports which feed cordless phones. I'm using the Zaptel TDM400 card. This gives 2 extensions + 2 lines in/out and the VOIP line has call waiting. This is the problem: Asterisk (or Zaptel) only interprets the flash from a handset to mean switch between FXO cards. Or at least I think that's what's happening. If I'm on an extension and using the VOIP line, and a call comes in on the POTS line, I get the audible beep and I can answer it by pressing flash on the extension. However, if I'm on the VOIP line and another call is placed to the VOIP number, I hear the beep, but pressing flash doesn't answer it. Instead it gives me a dial tone. Pressing flash again gets me back to the original VOIP call but there is no way to answer the call waiting on the VOIP line. What I'm looking for is a way to tell * not to interpret the flash and instead pass it out the line. I can always answer non-call-waiting incomming calls on the other extension. I'd like to be able to use flash for signaling the VOIP (upstream?) to switch to the callwaiting call and then back as needed. I tried setting callwaiting=no in zapata.conf and restarting * but that didn't have the desired effect. I think it prevented me from hearing an audible beep when one line and extension were in use and a call came in the other line... that's okay. But it didn't help answer the callwaiting call on the voip line. I need a way to tell * not to interpret the flash, and instead, pass it out the line connected to this extension. Any ideas? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Jobs
I knew someone will not be able to resist :) On 1/8/06, Steve Totaro [EMAIL PROTECTED] wrote: I am not sure why you are looking for jobs doing Asterisk work when less than two weeks ago you were publicly bashing on the list. Steve Consulting is fine, as long as I'm working for someone else. Setting up my own company etc isn't really what I'm looking for. I don't want the risk. If there aren't actual companies offering good paying positions, then there's really no opportunities for me. -Original Message- From: Steven Kalcevich [mailto:[EMAIL PROTECTED] Sent: Sat 1/7/2006 7:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Asterisk Jobs I think it would be biggest is in consulting. The people that refuse or cant to pay for call manager or Avaya's one. Example asterisk sugarcrm.com they work together. Thats really good to sell. They arent in monster.ca they are banging on doors making $. Make a buch of pre setup asterisk configs that would be most popular make marketing material, dump on website. go in trade shows. Demo and make $ Steve kalcevich Douglas Garstang wrote: I'm curious why the number of jobs out there requiring Asterisk seems to be pretty low. After looking around dice, monster, careerbuilder etc, I was surprised to find no more than 3-4 employment opportunities with Asterisk throughout the US. Is it really that low? There seems to be a job of opportunities for Cisco and other vendors solutions (duh... GUI's are good... duh). I wonder if demand will increase, or am I just looking in the wrong places? - Doug. --- - ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk featdmf signalling.
I've recently started PIC'ing some calls into a asterisk box across a feature group D trunk from Verizon. Everything seems to work ok, except for some reason Asterisk doesn't grab the full caller ID from Verizon. I can see that they do send it, but Asterisk drops the first 2 numbers. Looking at the debug log I see. I see that Verizon is sending the digits and the asterisk debug seems to understand it fine. But within Asterisk it appears to truncate the first two digits of the callerid. The called-from appears in both the .csv log and my mysql cdr log as 72652437 and it sends this number out as the callerid as well. I'm at a loss as to why it misses the first 2 digits, any suggestions would be appreciated. Jan 6 11:00:19 DEBUG[18897]: DTMF digit: * on Zap/73-1 Jan 6 11:00:19 DEBUG[18897]: DTMF digit: 5 on Zap/73-1 Jan 6 11:00:19 DEBUG[18897]: DTMF digit: 1 on Zap/73-1 Jan 6 11:00:19 DEBUG[18897]: DTMF digit: 7 on Zap/73-1 Jan 6 11:00:19 DEBUG[18897]: DTMF digit: 2 on Zap/73-1 Jan 6 11:00:19 DEBUG[18897]: DTMF digit: 6 on Zap/73-1 Jan 6 11:00:19 DEBUG[18897]: DTMF digit: 5 on Zap/73-1 Jan 6 11:00:19 DEBUG[18897]: DTMF digit: 2 on Zap/73-1 Jan 6 11:00:20 DEBUG[18897]: DTMF digit: 4 on Zap/73-1 Jan 6 11:00:20 DEBUG[18897]: DTMF digit: 3 on Zap/73-1 Jan 6 11:00:20 DEBUG[18897]: DTMF digit: 7 on Zap/73-1 Jan 6 11:00:20 DEBUG[18897]: DTMF digit: # on Zap/73-1 Jan 6 11:00:20 DEBUG[18897]: DTMF digit: * on Zap/73-1 Jan 6 11:00:20 DEBUG[18897]: DTMF digit: 4 on Zap/73-1 Jan 6 11:00:20 DEBUG[18897]: DTMF digit: 1 on Zap/73-1 Jan 6 11:00:20 DEBUG[18897]: DTMF digit: 9 on Zap/73-1 Jan 6 11:00:21 DEBUG[18897]: DTMF digit: 3 on Zap/73-1 Jan 6 11:00:21 DEBUG[18897]: DTMF digit: 7 on Zap/73-1 Jan 6 11:00:21 DEBUG[18897]: DTMF digit: 6 on Zap/73-1 Jan 6 11:00:21 DEBUG[18897]: DTMF digit: 6 on Zap/73-1 Jan 6 11:00:21 DEBUG[18897]: DTMF digit: 1 on Zap/73-1 Jan 6 11:00:21 DEBUG[18897]: DTMF digit: 0 on Zap/73-1 Jan 6 11:00:21 DEBUG[18897]: DTMF digit: 6 on Zap/73-1 Jan 6 11:00:21 DEBUG[18897]: DTMF digit: # on Zap/73-1 Regards Michael Baird ___ --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] Zaptel errors (power alarm?)
We've been having lost dialtone problems on one of our analog station ports. Just before rebooting this time I noticed these in our dmesg outputonce the PBX comes back I'll get the times, but I can't help but think this must have something to do with it. Anyone? Do we need to have digium send us a replacement part? Ouch, part reset, quickly restoring reality (1) Power alarm on module 2, resetting! Ouch, part reset, quickly restoring reality (1) Power alarm on module 2, resetting! Ouch, part reset, quickly restoring reality (1) Power alarm on module 2, resetting! Ouch, part reset, quickly restoring reality (1) Power alarm on module 2, resetting! zaptel Disabled echo canceller because of tone (rx) on channel 23 zaptel Disabled echo canceller because of tone (rx) on channel 23 Ouch, part reset, quickly restoring reality (1) Power alarm on module 2, resetting! zaptel Disabled echo canceller because of tone (rx) on channel 23 -- Genius might be described as a supreme capacity for getting its possessors into trouble of all kinds. -- Samuel Butler ___ --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] OT: IAXModem in inittab causes modem to be unres ponsive, running from console it's OK
faxguy, maybe you can tell me why As I've noted in previous posts I'm evaluating HylaFax with IAXModem. When I run iaxmodem and faxgetty through a console the modem works 100% I have yet to find a fax that it won't tie up with. When I run IAXmodem and faxgetty in initttab, the modem is extremely slow to respond and only actually does anything about half the time, the rest of the time the HylaFax client says: Initializing server and it stays there forever. If it does send a fax, the fax is usually corrupt. faxguy, can you comment? Running FC2 latest, SMP NetFinity, Asterisk 1.0.9 as non-root, latest SpanDSP for -users, not -dev (Pre21?) all on same box. The 2 virtual modems are set up to a non-obvious port (not 4569) to avoid port conflict. ___ --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] Problem Compiling Zaptel 1.2.1
[EMAIL PROTECTED] zaptel-1.2.1]# make gcc -I/lib/modules/2.4.21-4.ELsmp/build/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -I/lib/modules/2.4.21-4.ELsmp/build/drivers/net -Wall -I. -Wstrict-prototypes -fomit-frame-pointer -I/lib/modules/2.4.21-4.ELsmp/build/drivers/net/wan -I/lib/modules/2.4.21-4.ELsmp/build/include/net -DMODVERSIONS -include /lib/modules/2.4.21-4.ELsmp/build/include/linux/modversions.h -DSTANDALONE_ZAPATA -o zaptel.o -c zaptel.c In file included from /lib/modules/2.4.21-4.ELsmp/build/include/linux/tqueue.h:19, from /lib/modules/2.4.21-4.ELsmp/build/include/linux/aio.h:4, from /lib/modules/2.4.21-4.ELsmp/build/include/linux/net.h:88, from /lib/modules/2.4.21-4.ELsmp/build/include/linux/fs.h:15, from /lib/modules/2.4.21-4.ELsmp/build/include/linux/capability.h:17, from /lib/modules/2.4.21-4.ELsmp/build/include/linux/binfmts.h:4, from /lib/modules/2.4.21-4.ELsmp/build/include/linux/sched.h:10, from /lib/modules/2.4.21-4.ELsmp/build/include/linux/mm.h:22, from /lib/modules/2.4.21-4.ELsmp/build/include/linux/slab.h:14, from /lib/modules/2.4.21-4.ELsmp/build/include/linux/proc_fs.h:5, from zaptel.c:45: /lib/modules/2.4.21-4.ELsmp/build/include/asm/system.h: In function `__set_64bit_var': /lib/modules/2.4.21-4.ELsmp/build/include/asm/system.h:189: warning: dereferencing type-punned pointer will break strict-aliasing rules /lib/modules/2.4.21-4.ELsmp/build/include/asm/system.h:189: warning: dereferencing type-punned pointer will break strict-aliasing rules In file included from /lib/modules/2.4.21-4.ELsmp/build/include/linux/sched.h:24, from /lib/modules/2.4.21-4.ELsmp/build/include/linux/mm.h:22, from /lib/modules/2.4.21-4.ELsmp/build/include/linux/slab.h:14, from /lib/modules/2.4.21-4.ELsmp/build/include/linux/proc_fs.h:5, from zaptel.c:45: /lib/modules/2.4.21-4.ELsmp/build/include/linux/smp.h: At top level: /lib/modules/2.4.21-4.ELsmp/build/include/linux/smp.h:31: error: conflicting types for 'smp_send_reschedule' /lib/modules/2.4.21-4.ELsmp/build/include/asm/smp.h:41: error: previous declaration of 'smp_send_reschedule' was here /lib/modules/2.4.21-4.ELsmp/build/include/linux/smp.h:31: error: conflicting types for 'smp_send_reschedule' /lib/modules/2.4.21-4.ELsmp/build/include/asm/smp.h:41: error: previous declaration of 'smp_send_reschedule' was here make: *** [zaptel.o] Error 1 What could I be missing? :) Thank you -- Leandro Rzezak[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] Problem Compiling Zaptel 1.2.1
If you've ever compiled and installed an older version of * on this box, specifically from the 1.0 era, it's possible you need to try removing /usr/include/asterisk and see if that helps. Moj Leandro Rzezak wrote: [EMAIL PROTECTED] zaptel-1.2.1]# make gcc -I/lib/modules/2.4.21-4.ELsmp/build/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -I/lib/modules/2.4.21-4.ELsmp/build/drivers/net -Wall -I. -Wstrict-prototypes -fomit-frame-pointer -I/lib/modules/2.4.21-4.ELsmp/build/drivers/net/wan -I/lib/modules/2.4.21-4.ELsmp/build/include/net -DMODVERSIONS -include /lib/modules/2.4.21-4.ELsmp/build/include/linux/modversions.h -DSTANDALONE_ZAPATA -o zaptel.o -c zaptel.c In file included from /lib/modules/2.4.21-4.ELsmp/build/include/linux/tqueue.h:19, from /lib/modules/2.4.21-4.ELsmp/build/include/linux/aio.h:4, from /lib/modules/2.4.21-4.ELsmp/build/include/linux/net.h:88, from /lib/modules/2.4.21-4.ELsmp/build/include/linux/fs.h:15, from /lib/modules/2.4.21-4.ELsmp/build/include/linux/capability.h:17, from /lib/modules/2.4.21-4.ELsmp/build/include/linux/binfmts.h:4, from /lib/modules/2.4.21-4.ELsmp/build/include/linux/sched.h:10, from /lib/modules/2.4.21-4.ELsmp/build/include/linux/mm.h:22, from /lib/modules/2.4.21-4.ELsmp/build/include/linux/slab.h:14, from /lib/modules/2.4.21-4.ELsmp/build/include/linux/proc_fs.h:5, from zaptel.c:45: /lib/modules/2.4.21-4.ELsmp/build/include/asm/system.h: In function `__set_64bit_var': /lib/modules/2.4.21-4.ELsmp/build/include/asm/system.h:189: warning: dereferencing type-punned pointer will break strict-aliasing rules /lib/modules/2.4.21-4.ELsmp/build/include/asm/system.h:189: warning: dereferencing type-punned pointer will break strict-aliasing rules In file included from /lib/modules/2.4.21-4.ELsmp/build/include/linux/sched.h:24, from /lib/modules/2.4.21-4.ELsmp/build/include/linux/mm.h:22, from /lib/modules/2.4.21-4.ELsmp/build/include/linux/slab.h:14, from /lib/modules/2.4.21-4.ELsmp/build/include/linux/proc_fs.h:5, from zaptel.c:45: /lib/modules/2.4.21-4.ELsmp/build/include/linux/smp.h: At top level: /lib/modules/2.4.21-4.ELsmp/build/include/linux/smp.h:31: error: conflicting types for 'smp_send_reschedule' /lib/modules/2.4.21-4.ELsmp/build/include/asm/smp.h:41: error: previous declaration of 'smp_send_reschedule' was here /lib/modules/2.4.21-4.ELsmp/build/include/linux/smp.h:31: error: conflicting types for 'smp_send_reschedule' /lib/modules/2.4.21-4.ELsmp/build/include/asm/smp.h:41: error: previous declaration of 'smp_send_reschedule' was here make: *** [zaptel.o] Error 1 What could I be missing? :) Thank you -- Leandro Rzezak [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk featdmf signalling.
On Mon, 9 Jan 2006, Michael Baird wrote: I've recently started PIC'ing some calls into a asterisk box across a feature group D trunk from Verizon. Everything seems to work ok, except for some reason Asterisk doesn't grab the full caller ID from Verizon. I can see that they do send it, but Asterisk drops the first 2 numbers. Looking at the debug log I see. I see that Verizon is sending the digits and the asterisk debug seems to understand it fine. But within Asterisk it appears to truncate the first two digits of the callerid. The called-from appears in both the .csv log and my mysql cdr log as 72652437 and it sends this number out as the callerid as well. I'm at a loss as to why it misses the first 2 digits, any suggestions would be appreciated. There are two kinds of FGD protocols on Asterisk, I got stuck with this when it only supported the one that my IXC didn't support. Read through chan_zap to see to see the format of each one. dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail emailed volume
Aaron Daniel wrote: We currently have most of our voicemail forwarded to user's email addresses, but the message is coming in at a way low volume. It sounds great when you listen on the phone, but it's very hard to hear when you listen on the computer. Does anyone know of a way to increase the gain on the file before sending it off? Could you provide a little more information? Is this incoming VOIP or incoming via a Zap channel? If it's Zap, what hardware are you using? Did you try increasing the rxgain? Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.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] Problem Compiling Zaptel 1.2.1
Removed /usr/include/asterisk, same thing.. Any clue?On 1/9/06, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote:If you've ever compiled and installed an older version of * on this box, specifically from the 1.0 era, it's possible you need to try removing/usr/include/asterisk and see if that helps.MojLeandro Rzezak wrote: [EMAIL PROTECTED] zaptel-1.2.1]# make gcc -I/lib/modules/2.4.21- 4.ELsmp/build/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -I/lib/modules/2.4.21-4.ELsmp/build/drivers/net -Wall -I. -Wstrict-prototypes -fomit-frame-pointer -I/lib/modules/2.4.21-4.ELsmp /build/drivers/net/wan -I/lib/modules/2.4.21-4.ELsmp/build/include/net -DMODVERSIONS -include /lib/modules/2.4.21-4.ELsmp/build/include/linux/modversions.h -DSTANDALONE_ZAPATA -o zaptel.o -c zaptel.c In file included from /lib/modules/2.4.21-4.ELsmp/build/include/linux/tqueue.h:19,from /lib/modules/2.4.21-4.ELsmp/build/include/linux/aio.h:4,from /lib/modules/2.4.21-4.ELsmp/build/include/linux/net.h:88,from /lib/modules/2.4.21-4.ELsmp/build/include/linux/fs.h:15,from /lib/modules/2.4.21-4.ELsmp /build/include/linux/capability.h:17,from /lib/modules/2.4.21-4.ELsmp/build/include/linux/binfmts.h:4,from /lib/modules/2.4.21-4.ELsmp/build/include/linux/sched.h:10, from /lib/modules/2.4.21-4.ELsmp/build/include/linux/mm.h:22,from /lib/modules/2.4.21-4.ELsmp/build/include/linux/slab.h:14,from /lib/modules/2.4.21-4.ELsmp/build/include/linux/proc_fs.h:5,from zaptel.c:45: /lib/modules/2.4.21-4.ELsmp/build/include/asm/system.h: In function `__set_64bit_var': /lib/modules/2.4.21-4.ELsmp/build/include/asm/system.h:189: warning: dereferencing type-punned pointer will break strict-aliasing rules /lib/modules/2.4.21-4.ELsmp/build/include/asm/system.h:189: warning: dereferencing type-punned pointer will break strict-aliasing rules In file included from /lib/modules/2.4.21-4.ELsmp/build/include/linux/sched.h:24, from /lib/modules/2.4.21-4.ELsmp/build/include/linux/mm.h:22,from /lib/modules/2.4.21-4.ELsmp/build/include/linux/slab.h:14,from /lib/modules/2.4.21-4.ELsmp/build/include/linux/proc_fs.h:5,from zaptel.c:45: /lib/modules/2.4.21-4.ELsmp/build/include/linux/smp.h: At top level: /lib/modules/2.4.21-4.ELsmp/build/include/linux/smp.h:31: error: conflicting types for 'smp_send_reschedule' /lib/modules/2.4.21- 4.ELsmp/build/include/asm/smp.h:41: error: previous declaration of 'smp_send_reschedule' was here /lib/modules/2.4.21-4.ELsmp/build/include/linux/smp.h:31: error: conflicting types for 'smp_send_reschedule' /lib/modules/2.4.21-4.ELsmp/build/include/asm/smp.h:41: error: previous declaration of 'smp_send_reschedule' was here make: *** [zaptel.o] Error 1 What could I be missing? :) Thank you -- Leandro Rzezak [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users--Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC(907) 747- x112___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Leandro Rzezak[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
[Asterisk-Users] Problem with Chan_zap.so
I just upgraded to Asterisk 1.2.1 and Asterisk fails to start with the error below. Jan 9 21:25:38 NOTICE[1339]: cdr.c:1171 do_reload: CDR simple logging enabled. Jan 9 21:25:38 WARNING[1339]: loader.c:326 __load_resource: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: pri_restart Jan 9 21:25:38 WARNING[1339]: loader.c:555 load_modules: Loading module chan_zap.so failed! Can Anyone tell me whats wrong. A. Izukanne Atani Communications ___ To help you stay safe and secure online, we've developed the all new Yahoo! Security Centre. http://uk.security.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Decent sub-$100 SIP phone.
Hey, all. I quoted a customer about $100 for some cheap SIP phones. I was planning on using the BT-102's, but he called said they look like Princess phones, and I have to admit that he has a point. Some of the other inexpensive phones look decent, but (for example) the SPA-841's wiki entry says the remote end gets a lot of static. Since it'll be being used from a noisy environment (a cleanroom), the less overall static, the better. Someone suggested the Polycom 301's, but I'd lose money on them. [I'll go with them if I have to, as I'm making money elswhere, but still...] So, does anyone have any suggestions for decent sub-$100, professional-looking SIP phones? Thanks! Ken D'Ambrosio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail emailed volume
It's any voicemail from any line on the system, whether it's SIP, IAX, or ZAP... The voicemail message is basically so low in volume that my boss has almost blown his speakers switching between listening to voicemail and listening to whatever music he listens to lol... I've got the rxgain's set the highest we can have it without affecting the echo, so I can't really do much there. Aaron Darrick Hartman wrote: Aaron Daniel wrote: We currently have most of our voicemail forwarded to user's email addresses, but the message is coming in at a way low volume. It sounds great when you listen on the phone, but it's very hard to hear when you listen on the computer. Does anyone know of a way to increase the gain on the file before sending it off? Could you provide a little more information? Is this incoming VOIP or incoming via a Zap channel? If it's Zap, what hardware are you using? Did you try increasing the rxgain? Darrick ___ --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] Decent sub-$100 SIP phone.
Grandstream GXP-2000 is decent. On 1/9/06, Ken D'Ambrosio [EMAIL PROTECTED] wrote: Hey, all. I quoted a customer about $100 for some cheap SIP phones. I was planning on using the BT-102's, but he called said they look like Princess phones, and I have to admit that he has a point. Some of the other inexpensive phones look decent, but (for example) the SPA-841's wiki entry says the remote end gets a lot of static. Since it'll be being used from a noisy environment (a cleanroom), the less overall static, the better. Someone suggested the Polycom 301's, but I'd lose money on them. [I'll go with them if I have to, as I'm making money elswhere, but still...] So, does anyone have any suggestions for decent sub-$100, professional-looking SIP phones? Thanks! Ken D'Ambrosio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to connect to Asterisk
Hello All Everything was working OK, and decided to install AMP 1.10.010... and problem started. AMP took control of Asterisk... For some odd reasons I can not connect to Asterisk CLI any more. I get the following error: - [EMAIL PROTECTED] ~]$ sudo /usr/sbin/asterisk -r Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) [EMAIL PROTECTED] ~]$ But if I check the process, I do see Asterisk is running. I am running Asterisk 1.2 Any ideas? Thanking in advance... Thanks, Neal ___ --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] Cisco phones 7940
I know this isn't a specifically asterisk question, but does anyone know how to make the phone NOT use it's old config? I'm trying to get rid of the second line registration crap and it's not working. Aaron ___ --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] Decent sub-$100 SIP phone.
On Mon, 2006-01-09 at 15:28 -0500, Ken D'Ambrosio wrote: Hey, all. I quoted a customer about $100 for some cheap SIP phones. I was planning on using the BT-102's, but he called said they look like Princess phones, and I have to admit that he has a point. Some of the other inexpensive phones look decent, but (for example) the SPA-841's wiki entry says the remote end gets a lot of static. Since it'll be being used from a noisy environment (a cleanroom), the less overall static, the better. Someone suggested the Polycom 301's, but I'd lose money on them. [I'll go with them if I have to, as I'm making money elswhere, but still...] So, does anyone have any suggestions for decent sub-$100, professional-looking SIP phones? If you were looking at BudgeTones, you *might* want to look at the GXP-2000. A little nicer, and if you shop around you can get them for a decent price. http://snipurl.com/lfa3 for instance. The Polycom is nice, but I have found that the only Polycom that seems to do PoE correctly is the 601, which is definitely out of your sub-$100 price-range... Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Unable to connect to Asterisk
Check manager.conf and manager.custom.conf (installed by amp) for access lists which may be preventing you from reaching it. -Original Message- From: Nitesh Divecha [mailto:[EMAIL PROTECTED] Sent: Monday, January 09, 2006 2:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Unable to connect to Asterisk Hello All Everything was working OK, and decided to install AMP 1.10.010... and problem started. AMP took control of Asterisk... For some odd reasons I can not connect to Asterisk CLI any more. I get the following error: - [EMAIL PROTECTED] ~]$ sudo /usr/sbin/asterisk -r Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) [EMAIL PROTECTED] ~]$ But if I check the process, I do see Asterisk is running. I am running Asterisk 1.2 Any ideas? Thanking in advance... Thanks, Neal ___ --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
