Re: [asterisk-users] OT - How to add a new TAPI driver on an XP system ?
2007/11/30, Olivier <[EMAIL PROTECTED]>: > > Hi, > > To make a long story short, I can't install any TAPI driver on my XP > platform. > > A. Within Config Panel|Modems and Telephony options|Advanced parameters, > I've got a list of 7 TAPI drivers. Among them is Omniis TAPI driver for > Asterisk. > B. I can properly configure this driver (line, context, ...). > C. When I open Outlook 2002 Contacts panel, I can select "Call this > contact" from Actions menu. > D. When the "New call" popup appears, I can click on "Dialing Options ..." > > E. When the "Dialing options" popup appears, there is a scrolling list > "Dialing using line" in which I can find a list of modem drivers but not a > single TAPI driver. > F. If I check running Services (Config Panel|Administration > Tools|Services], Telephony service is said to be running. > > My questions are: > 1. Is there a way to set a TSP driver to be default driver to be used and > skip "Dialing options" windows ? > 2. Should I see TAPI drivers within "Dialing using line" scrolling list ? > > Regards > > Hi, Replying to myself, it appears installing TAPI drivers on a new XP system worked flawlessly : new driver now appears in "Dialing using line" scrolling list. I couldn't find root cause bug but re-installing Windows XP seems to be a workaround. Regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Oracle and asterisk
thnsk for giving me reply, Bhrugu mehta On Dec 3, 2007 12:41 PM, Tilghman Lesher <[EMAIL PROTECTED]> wrote: > On Monday 03 December 2007 00:48:55 Bhrugu Mehta wrote: > > I want to connect asterisk with oracle database. > > You'll need to install the Oracle ODBC driver for Linux. One word of warning, > though: the ODBC driver linked against the InstantClient library has a very > nasty resource leak in the library itself. Specifically, on every connection, > it leaks 2 file descriptors and fails to close cursors properly on each > statement executed. Therefore, be sure that you're linking to the other > Oracle client library, not the InstantClient. > > http://home.fnal.gov/~dbox/oracle/odbc/ > > -- > Tilghman > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Oracle and asterisk
On Monday 03 December 2007 00:48:55 Bhrugu Mehta wrote: > I want to connect asterisk with oracle database. You'll need to install the Oracle ODBC driver for Linux. One word of warning, though: the ODBC driver linked against the InstantClient library has a very nasty resource leak in the library itself. Specifically, on every connection, it leaks 2 file descriptors and fails to close cursors properly on each statement executed. Therefore, be sure that you're linking to the other Oracle client library, not the InstantClient. http://home.fnal.gov/~dbox/oracle/odbc/ -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] New VICIDIAL astGUIclient Release: 2.0.4
Hello, We've released another update to our astGUIclient-VICIDIAL suite: 2.0.4 http://astguiclient.sf.net/ The client suite runs on most modern web browsers on almost any GUI-capable operating system, and it includes the VICIDIAL call center suite and the astGUIclient client-side web app which extends your phone's functionality. This package is free and GPL. (the suite is not an asterisk configuration tool) This package is geared towards Asterisk installations with SIP,IAX or Zap phones and Zaptel, IAX or SIP trunks. For this release, we have focused on adding new features to inbound call handling such as custom music-on-hold, agent alert messages per inbound group and agent-rank call routing per skill as well as several other new administrative features. We have also tested the suite on Asterisk versions through 1.2.24. All client web-apps and administration pages are available in English, Spanish, Greek and German, with rough translations of French, Polish, Italian, Portuguese and Brazillian Portuguese for the client web-apps only. Check out the project blog for more information: http://astguiclient.blogspot.com Let me know what you think. Thanks, MATT--- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Oracle and asterisk
hi, all I want to connect asterisk with oracle database. how to start this , that's i dont know . any pls help me thnks in advance Bhrugu mehta ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to setup redundant SIP peers
On Nov 30, 2007 11:46 PM, Thomas Balsfulland <[EMAIL PROTECTED]> wrote:
> Hello list,
>
> I try to setup an asterisk-server with different SIP-Peers to PSTN.
> The Peer are working and configured in sip.conf:
>
> [peer1]
> type=peer
> host=10.10.10.1
>
> [peer2]
> type=peer
> host=10.10.10.2
>
> Now dialout is no problem. Extensions.conf says:
>
> exten => _0Z.,1,Dial(SIP/49${EXTEN:[EMAIL PROTECTED],30)
>
add another line
exten => _0Z.,2,Dial(SIP/49${EXTEN:[EMAIL PROTECTED],30)
ram
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Re: [asterisk-users] Softswitch digim
On Dec 3, 2007 3:12 AM, Carlos Rojas <[EMAIL PROTECTED]> wrote: > Hello averybody, > > > I'm looking the softswitch in digium website, anyone test the softswitch? > > Try freeswitch.org ram ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.15 sip.conf register
Hello, I recently upgraded from Asterisk 1.4.0 to 1.4.15... I am registering to a sip provider in my sip.conf as below [general] register=>user:password:[EMAIL PROTECTED]/extension Later down in my sip.conf I have the definition for that service provider as follows [serviceprovider] type=peer host=x.x.x.x port= outboundproxy=t.t.t.t . . . With asterisk 1.4 it would know to look in the peer definition for the IP address information... Now it appears that asterisk 1.4.15 is trying to do a DNS looking on "serviceprovider"... Of course that's coming back as an unknown host, and it no longer registers. Any steers as to how I can get asterisk 1.4.15 to look at the peer definition for the address info would be appreciated. Thanks Much! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] get SIP extension status without calling it
I appreciate the feedback. I too am not using and hope
not to use agent login. I don't know if I can apply
your dialplan because I need to distinguish whether an
agent is busy (thus average conversation time is
usually around 1-2 minutes) or has DND on (agent can
be absent for quite a while, eg. 15 min.).
So if it's "busy" I want it to loop over and over. If
it's DND then bail out.
Also, the DND detection thing I'm obsessed with is
two-fold. I can also use it as a kind of "presence"
indicator so that other users can tell if the
extensions they want to reach are "available" or not
before even making/transferring the call.
Am wondering if IAX is better at this than SIP.
Thanks again.
--- Richard Revels <[EMAIL PROTECTED]> wrote:
> I'm using 1.2.6 with the dialplan I posted so I
> guess the UA you are
> using is just plain hosing you.
>
> Anyway, with the queue I believe the music on hold
> is played to the
> inbound side until the call is picked up by an
> agent. The queue tries
> every seconds to get an agent for
> seconds. If that
> fails for however long you set the limit to then the
> dialplan
> continues. I use this to set the priority of the
> call a little higher
> and loop it back into the queue but it could be used
> for a problem
> such as yours just as easily. I don't use the agent
> login and all
> that so I may be talking about something that
> doesn't apply to your
> configuration. If so, sorry for wasting your time.
>
> exten => +1X,1,NoOp(Inbound call from
> ${CALLERIDNUM})
> exten => +1X,n,Answer()
> exten => +1X,n,Set(GROUP()=cloud)
> exten => +1X,n,Set(QUEUE_PRIO=0)
> exten =>
> +1X,n(waiting),Queue(mainline600)
> exten =>
> +1X,n,Set(QUEUE_PRIO=$[${QUEUE_PRIO} + 5])
> exten => +1X,n,GoTo(waiting)
> exten => +1X,n,HangUp
>
> On Dec 2, 2007, at 7:02 PM, Vieri wrote:
>
> > I'd like to add that "show hints" on * CLI
> displays
> > the following for ext 4053 tested below:
> >
> >4053: SIP/4053
> > State:IdleWatchers 0
> >
> > (it should be "unavailable" or something, but
> anyway,
> > ChanIsAvail reports an AVAILSTATUS of 0, ie.
> unknown)
> >
> > --- Vieri <[EMAIL PROTECTED]> wrote:
> >
> > > Thanks Richard but I think that ChanIsAvail must
> be
> > > buggy (based on some user comments in the wiki,
> > > although quite outdated).
> > >
> > > I have the hint entry as you say (am using
> FreePBX
> > > and
> > > it's already there).
> > >
> > > But whenever I call ChanIsAvail with the s
> option I
> > > always get:
> > > ${AVAILSTATUS} = 0 AST_DEVICE_UNKNOWN -
> "Unknown";
> > > channel is valid, but unknown state.
> > >
> > > I might be doing something wrong but here is the
> > > code:
> > >
> > > [IVR-menu1]
> > > exten => s,1,Answer()
> > > (...)
> > > exten => s,n,Playback(welcome)
> > > exten => s,n,ChanIsAvail(SIP/4053|s)
> > > exten => s,n,NoOp(DEBUG: availstatus is
> > > ${AVAILSTATUS})
> > >
> > > In extensions.conf I also have:
> > > exten => 4053,hint,SIP/4053
> > >
> > > I'm using Astrisk 1.2. Is ChanIsAvail working
> well
> > > in
> > > 1.2?
> > >
> > > As far as setting a time limit on a call in the
> > > queue
> > > is concerned, it doesn't sound "nice" for the
> caller
> > > to be dropped after a few rings when it could
> have
> > > been dropped right fom the beginning. It could
> be a
> > > solution but it doesn't sound "right" ;-).
> > >
> > > Vieri
> > >
> > > --- Richard Revels <[EMAIL PROTECTED]>
> wrote:
> > >
> > > > In the sip.conf entry assign a context.
> > > >
> > > > In that context, hint the extension i.e. exten
> =>
> > > > 7302,hint,SIP/7302.
> > > >
> > > > Before you get ready to dial, or whatever, do
> > > > chanisavail i.e.
> > > >
> > > > exten =>
> > > >
> _1,n(CheckUse),ChanIsAvail(SIP/${EXTEN:1},js)
> > > > exten => _1,n,Playback(beep)
> > > > exten => _1,n,Dial(SIP/${EXTEN},2)
> > > > exten =>
> > > > _1,n,Goto(result-${DIALSTATUS},${EXTEN},1)
> > > > exten =>
> _1,CheckUse+101,SayDigits(${EXTEN:1})
> > > > exten =>
> > > _1,CheckUse+102,Playback(vm-isonphone)
> > > > exten => _1,CheckUse+103,Hangup()
> > > >
> > > > This is from the paging stuff. It checks the
> > > > primary extension before
> > > > ringing the auto answer extension of the
> phone. I
> > > > seem to remember it
> > > > detecting DND as well for the Cisco 7960.
> > > >
> > > > I don't see it in this message but I seem to
> > > > remember seeing somewhere
> > > > in this thread that the goal is to keep people
> > > from
> > > > being in a queue
> > > > forever. Why not just set a time limit on the
> > > queue
> > > > and play back
> > > > "all operators busy" and hang up if a call
> hits
> > > that
> > > > limit?
> > > >
> > > > Richard
Never miss a thing. Make Yahoo your home page.
http://www.yahoo.com/r/hs
[asterisk-users] Subject: Newb Question
Hi, Use orecx, voip call recording and monitoring. www.orecx.com Thanks & Regards, Vidura Senadeera, Sri Lanka. Tel - +94114520001 Mobile - +9466596 yahoo/skype Ids - vidurased > -- > > Message: 17 > Date: Fri, 30 Nov 2007 08:58:41 +0530 > From: ram <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] Newb Question > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Message-ID: ><[EMAIL PROTECTED]> > Content-Type: text/plain; charset="iso-8859-1" > > chan spy does the job i belive > > ram > > On Nov 30, 2007 7:37 AM, Jeff Adams <[EMAIL PROTECTED]> wrote: > > > I inherited an office with phones that are hosted off-site. Everything > is > > skinny and G729. I see that the FreeBSD asterisk port comes with a G729 > > codec. > > I want to record everything. If I use port mirroring on my switch, is it > > possible to configure asterisk to record and assemble packets that it > > doesn't otherwise route? Is it insane to user asterisk for this purpose? > > Advice or a link to a howto would be greatly appreciated. > > > > -- > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] get SIP extension status without calling it
I'm using 1.2.6 with the dialplan I posted so I guess the UA you are
using is just plain hosing you.
Anyway, with the queue I believe the music on hold is played to the
inbound side until the call is picked up by an agent. The queue tries
every seconds to get an agent for seconds. If that
fails for however long you set the limit to then the dialplan
continues. I use this to set the priority of the call a little higher
and loop it back into the queue but it could be used for a problem
such as yours just as easily. I don't use the agent login and all
that so I may be talking about something that doesn't apply to your
configuration. If so, sorry for wasting your time.
exten => +1X,1,NoOp(Inbound call from ${CALLERIDNUM})
exten => +1X,n,Answer()
exten => +1X,n,Set(GROUP()=cloud)
exten => +1X,n,Set(QUEUE_PRIO=0)
exten => +1X,n(waiting),Queue(mainline600)
exten => +1X,n,Set(QUEUE_PRIO=$[${QUEUE_PRIO} + 5])
exten => +1X,n,GoTo(waiting)
exten => +1X,n,HangUp
On Dec 2, 2007, at 7:02 PM, Vieri wrote:
I'd like to add that "show hints" on * CLI displays
the following for ext 4053 tested below:
4053: SIP/4053
State:IdleWatchers 0
(it should be "unavailable" or something, but anyway,
ChanIsAvail reports an AVAILSTATUS of 0, ie. unknown)
--- Vieri <[EMAIL PROTECTED]> wrote:
> Thanks Richard but I think that ChanIsAvail must be
> buggy (based on some user comments in the wiki,
> although quite outdated).
>
> I have the hint entry as you say (am using FreePBX
> and
> it's already there).
>
> But whenever I call ChanIsAvail with the s option I
> always get:
> ${AVAILSTATUS} = 0 AST_DEVICE_UNKNOWN - "Unknown";
> channel is valid, but unknown state.
>
> I might be doing something wrong but here is the
> code:
>
> [IVR-menu1]
> exten => s,1,Answer()
> (...)
> exten => s,n,Playback(welcome)
> exten => s,n,ChanIsAvail(SIP/4053|s)
> exten => s,n,NoOp(DEBUG: availstatus is
> ${AVAILSTATUS})
>
> In extensions.conf I also have:
> exten => 4053,hint,SIP/4053
>
> I'm using Astrisk 1.2. Is ChanIsAvail working well
> in
> 1.2?
>
> As far as setting a time limit on a call in the
> queue
> is concerned, it doesn't sound "nice" for the caller
> to be dropped after a few rings when it could have
> been dropped right fom the beginning. It could be a
> solution but it doesn't sound "right" ;-).
>
> Vieri
>
> --- Richard Revels <[EMAIL PROTECTED]> wrote:
>
> > In the sip.conf entry assign a context.
> >
> > In that context, hint the extension i.e. exten =>
> > 7302,hint,SIP/7302.
> >
> > Before you get ready to dial, or whatever, do
> > chanisavail i.e.
> >
> > exten =>
> > _1,n(CheckUse),ChanIsAvail(SIP/${EXTEN:1},js)
> > exten => _1,n,Playback(beep)
> > exten => _1,n,Dial(SIP/${EXTEN},2)
> > exten =>
> > _1,n,Goto(result-${DIALSTATUS},${EXTEN},1)
> > exten => _1,CheckUse+101,SayDigits(${EXTEN:1})
> > exten =>
> _1,CheckUse+102,Playback(vm-isonphone)
> > exten => _1,CheckUse+103,Hangup()
> >
> > This is from the paging stuff. It checks the
> > primary extension before
> > ringing the auto answer extension of the phone. I
> > seem to remember it
> > detecting DND as well for the Cisco 7960.
> >
> > I don't see it in this message but I seem to
> > remember seeing somewhere
> > in this thread that the goal is to keep people
> from
> > being in a queue
> > forever. Why not just set a time limit on the
> queue
> > and play back
> > "all operators busy" and hang up if a call hits
> that
> > limit?
> >
> > Richard
> >
> >
> >
> > On Dec 2, 2007, at 8:51 AM, Vieri wrote:
> >
> > > Hi,
> > >
> > > I am trying to get a SIP extension's status
> > without
> > > actually making a call.
> > >
> > > I am using sofia-sip's "options" example utility
> > and
> > > the sip clients are SJphone softphones.
> > >
> > > From Asterisk I run the "options" utility and
> > query a
> > > sip extension at 10.215.147.240. I get:
> > >
> > > # ./options -1 --all sip:10.215.147.240
> > > SIP/2.0 501 Not Implemented
> > > Via: SIP/2.0/UDP
> > >
> >
>
10.215.144.27:38098;branch=z9hG4bKUKS02S3F8H8ZS;received=10.215.144.27
> > > From: ;tag=U3DKgF7HgFKXH
> > > To: "unknown" ;tag=614733430
> > > Call-ID: b6968197-1b7d-122b-0ab0-00c09f10e472
> > > CSeq: 92182805 OPTIONS
> > > Content-Length: 0
> > > Server: SJphone/1.65.377a (SJ Labs)
> > >
> > > I guess that the softphone should be answering
> > with a
> > > 2xx code followed by a status description?
> > > So I tried with the INVITE method and set DND on
> > the
> > > SIP extension:
> > >
> > > # ./options -1 --all --method INVITE
> > > sip:10.215.147.240
> > > SIP/2.0 486 Busy Here
> > > Via: SIP/2.0/UDP
> > >
> >
>
10.215.144.27:38098;branch=z9hG4bK5Z3BS8F737t0e;received=10.215.144.27
> > > From: ;tag=590Z1ND8B6XpN
> > > To: "unknown"
> ;tag=1a2d77b524
> > > Call-ID: 668ad4fa-1b7e-122b-fcb6-00c09f10e472
> > > CSeq: 92182952 INVITE
> > > Content-Length: 0
> > > Se
Re: [asterisk-users] get SIP extension status without calling it
I'd like to add that "show hints" on * CLI displays
the following for ext 4053 tested below:
4053: SIP/4053
State:IdleWatchers 0
(it should be "unavailable" or something, but anyway,
ChanIsAvail reports an AVAILSTATUS of 0, ie. unknown)
--- Vieri <[EMAIL PROTECTED]> wrote:
> Thanks Richard but I think that ChanIsAvail must be
> buggy (based on some user comments in the wiki,
> although quite outdated).
>
> I have the hint entry as you say (am using FreePBX
> and
> it's already there).
>
> But whenever I call ChanIsAvail with the s option I
> always get:
> ${AVAILSTATUS} = 0 AST_DEVICE_UNKNOWN - "Unknown";
> channel is valid, but unknown state.
>
> I might be doing something wrong but here is the
> code:
>
> [IVR-menu1]
> exten => s,1,Answer()
> (...)
> exten => s,n,Playback(welcome)
> exten => s,n,ChanIsAvail(SIP/4053|s)
> exten => s,n,NoOp(DEBUG: availstatus is
> ${AVAILSTATUS})
>
> In extensions.conf I also have:
> exten => 4053,hint,SIP/4053
>
> I'm using Astrisk 1.2. Is ChanIsAvail working well
> in
> 1.2?
>
> As far as setting a time limit on a call in the
> queue
> is concerned, it doesn't sound "nice" for the caller
> to be dropped after a few rings when it could have
> been dropped right fom the beginning. It could be a
> solution but it doesn't sound "right" ;-).
>
> Vieri
>
> --- Richard Revels <[EMAIL PROTECTED]> wrote:
>
> > In the sip.conf entry assign a context.
> >
> > In that context, hint the extension i.e. exten =>
> > 7302,hint,SIP/7302.
> >
> > Before you get ready to dial, or whatever, do
> > chanisavail i.e.
> >
> > exten =>
> > _1,n(CheckUse),ChanIsAvail(SIP/${EXTEN:1},js)
> > exten => _1,n,Playback(beep)
> > exten => _1,n,Dial(SIP/${EXTEN},2)
> > exten =>
> > _1,n,Goto(result-${DIALSTATUS},${EXTEN},1)
> > exten => _1,CheckUse+101,SayDigits(${EXTEN:1})
> > exten =>
> _1,CheckUse+102,Playback(vm-isonphone)
> > exten => _1,CheckUse+103,Hangup()
> >
> > This is from the paging stuff. It checks the
> > primary extension before
> > ringing the auto answer extension of the phone. I
> > seem to remember it
> > detecting DND as well for the Cisco 7960.
> >
> > I don't see it in this message but I seem to
> > remember seeing somewhere
> > in this thread that the goal is to keep people
> from
> > being in a queue
> > forever. Why not just set a time limit on the
> queue
> > and play back
> > "all operators busy" and hang up if a call hits
> that
> > limit?
> >
> > Richard
> >
> >
> >
> > On Dec 2, 2007, at 8:51 AM, Vieri wrote:
> >
> > > Hi,
> > >
> > > I am trying to get a SIP extension's status
> > without
> > > actually making a call.
> > >
> > > I am using sofia-sip's "options" example utility
> > and
> > > the sip clients are SJphone softphones.
> > >
> > > From Asterisk I run the "options" utility and
> > query a
> > > sip extension at 10.215.147.240. I get:
> > >
> > > # ./options -1 --all sip:10.215.147.240
> > > SIP/2.0 501 Not Implemented
> > > Via: SIP/2.0/UDP
> > >
> >
>
10.215.144.27:38098;branch=z9hG4bKUKS02S3F8H8ZS;received=10.215.144.27
> > > From: ;tag=U3DKgF7HgFKXH
> > > To: "unknown" ;tag=614733430
> > > Call-ID: b6968197-1b7d-122b-0ab0-00c09f10e472
> > > CSeq: 92182805 OPTIONS
> > > Content-Length: 0
> > > Server: SJphone/1.65.377a (SJ Labs)
> > >
> > > I guess that the softphone should be answering
> > with a
> > > 2xx code followed by a status description?
> > > So I tried with the INVITE method and set DND on
> > the
> > > SIP extension:
> > >
> > > # ./options -1 --all --method INVITE
> > > sip:10.215.147.240
> > > SIP/2.0 486 Busy Here
> > > Via: SIP/2.0/UDP
> > >
> >
>
10.215.144.27:38098;branch=z9hG4bK5Z3BS8F737t0e;received=10.215.144.27
> > > From: ;tag=590Z1ND8B6XpN
> > > To: "unknown"
> ;tag=1a2d77b524
> > > Call-ID: 668ad4fa-1b7e-122b-fcb6-00c09f10e472
> > > CSeq: 92182952 INVITE
> > > Content-Length: 0
> > > Server: SJphone/1.65.377a (SJ Labs)
> > >
> > > The above would suit me fine because I get a
> "486
> > Busy
> > > Here" response.
> > > However, if DND is off then I get:
> > >
> > > # ./options -1 --all --method INVITE
> > > sip:10.215.147.240
> > > SIP/2.0 180 Ringing
> > >
> > > and the SIP extension actually "rings", as
> > > expected.(but this is undesireable)
> > >
> > > Now, does someone know another way to get the
> > status
> > > (ie. does it accept calls or not?) without
> making
> > the
> > > extension "ring"?
> > >
> > > Thanks
> > >
> > > Vieri
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Re: [asterisk-users] IAX complaints? What are they?
Ouch. Why do they almost all feel they have to use 'IAX' in their title? Pain. While not half as bad, it's somewhat reminiscent of the iJunk everyone seems to sell to capitalise on the iPhone phenomenon. N. Zoa wrote: > There are many, (i'm one of the people working for zoiper): > Look at the iaxclient homepage, > There are iaxcomm, loudhush, kiax, mediax , diax and many more, > (you could also easily make your own). > > Cheers, > > Zoa > > > Vincent wrote: > >> On Fri, 30 Nov 2007 09:52:59 +0100, randulo <[EMAIL PROTECTED]> >> wrote: >> >> >>> I have used SIP and IAX for about three years now. We don't do a lot >>> of traffic, but I haven't really seen a difference in quality or >>> dropped calls. >>> >>> >> Sorry for jumping in, but besides ZoIPer/Idefisk, are there >> IAX-capable softphones for Windows? >> >> Thanks. >> >> >> ___ >> --Bandwidth and Colocation Provided by http://www.api-digital.com-- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users >> >> > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] get SIP extension status without calling it
Thanks Richard but I think that ChanIsAvail must be
buggy (based on some user comments in the wiki,
although quite outdated).
I have the hint entry as you say (am using FreePBX and
it's already there).
But whenever I call ChanIsAvail with the s option I
always get:
${AVAILSTATUS} = 0 AST_DEVICE_UNKNOWN - "Unknown";
channel is valid, but unknown state.
I might be doing something wrong but here is the code:
[IVR-menu1]
exten => s,1,Answer()
(...)
exten => s,n,Playback(welcome)
exten => s,n,ChanIsAvail(SIP/4053|s)
exten => s,n,NoOp(DEBUG: availstatus is
${AVAILSTATUS})
In extensions.conf I also have:
exten => 4053,hint,SIP/4053
I'm using Astrisk 1.2. Is ChanIsAvail working well in
1.2?
As far as setting a time limit on a call in the queue
is concerned, it doesn't sound "nice" for the caller
to be dropped after a few rings when it could have
been dropped right fom the beginning. It could be a
solution but it doesn't sound "right" ;-).
Vieri
--- Richard Revels <[EMAIL PROTECTED]> wrote:
> In the sip.conf entry assign a context.
>
> In that context, hint the extension i.e. exten =>
> 7302,hint,SIP/7302.
>
> Before you get ready to dial, or whatever, do
> chanisavail i.e.
>
> exten =>
> _1,n(CheckUse),ChanIsAvail(SIP/${EXTEN:1},js)
> exten => _1,n,Playback(beep)
> exten => _1,n,Dial(SIP/${EXTEN},2)
> exten =>
> _1,n,Goto(result-${DIALSTATUS},${EXTEN},1)
> exten => _1,CheckUse+101,SayDigits(${EXTEN:1})
> exten => _1,CheckUse+102,Playback(vm-isonphone)
> exten => _1,CheckUse+103,Hangup()
>
> This is from the paging stuff. It checks the
> primary extension before
> ringing the auto answer extension of the phone. I
> seem to remember it
> detecting DND as well for the Cisco 7960.
>
> I don't see it in this message but I seem to
> remember seeing somewhere
> in this thread that the goal is to keep people from
> being in a queue
> forever. Why not just set a time limit on the queue
> and play back
> "all operators busy" and hang up if a call hits that
> limit?
>
> Richard
>
>
>
> On Dec 2, 2007, at 8:51 AM, Vieri wrote:
>
> > Hi,
> >
> > I am trying to get a SIP extension's status
> without
> > actually making a call.
> >
> > I am using sofia-sip's "options" example utility
> and
> > the sip clients are SJphone softphones.
> >
> > From Asterisk I run the "options" utility and
> query a
> > sip extension at 10.215.147.240. I get:
> >
> > # ./options -1 --all sip:10.215.147.240
> > SIP/2.0 501 Not Implemented
> > Via: SIP/2.0/UDP
> >
>
10.215.144.27:38098;branch=z9hG4bKUKS02S3F8H8ZS;received=10.215.144.27
> > From: ;tag=U3DKgF7HgFKXH
> > To: "unknown" ;tag=614733430
> > Call-ID: b6968197-1b7d-122b-0ab0-00c09f10e472
> > CSeq: 92182805 OPTIONS
> > Content-Length: 0
> > Server: SJphone/1.65.377a (SJ Labs)
> >
> > I guess that the softphone should be answering
> with a
> > 2xx code followed by a status description?
> > So I tried with the INVITE method and set DND on
> the
> > SIP extension:
> >
> > # ./options -1 --all --method INVITE
> > sip:10.215.147.240
> > SIP/2.0 486 Busy Here
> > Via: SIP/2.0/UDP
> >
>
10.215.144.27:38098;branch=z9hG4bK5Z3BS8F737t0e;received=10.215.144.27
> > From: ;tag=590Z1ND8B6XpN
> > To: "unknown" ;tag=1a2d77b524
> > Call-ID: 668ad4fa-1b7e-122b-fcb6-00c09f10e472
> > CSeq: 92182952 INVITE
> > Content-Length: 0
> > Server: SJphone/1.65.377a (SJ Labs)
> >
> > The above would suit me fine because I get a "486
> Busy
> > Here" response.
> > However, if DND is off then I get:
> >
> > # ./options -1 --all --method INVITE
> > sip:10.215.147.240
> > SIP/2.0 180 Ringing
> >
> > and the SIP extension actually "rings", as
> > expected.(but this is undesireable)
> >
> > Now, does someone know another way to get the
> status
> > (ie. does it accept calls or not?) without making
> the
> > extension "ring"?
> >
> > Thanks
> >
> > Vieri
Be a better pen pal.
Text or chat with friends inside Yahoo! Mail. See how.
http://overview.mail.yahoo.com/
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Re: [asterisk-users] Softswitch digim
On Dec 2, 2007 3:42 PM, Carlos Rojas <[EMAIL PROTECTED]> wrote: > > > I'm looking the softswitch in digium website, anyone test the softswitch? > > > Nope. No one has tested it or used it. Try the one at cisco.com. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] get SIP extension status without calling it
In the sip.conf entry assign a context.
In that context, hint the extension i.e. exten => 7302,hint,SIP/7302.
Before you get ready to dial, or whatever, do chanisavail i.e.
exten => _1,n(CheckUse),ChanIsAvail(SIP/${EXTEN:1},js)
exten => _1,n,Playback(beep)
exten => _1,n,Dial(SIP/${EXTEN},2)
exten => _1,n,Goto(result-${DIALSTATUS},${EXTEN},1)
exten => _1,CheckUse+101,SayDigits(${EXTEN:1})
exten => _1,CheckUse+102,Playback(vm-isonphone)
exten => _1,CheckUse+103,Hangup()
This is from the paging stuff. It checks the primary extension before
ringing the auto answer extension of the phone. I seem to remember it
detecting DND as well for the Cisco 7960.
I don't see it in this message but I seem to remember seeing somewhere
in this thread that the goal is to keep people from being in a queue
forever. Why not just set a time limit on the queue and play back
"all operators busy" and hang up if a call hits that limit?
Richard
On Dec 2, 2007, at 8:51 AM, Vieri wrote:
Hi,
I am trying to get a SIP extension's status without
actually making a call.
I am using sofia-sip's "options" example utility and
the sip clients are SJphone softphones.
From Asterisk I run the "options" utility and query a
sip extension at 10.215.147.240. I get:
# ./options -1 --all sip:10.215.147.240
SIP/2.0 501 Not Implemented
Via: SIP/2.0/UDP
10.215.144.27:38098;branch=z9hG4bKUKS02S3F8H8ZS;received=10.215.144.27
From: ;tag=U3DKgF7HgFKXH
To: "unknown" ;tag=614733430
Call-ID: b6968197-1b7d-122b-0ab0-00c09f10e472
CSeq: 92182805 OPTIONS
Content-Length: 0
Server: SJphone/1.65.377a (SJ Labs)
I guess that the softphone should be answering with a
2xx code followed by a status description?
So I tried with the INVITE method and set DND on the
SIP extension:
# ./options -1 --all --method INVITE
sip:10.215.147.240
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP
10.215.144.27:38098;branch=z9hG4bK5Z3BS8F737t0e;received=10.215.144.27
From: ;tag=590Z1ND8B6XpN
To: "unknown" ;tag=1a2d77b524
Call-ID: 668ad4fa-1b7e-122b-fcb6-00c09f10e472
CSeq: 92182952 INVITE
Content-Length: 0
Server: SJphone/1.65.377a (SJ Labs)
The above would suit me fine because I get a "486 Busy
Here" response.
However, if DND is off then I get:
# ./options -1 --all --method INVITE
sip:10.215.147.240
SIP/2.0 180 Ringing
and the SIP extension actually "rings", as
expected.(but this is undesireable)
Now, does someone know another way to get the status
(ie. does it accept calls or not?) without making the
extension "ring"?
Thanks
Vieri
Be a better pen pal.
Text or chat with friends inside Yahoo! Mail. See how.
http://overview.mail.yahoo.com/
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Re: [asterisk-users] get SIP extension status without calling it
--- Raj Jain <[EMAIL PROTECTED]> wrote: > In theory, UAs that respond to OPTIONS and INVITE > differently are "broken". > Below is a quote from section 11.2 of RFC 3261. > >The response to an OPTIONS is constructed using > the standard rules >for a SIP response as discussed in Section 8.2.6. > The response code >chosen MUST be the same that would have been > chosen had the request >been an INVITE. That is, a 200 (OK) would be > returned if the UAS is >ready to accept a call, a 486 (Busy Here) would > be returned if the >UAS is busy, etc. This allows an OPTIONS request > to be used to >determine the basic state of a UAS, which can be > an indication of >whether the UAS will accept an INVITE request. > > In practice, as you're seeing it yourself most UA > implementations treat > OPTIONS as a health-check and capability discovery > mechanism. Thank you for explaining Raj. It sounds depressing though. Looks like all major UAs have the same behavior and don't comply to section 11.2 of RFC 3261 (tested Grandstream, SJphone and X-Lite). If someone knows of a UA that actually does comply please let me know. Thanks. Be a better sports nut! Let your teams follow you with Yahoo Mobile. Try it now. http://mobile.yahoo.com/sports;_ylt=At9_qDKvtAbMuh1G1SQtBI7ntAcJ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] setting up two asterisk server as ss7 back to back.
I have used asterisk-1.4.14, zaptel-1.4.7, chan_ss7-1.0.0 on FC7 all went okay. using sangoma a104dx on both machine. I followed the write up on http://www.voip-info.org/wiki/index.php?page=Asterisk+ss7+setup I have the cross over cable between them. however, wanpipe shows connected but the signaling link does not align. i have my configs for host A ##wanpipe1.conf [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort= PRI AUTO_PCISLOT= NO PCISLOT = 4 PCIBUS = 15 FE_MEDIA= E1 FE_LCODE= HDB3 FE_FRAME= NCRC4 FE_LINE = 1 TE_CLOCK= MASTER TE_REF_CLOCK= 0 TE_HIGHIMPEDANCE= NO LBO = 120OH TE_SIG_MODE = CCS FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 0 [w1g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = YES # zaptel.conf loadzone=uk defaultzone=uk span=1,1,0,ccs,hdb3 bchan=1-31 ### ss7.conf [linkset-ennie] enabled => yes use_connect => yes enable_st => no hunting_policy => odd_lru subservice => auto context = default [link-l1] linkset => ennie channels => 1-15,17-31 schannel => 16 firstcic => 1 enabled => yes [link-l2] linkset => ennie channels => 1-31 schannel => firstcic => 33 enabled => yes [link-l3] linkset => ennie channels => 1-31 schannel => firstcic => 65 enabled => no [link-l4] linkset => ennie channels => 1-31 schannel => firstcic => 97 enabled => no [host-A] enabled => yes opc => 998 dpc => ennie:98 links => l1:1 for the slave host B ##wanpipe1.conf [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort= PRI AUTO_PCISLOT= NO PCISLOT = 4 PCIBUS = 15 FE_MEDIA= E1 FE_LCODE= HDB3 FE_FRAME= NCRC4 FE_LINE = 1 TE_CLOCK= NORMAL TE_REF_CLOCK= 0 TE_HIGHIMPEDANCE= NO LBO = 120OH TE_SIG_MODE = CCS FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 0 [w1g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = YES # zaptel.conf loadzone=uk defaultzone=uk span=1,0,0,ccs,hdb3 bchan=1-31 ### ss7.conf [linkset-mahmud] enabled => yes use_connect => yes enable_st => no hunting_policy => odd_lru subservice => auto context = default [link-l1] linkset => mahmud channels => 1-15,17-31 schannel => 16 firstcic => 1 enabled => yes [link-l2] linkset => mahmud channels => 1-31 schannel => firstcic => 33 enabled => no [link-l3] linkset => mahmud channels => 1-31 schannel => firstcic => 65 enabled => no [link-l4] linkset => mahmud channels => 1-31 schannel => firstcic => 97 enabled => no [host-B] enabled => yes opc => 98 dpc => mahmud:998 links => l1:1 Can someone advice on the way forward or has anyone implemented chan_ss7-1.0.0. Thanks Goksie ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX complaints? What are they?
There are many, (i'm one of the people working for zoiper): Look at the iaxclient homepage, There are iaxcomm, loudhush, kiax, mediax , diax and many more, (you could also easily make your own). Cheers, Zoa Vincent wrote: > On Fri, 30 Nov 2007 09:52:59 +0100, randulo <[EMAIL PROTECTED]> > wrote: > >> I have used SIP and IAX for about three years now. We don't do a lot >> of traffic, but I haven't really seen a difference in quality or >> dropped calls. >> > > Sorry for jumping in, but besides ZoIPer/Idefisk, are there > IAX-capable softphones for Windows? > > Thanks. > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Softswitch digim
Hello averybody, I'm looking the softswitch in digium website, anyone test the softswitch? Best Regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] get SIP extension status without calling it
In theory, UAs that respond to OPTIONS and INVITE differently are "broken". Below is a quote from section 11.2 of RFC 3261. The response to an OPTIONS is constructed using the standard rules for a SIP response as discussed in Section 8.2.6. The response code chosen MUST be the same that would have been chosen had the request been an INVITE. That is, a 200 (OK) would be returned if the UAS is ready to accept a call, a 486 (Busy Here) would be returned if the UAS is busy, etc. This allows an OPTIONS request to be used to determine the basic state of a UAS, which can be an indication of whether the UAS will accept an INVITE request. In practice, as you're seeing it yourself most UA implementations treat OPTIONS as a health-check and capability discovery mechanism. - Raj > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Vieri > Sent: Sunday, December 02, 2007 12:57 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] get SIP extension status > without calling it > > I tried another popular user agent: X-Lite. > > I dialed *78 which in */FreePBX turns DND on AND I pushed the > DND button on the softphone. > > # asterisk -vvvr > CLI> database show dnd > /DND/4053 : > YES > > Despite all this when I send an OPTIONS request I always get > a "200 ok" reply. > > Is X-Lite also "broken" with respect to the SIP RFC? > Or am I doing things wrong? > > # ./options -1 -a --method OPTIONS > sip:[EMAIL PROTECTED]:6486 > SIP/2.0 200 OK > Via: SIP/2.0/UDP > 10.215.144.27:38102;branch=z9hG4bKZDm0j0KD5BSBQ > Contact: > To: ;tag=681c6278 > From: ;tag=Z1QHmBt52Dp1Q > Call-ID: 6b9f7f35-1ba1-122b-d4b7-00c09f10e472 > CSeq: 92190473 OPTIONS > Accept: application/sdp > Accept-Language: en > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, > MESSAGE, SUBSCRIBE, INFO > User-Agent: X-Lite release 1011s stamp 41150 > Content-Length: 0 > > > CLI> sip show peer 4053 > INF-VOIP*CLI> > > * Name : 4053 > Secret : > MD5Secret: > Context : from-internal > Subscr.Cont. : > Language : es > AMA flags: Unknown > CallingPres : Presentation Allowed, Not Screened > Callgroup: 2 > Pickupgroup : 2 > Mailbox : [EMAIL PROTECTED] > VM Extension : asterisk > LastMsgsSent : 0/0 > Call limit : 0 > Dynamic : Yes > Callerid : "device" <4053> > Expire : 3597 > Insecure : no > Nat : Always > ACL : No > CanReinvite : No > PromiscRedir : No > User=Phone : No > Trust RPID : No > Send RPID: No > DTMFmode : rfc2833 > LastMsg : 0 > ToHost : > Addr->IP : 10.215.147.240 Port 6486 > Defaddr->IP : 0.0.0.0 Port 5060 > Def. Username: 4053 > SIP Options : (none) > Codecs : 0x400 (ilbc) > Codec Order : (ilbc) > Status : OK (169 ms) > Useragent: X-Lite release 1011s stamp 41150 > Reg. Contact : > sip:[EMAIL PROTECTED]:6486;rinstance=ff64e47c4f35bdef > > > > > __ > __ > Never miss a thing. Make Yahoo your home page. > http://www.yahoo.com/r/hs > > > > __ > __ > Be a better pen pal. > Text or chat with friends inside Yahoo! Mail. See how. > http://overview.mail.yahoo.com/ > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX complaints? What are they?
On Fri, 30 Nov 2007 09:52:59 +0100, randulo <[EMAIL PROTECTED]> wrote: >I have used SIP and IAX for about three years now. We don't do a lot >of traffic, but I haven't really seen a difference in quality or >dropped calls. Sorry for jumping in, but besides ZoIPer/Idefisk, are there IAX-capable softphones for Windows? Thanks. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk install beta testing/config help
Make certain that selinux, iptables and ip6tables are disabled and off. Bryan M. Johns Shelton | Johns Office: 678.248.2637 FindMe: 678.229.1809 Support: [EMAIL PROTECTED] http://www.sheltonjohns.com On Dec 2, 2007, at 3:18 PM, James Cox wrote: > I have asterisk up and running on a fedora system but > having trouble accessing system via softphone (ekiga > and xlite). Im a newbie to asterisk and was looking > for some help walking through this. I imagine 10 - 15 > mins would be all needed to make proper config changes > needed. Once I get this setup I'd be interested in > discussing customizations and scripts so any > freelancers or companies welcome since the sooner i > get this working the sooner can move to that next > stage. thanks in advance! > > My yahoo IM is jameswcox2001 > > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk install beta testing/config help
I have asterisk up and running on a fedora system but having trouble accessing system via softphone (ekiga and xlite). Im a newbie to asterisk and was looking for some help walking through this. I imagine 10 - 15 mins would be all needed to make proper config changes needed. Once I get this setup I'd be interested in discussing customizations and scripts so any freelancers or companies welcome since the sooner i get this working the sooner can move to that next stage. thanks in advance! My yahoo IM is jameswcox2001 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] When calling in via AGI, gsm sound file plays but sometimes drops out
Hi. I am using the 'get_data' function from an AGI, and i find that sometimes when users call in, it won't play the full gsm soundfile, and when i try to press a number (or pound, or star), nothing will happen - it just hangs there... anyone else experience this? - Dominic Son "It is not the force of a stroke that makes fine art" ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is the status and future of chan_mobile
On Sunday 02 December 2007 12:22:23 Robert Moskowitz wrote: > I have been looking forward for months to get chan_mobile working. I am > limited to using prepackaged Asterisk code, mostly Trixbox. > > I have recently heard that chan_mobile is considered 'beta' and there is > no effort to move it into the main code of Asterisk. Not even for > Asterisk 1.6. I'm not sure where you heard that, but it is not true. It is merely segregated into asterisk-addons, due to the licensing on the library on which it depends. > So what is the future for chan_mobile? It is easy to make a very strong > case for a cellular trunk. It's already in SVN, which will become asterisk-addons-1.6. -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] T1 Timing Troubleshooting
I'm having (I think) timing issues in relation to bridged T1-T1 calls via dynamic spans. Fax calls are intermittently working, but voice is fine. My box has a Sangoma A400 inside it as the primary Zaptel timing source. My T1 PRIs that are hooked to the box come in via a foneBRIDGE2 (dynamic TDMoE spans). PRI #1 is the telco and PRI #2 is an existing Comdial FX-II. For some reason, bridged TDM calls (when it comes to faxing) must be having timing issues since they intermittently fail. I found what seems to be an issue in zaptel.conf (timing source for the Comdial side was 2 - changed to 0), but I don't know if that's it. I've also turned off echo cancellation. Any other thoughts on why I may be having what seem to be timing issues? Also, is timing passed through on dynamic spans & bridged calls? And is there a way to verify this? Thanks! - /etc/zaptel.conf (16 channels on each PRI): loadzone=us defaultzone=us #Sangoma A400 [slot:7 bus:1 span:1] fxsks=1 fxsks=2 fxsks=3 fxsks=4 fxsks=5 fxsks=6 fxsks=7 fxsks=8 fxoks=11 fxoks=12 dynamic=eth,eth1/00:50:c2:65:d0:3c/0,24,1 dynamic=eth,eth1/00:50:c2:65:d0:3c/1,24,0 # bchan=25-47 bchan=25-40 dchan=48 # bchan=49-71 bchan=49-64 dchan=72 - /etc/asterisk/zapata.conf: [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes ; Turned echo cancellation off 11-15-2007 due to possible fax issues on bridged calls. echocancel=no faxdetect=no echocancelwhenbridged=no rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no overlapdial=yes ;Sangoma A400 [slot:7 bus:1 span:1] context=from-zaptel group=0 signalling = fxs_ks channel => 1-8 context=from-internal group=1 signalling = fxo_ks channel => 11-12 ; First port on foneBRIDGE2 - This is the PSTN side group=2 signalling = pri_cpe context=from-pstn ;channel => 25-47 (for a full PRI) ; Channels 25-40 are for a partial PRI (16 channels) channel => 25-40 ; Second port on foneBRIDGE2 - This is the Comdial side group=3 context=from-comdial signalling = pri_net ;channel => 49-71 (for a full PRI) ; Channels 49-64 are for a partial PRI (16 channels) channel => 49-64 -Jon ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cross-compiling asterisk-1.4 for Debian on a slug
Fabiano Sidler wrote: > I'll probably post this again on the slug mailing list, when that seems > more appropriate for this topic. I guess you already know this page http://www.voip-info.org/wiki/view/Asterisk+Linksys+NSLU2 Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What is the status and future of chan_mobile
I have been looking forward for months to get chan_mobile working. I am limited to using prepackaged Asterisk code, mostly Trixbox. I have recently heard that chan_mobile is considered 'beta' and there is no effort to move it into the main code of Asterisk. Not even for Asterisk 1.6. So what is the future for chan_mobile? It is easy to make a very strong case for a cellular trunk. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] get SIP extension status without calling it
I tried another popular user agent: X-Lite. I dialed *78 which in */FreePBX turns DND on AND I pushed the DND button on the softphone. # asterisk -vvvr CLI> database show dnd /DND/4053 : YES Despite all this when I send an OPTIONS request I always get a "200 ok" reply. Is X-Lite also "broken" with respect to the SIP RFC? Or am I doing things wrong? # ./options -1 -a --method OPTIONS sip:[EMAIL PROTECTED]:6486 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.215.144.27:38102;branch=z9hG4bKZDm0j0KD5BSBQ Contact: To: ;tag=681c6278 From: ;tag=Z1QHmBt52Dp1Q Call-ID: 6b9f7f35-1ba1-122b-d4b7-00c09f10e472 CSeq: 92190473 OPTIONS Accept: application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 CLI> sip show peer 4053 INF-VOIP*CLI> * Name : 4053 Secret : MD5Secret: Context : from-internal Subscr.Cont. : Language : es AMA flags: Unknown CallingPres : Presentation Allowed, Not Screened Callgroup: 2 Pickupgroup : 2 Mailbox : [EMAIL PROTECTED] VM Extension : asterisk LastMsgsSent : 0/0 Call limit : 0 Dynamic : Yes Callerid : "device" <4053> Expire : 3597 Insecure : no Nat : Always ACL : No CanReinvite : No PromiscRedir : No User=Phone : No Trust RPID : No Send RPID: No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr->IP : 10.215.147.240 Port 6486 Defaddr->IP : 0.0.0.0 Port 5060 Def. Username: 4053 SIP Options : (none) Codecs : 0x400 (ilbc) Codec Order : (ilbc) Status : OK (169 ms) Useragent: X-Lite release 1011s stamp 41150 Reg. Contact : sip:[EMAIL PROTECTED]:6486;rinstance=ff64e47c4f35bdef Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs Be a better pen pal. Text or chat with friends inside Yahoo! Mail. See how. http://overview.mail.yahoo.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dictaphone Freedom interface to Asterisk ABE
Hello list, I am trying to find a solution for interfacing a Dictaphone Freedom recorder. Currently, 4 POTS lines interface to the recorder, and the future will have the 4 lines coming into an ABE server on PRI. The Freedom system is using a standard amphinol connector to a punch down block, where the 4 lines, shared with a Nortel Meridian system, are located. The lines are analog, and per NICE, the Freedom system is a passive resident recording any traffic on the lines. I need to interface this with an FXO or FXS card in the ABE server, and allow all calls on these 4 numbers to be "seen" by the card to the Dictaphone channels. Thanks in advance for any suggestions. Regards, Paul ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] get SIP extension status without calling it
I also tried:
# ./options -1 -a sip:[EMAIL PROTECTED]:5072
but still received a
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.215.144.27:38102;branch=z9hG4bKFB4rQrr5aXp9H
From: ;tag=FF1tQy74X81rm
To: ;tag=1639856599
Call-ID: 83259dd2-1b9e-122b-10a3-00c09f10e472
CSeq: 92189848 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXW-4008 V1.2A 1.0.0.67
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE,
NOTIFY, INFO, REFER, UPDATE
Content-Length: 0
Is Grandstream GXW-4008 not 100% SIP compliant, too?
--- Vieri <[EMAIL PROTECTED]> wrote:
>
> --- Tilghman Lesher
> <[EMAIL PROTECTED]> wrote:
>
> > OPTIONS is the right type of request
>
> Suppose that the user agent is not a softphone but a
> gateway such as the Grandstream GXW-4008 ATA.
> One of the FXS-port-connected phones of the gateway
> has DND turned on.
> IF I send an OPTIONS request then the UA always
> answers with a 200 ok even if the extension is
> actually busy.
>
> In this particular case, how does one get the status
> of an extension "behind a user agent/gateway"? (am I
> writing the sip url correctly in the sofia-sip
> options
> utility below?)
>
> ("database show dnd" does not yield anything)
>
> # ./options -1 -a sip:[EMAIL PROTECTED]
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 10.215.144.27:38102;branch=z9hG4bKF8F0cQa7N9cva
> From: ;tag=Fcj2gFcX9ctpg
> To: ;tag=1260250638
> Call-ID: 3cb23664-1b9c-122b-6393-00c09f10e472
> CSeq: 92189360 OPTIONS
> Supported: replaces, path, timer
> User-Agent: Grandstream GXW-4008 V1.2A 1.0.0.67
> Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE,
> NOTIFY, INFO, REFER, UPDATE
> Content-Length: 0
>
>
> On the asterisk cli:
>
> CLI> sip show peers
> Name/username HostDyn Nat
> ACL
> Port Status
> / 192.168.250.1D N
>
> 5060 OK (8 ms)
> 4173/4173 (Unspecified)D N
>
> 0UNKNOWN
> 4172/4172 (Unspecified)D N
>
> 0Unmonitored
> 4171/4171 (Unspecified)D N
>
> 0Unmonitored
> 4170/4170 (Unspecified)D N
>
> 0Unmonitored
> 4065/4065 192.168.250.1D N
>
> 6074 OK (8 ms)
> 4064/4064 192.168.250.1D N
>
> 5072 OK (8 ms)
> 4063/4063 192.168.250.1D N
>
> 5068 OK (8 ms)
> 4061/4061 192.168.250.1D N
>
> 5070 OK (8 ms)
> 4059/4059 (Unspecified)D N
>
> 0Unmonitored
> 4058/4058 (Unspecified)D N
>
> 0UNKNOWN
> 4057/4057 (Unspecified)D N
>
> 0UNKNOWN
> 4056/4056 (Unspecified)D N
>
> 0UNKNOWN
> 4055/4055 (Unspecified)D N
>
> 0Unmonitored
> 4054/4054 (Unspecified)D N
>
> 0UNKNOWN
> 4053/4053 10.215.147.240 D N
>
> 5060 OK (139 ms)
> 4052/4052 (Unspecified)D N
>
> 0Unmonitored
> 4022/4022 192.168.250.1D N
>
> 5062 OK (8 ms)
> 4013/4013 (Unspecified)D N
>
> 0UNKNOWN
> 4012/4012 (Unspecified)D N
>
> 0UNKNOWN
> 4004/4004 (Unspecified)D N
>
> 0UNKNOWN
> 4003/4003 10.215.145.170 D N
>
> 5060 OK (1 ms)
> 4002/4002 (Unspecified)D N
>
> 0UNKNOWN
> 23 sip peers [14 online , 9 offline]
>
> CLI> sip show peer 4064
> CLI>
>
> * Name : 4064
> Secret :
> MD5Secret:
> Context : from-internal
> Subscr.Cont. :
> Language : es
> AMA flags: Unknown
> CallingPres : Presentation Allowed, Not Screened
> Callgroup: 2
> Pickupgroup : 2
> Mailbox : [EMAIL PROTECTED]
> VM Extension : asterisk
> LastMsgsSent : 0/0
> Call limit : 0
> Dynamic : Yes
> Callerid : "device" <4064>
> Expire : 22569
> Insecure : no
> Nat : Always
> ACL : No
> CanReinvite : No
> PromiscRedir : No
> User=Phone : No
> Trust RPID : No
> Send RPID: No
> DTMFmode : rfc2833
> LastMsg : 0
> ToHost :
> Addr->IP : 192.168.250.1 Port 5072
> Defaddr->IP : 0.0.0.0 Port 5060
> Def. Username: 4064
> SIP Options : (none)
> Codecs : 0x400 (ilbc)
> Codec Order : (ilbc)
> Status : OK (8 ms)
> Useragent: Grandstream GXW-4008 V1.2A
> 1.0.0.67
> Reg. Contact : sip:[EMAIL PROTECTED]:5072
Never miss a thing. Make Yahoo your home page.
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aster
Re: [asterisk-users] get SIP extension status without calling it
--- Tilghman Lesher
<[EMAIL PROTECTED]> wrote:
> OPTIONS is the right type of request
Suppose that the user agent is not a softphone but a
gateway such as the Grandstream GXW-4008 ATA.
One of the FXS-port-connected phones of the gateway
has DND turned on.
IF I send an OPTIONS request then the UA always
answers with a 200 ok even if the extension is
actually busy.
In this particular case, how does one get the status
of an extension "behind a user agent/gateway"? (am I
writing the sip url correctly in the sofia-sip options
utility below?)
("database show dnd" does not yield anything)
# ./options -1 -a sip:[EMAIL PROTECTED]
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.215.144.27:38102;branch=z9hG4bKF8F0cQa7N9cva
From: ;tag=Fcj2gFcX9ctpg
To: ;tag=1260250638
Call-ID: 3cb23664-1b9c-122b-6393-00c09f10e472
CSeq: 92189360 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXW-4008 V1.2A 1.0.0.67
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE,
NOTIFY, INFO, REFER, UPDATE
Content-Length: 0
On the asterisk cli:
CLI> sip show peers
Name/username HostDyn Nat ACL
Port Status
/ 192.168.250.1D N
5060 OK (8 ms)
4173/4173 (Unspecified)D N
0UNKNOWN
4172/4172 (Unspecified)D N
0Unmonitored
4171/4171 (Unspecified)D N
0Unmonitored
4170/4170 (Unspecified)D N
0Unmonitored
4065/4065 192.168.250.1D N
6074 OK (8 ms)
4064/4064 192.168.250.1D N
5072 OK (8 ms)
4063/4063 192.168.250.1D N
5068 OK (8 ms)
4061/4061 192.168.250.1D N
5070 OK (8 ms)
4059/4059 (Unspecified)D N
0Unmonitored
4058/4058 (Unspecified)D N
0UNKNOWN
4057/4057 (Unspecified)D N
0UNKNOWN
4056/4056 (Unspecified)D N
0UNKNOWN
4055/4055 (Unspecified)D N
0Unmonitored
4054/4054 (Unspecified)D N
0UNKNOWN
4053/4053 10.215.147.240 D N
5060 OK (139 ms)
4052/4052 (Unspecified)D N
0Unmonitored
4022/4022 192.168.250.1D N
5062 OK (8 ms)
4013/4013 (Unspecified)D N
0UNKNOWN
4012/4012 (Unspecified)D N
0UNKNOWN
4004/4004 (Unspecified)D N
0UNKNOWN
4003/4003 10.215.145.170 D N
5060 OK (1 ms)
4002/4002 (Unspecified)D N
0UNKNOWN
23 sip peers [14 online , 9 offline]
CLI> sip show peer 4064
CLI>
* Name : 4064
Secret :
MD5Secret:
Context : from-internal
Subscr.Cont. :
Language : es
AMA flags: Unknown
CallingPres : Presentation Allowed, Not Screened
Callgroup: 2
Pickupgroup : 2
Mailbox : [EMAIL PROTECTED]
VM Extension : asterisk
LastMsgsSent : 0/0
Call limit : 0
Dynamic : Yes
Callerid : "device" <4064>
Expire : 22569
Insecure : no
Nat : Always
ACL : No
CanReinvite : No
PromiscRedir : No
User=Phone : No
Trust RPID : No
Send RPID: No
DTMFmode : rfc2833
LastMsg : 0
ToHost :
Addr->IP : 192.168.250.1 Port 5072
Defaddr->IP : 0.0.0.0 Port 5060
Def. Username: 4064
SIP Options : (none)
Codecs : 0x400 (ilbc)
Codec Order : (ilbc)
Status : OK (8 ms)
Useragent: Grandstream GXW-4008 V1.2A 1.0.0.67
Reg. Contact : sip:[EMAIL PROTECTED]:5072
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Re: [asterisk-users] get SIP extension status without calling it
Thanks for the "sip show peers" script, Dave.
But that won't work for me.
It won't tell me whether the extension will actually
accept a call or not (eg. if DND is ON only on the
"client side").
This link might clarify the problem I am facing:
http://lists.digium.com/pipermail/asterisk-users/2007-September/195936.html
and the following links discuss a way to determine an
extension's DND state in order to use the
{Add,Remove}QueueMember function efficiently from a
custom cron script.
http://lists.digium.com/pipermail/asterisk-users/2007-September/196345.html
http://lists.digium.com/pipermail/asterisk-users/2007-September/196437.html
The need to determine if an extension accepts calls or
not (and what's missing here is to detect DND on/off
on the client side) is related to queues and agents.
Basically, if, say, all agents are in the queue but
have DND on then what I need is to bail the caller out
because it doesn't make much sense from a practical
point of view to have he/she wait "forever" for an
agent to turn DND off.
Maybe it's a big limitation in SIP protocol but I'd
like to know if other users have found a viable, open
source solution.
--- dave cantera <[EMAIL PROTECTED]> wrote:
> vieri,
> you can get sip status with the following shell
> script... I named it
> 'sipshowpeer'...
> Vieri wrote:
> > Hi,
> >
> > I am trying to get a SIP extension's status
> without
> > actually making a call.
> >
> > I am using sofia-sip's "options" example utility
> and
> > the sip clients are SJphone softphones.
> >
> > >From Asterisk I run the "options" utility and
> query a
> > sip extension at 10.215.147.240. I get:
> >
> > # ./options -1 --all sip:10.215.147.240
> > SIP/2.0 501 Not Implemented
> > Via: SIP/2.0/UDP
> >
>
10.215.144.27:38098;branch=z9hG4bKUKS02S3F8H8ZS;received=10.215.144.27
> > From: ;tag=U3DKgF7HgFKXH
> > To: "unknown" ;tag=614733430
> > Call-ID: b6968197-1b7d-122b-0ab0-00c09f10e472
> > CSeq: 92182805 OPTIONS
> > Content-Length: 0
> > Server: SJphone/1.65.377a (SJ Labs)
> >
> > I guess that the softphone should be answering
> with a
> > 2xx code followed by a status description?
> > So I tried with the INVITE method and set DND on
> the
> > SIP extension:
> >
> > # ./options -1 --all --method INVITE
> > sip:10.215.147.240
> > SIP/2.0 486 Busy Here
> > Via: SIP/2.0/UDP
> >
>
10.215.144.27:38098;branch=z9hG4bK5Z3BS8F737t0e;received=10.215.144.27
> > From: ;tag=590Z1ND8B6XpN
> > To: "unknown" ;tag=1a2d77b524
> > Call-ID: 668ad4fa-1b7e-122b-fcb6-00c09f10e472
> > CSeq: 92182952 INVITE
> > Content-Length: 0
> > Server: SJphone/1.65.377a (SJ Labs)
> >
> > The above would suit me fine because I get a "486
> Busy
> > Here" response.
> > However, if DND is off then I get:
> >
> > # ./options -1 --all --method INVITE
> > sip:10.215.147.240
> > SIP/2.0 180 Ringing
> >
> > and the SIP extension actually "rings", as
> > expected.(but this is undesireable)
> >
> > Now, does someone know another way to get the
> status
> > (ie. does it accept calls or not?) without making
> the
> > extension "ring"?
> >
> > Thanks
> >
> > Vieri
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Re: [asterisk-users] get SIP extension status without calling it
--- Tilghman Lesher <[EMAIL PROTECTED]> wrote: > To the original poster: OPTIONS is the right type > of request (the client > should respond exactly as the same way as if you had > sent it an INVITE). > Your next step should be to contact the author of > the softphone and impress > upon them the necessity of having OPTIONS > implemented in their client. > Or use another softphone. Thank you for clarifying, Tilghman. Now I know that the client UA is "not 100% SIP compliant". Great. Be a better sports nut! Let your teams follow you with Yahoo Mobile. Try it now. http://mobile.yahoo.com/sports;_ylt=At9_qDKvtAbMuh1G1SQtBI7ntAcJ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Answering Machine Detection
If i use AMD() or the code below, now the problem is the fax machine/modem
detection and answer machine detection get detected as the same. If i need to
seperate the two how do i do that? For example, if i use AMD() to detect an
answer machine by saying any greeting exceeding 2.5 seconds is a machine, how
do i distinguish between a fax/modem and a long greeting?
dave cantera <[EMAIL PROTECTED]> wrote:
> carlos,
> you got further than I did... AMD didn't work at all on my release.. I
> think I was using 1.4.11 at the time...
> I ended up using the below
> daveC
>
> ;---< amdtest (ext 13) starts here >
> ;
> ; restructure this for the following conditions:
> ; 13 using waitforsilence(variable set) then play message
> ; only works when there is an answering machine picking up and it doesn't
> ; cut off (hangup) before SILENCEDURATION ms
> exten => 13,1,NoOp( Starting exten 13 AMD stuff)
> exten => 13,n,Wait(1)
> exten => 13,n,Set(SILENCEDURATION=4300)
> exten => 13,n,Set(SILENCEOCCURANCES="")
> exten => 13,n,Set(SILENCETIMEOUT=38)
> exten => 13,n,NoOp( SILENCEDURATION=${SILENCEDURATION} )
> exten => 13,n,NoOp( SILENCEOCCURANCES=${SILENCEOCCURANCES} )
> exten => 13,n,NoOp( SILENCETIMEOUT=${SILENCETIMEOUT} )
> ;exten => 13,n,Answer
> exten =>
> 13,n,WaitForSilence(${SILENCEDURATION},${SILENCEOCCURANCES},${SILENCETIMEOUT})
> exten => 13,n,NoOp(Retnd WAITSTATUS=${WAITSTATUS} )
> exten => 13,n,Playback(lax/lax-important-msg-from)
> exten => 13,n,PlayBack(lax/to-hear-msg-press-1)
> exten => 13,n,Read(CALL_ACK,beep,1,,,3)
> exten => 13,n,NoOp(CALL_ACK is >${CALL_ACK}<)
> exten => 13,n,GotoIf([${CALL_ACK} = ""]?nak13)
> exten => 13,n(ack13),NoOp( Ack )
> exten => 13,n,NoOp( log the ACK acknowlegement here calling the AGI script)
> ;exten => 13,n,AGI(lax/track-laxcalls.sh,${EXTEN},${CALL_ACK})
> exten => 13,n,GotoIf([${CALL_ACK} = ""]?play13)
> exten => 13,n(nak13),NoOp( Nak )
> exten => 13,n,NoOp( log the NAK acknowlegement here calling the AGI script)
> ;exten => 13,n,AGI(lax/track-laxcalls.sh,${EXTEN},${CALL_ACK})
> exten => 13,n(play13),Playback(lax/lax-important-msg-from)
> exten => 13,n,Playback(tt-weasels)
> exten => 13,n,Playback(tt-monkeysintro)
> exten => 13,n,Wait(1)
> exten => 13,n,Hangup
>
>
>
>
>
> Carlos Chavez wrote:
> > I am having a bit of a problem getting AMD to work on a new server. On
> > my regular office server it works like a charm. I am running Asterisk
> > 1.4.13, Zaptel 1.4.5.1 on both machines. Both servers run CentOS 5 and
> > I am using a SIP trunk to send out calls (the same one on both servers).
> >
> > Here is the output of a call on my office server:
> >
> > -- Attempting call on Local/[EMAIL PROTECTED] for [EMAIL PROTECTED]:1
> > (Retry
> > 1)
> > -- Executing [EMAIL PROTECTED]:1]
> > Set("Local/[EMAIL PROTECTED],2", "CIDTEMP="1" <5540881644>") in new
> > stack
> > -- Executing [EMAIL PROTECTED]:2]
> > Dial("Local/[EMAIL PROTECTED],2", "SIP/protel-out/0445540881644|
> > 25") in new stack
> > -- Called protel-out/0445540881644
> > -- SIP/protel-out-0934bb28 is making progress passing it to
> > Local/[EMAIL PROTECTED],2
> > -- SIP/protel-out-0934bb28 answered Local/[EMAIL PROTECTED],2
> > -- Executing [EMAIL PROTECTED]:1] NoOp("Local/[EMAIL PROTECTED],1", ""1"
> > <5540881644>") in new stack
> > -- Executing [EMAIL PROTECTED]:2] AMD("Local/[EMAIL PROTECTED],1", "")
> > in new stack
> > -- AMD: Local/[EMAIL PROTECTED],1 5540881644 (null) (Fmt: 64)
> > -- AMD: initialSilence [2500] greeting [1500] afterGreetingSilence
> > [800] totalAnalysisTime [5000] minimumWordLength [100]
> > betweenWordsSilence [50] maximumNumberOfWords [5] silenceThreshold
> > [256]
> > == Spawn extension (CC2, 0445540881644, 2) exited non-zero on
> > 'Local/[EMAIL PROTECTED],2'
> > -- Executing [EMAIL PROTECTED]:1] DeadAGI("Local/[EMAIL PROTECTED],2",
> > "agi://localhost/updateCallStatus.agi?callStatus=hangupcc2") in new
> > stack
> > -- AGI Script
> > agi://localhost/updateCallStatus.agi?callStatus=hangupcc2 completed,
> > returning 0
> > -- AMD: Word detected. iWordsCount:1
> > -- AMD: Changed state to STATE_IN_SILENCE
> > -- AMD: HUMAN: silenceDuration:800 afterGreetingSilence:800
> > -- Executing [EMAIL PROTECTED]:3] GotoIf("SIP/protel-out-0934bb28",
> > "1?7:4")
> > in new stack
> > -- Goto (CC,2001,7)
> > -- Executing [EMAIL PROTECTED]:7] AGI("SIP/protel-out-0934bb28",
> > "agi://localhost/updateCallStatus.agi?callStatus=answered") in new stack
> > -- AGI Script
> > agi://localhost/updateCallStatus.agi?callStatus=answered completed,
> > returning 0
> > -- Executing [EMAIL PROTECTED]:8] Set("SIP/protel-out-0934bb28",
> > "CALLERID(all)=") in new stack
> > -- Executing [EMAIL PROTECTED]:9] MixMonitor("SIP/protel-out-0934bb28",
> > "1192468625.7.wav|b") in new stack
> > -- Executing [EMAIL PROTECTED]:10] Dial("SIP/protel-out-0934bb28",
> > "SIP/2001|
> > 2
Re: [asterisk-users] get SIP extension status without calling it
On Sunday 02 December 2007 09:25:06 dave cantera wrote: > Vieri wrote: > > I am trying to get a SIP extension's status without > > actually making a call. > > > > I am using sofia-sip's "options" example utility and > > the sip clients are SJphone softphones. > > > > >From Asterisk I run the "options" utility and query a > > > > sip extension at 10.215.147.240. I get: > > > > # ./options -1 --all sip:10.215.147.240 > > SIP/2.0 501 Not Implemented > > # show current asterisk SIP peers > > asterisk -r -x 'sip show peers' | awk ' This method relies on the OPTIONS request being handled correctly, so it will not work. To the original poster: OPTIONS is the right type of request (the client should respond exactly as the same way as if you had sent it an INVITE). Your next step should be to contact the author of the softphone and impress upon them the necessity of having OPTIONS implemented in their client. Or use another softphone. -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cross-compiling asterisk-1.4 for Debian on a slug
On Saturday 01 December 2007 19:27:09 Philipp Kempgen wrote: > You did not make clear if you try to build on an i686 or on > a slug (as your subject says) which is not x86 but Intel > XScale. Oh, sorry! I want to run asterisk on the slug, but compile it on my desktop box, which is an i686. Wasn't clear, I see... > Anyhow: All I can say is that cross-compiling Asterisk is > probably not an easy task although some improvements have > been made recently. Well, compiling it for the same arch but big-endian works well. This is what I can't really understand! I'll probably post this again on the slug mailing list, when that seems more appropriate for this topic. Thank you for your answer anyway! Greetings, Fabiano ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Solaris
Hi, try adding this in your stdtime/localtime.c #define _POSIX_PTHREAD_SEMANTICS #undef TM_ZONE #undef TM_GMTOFF if this does not work just google it, there are workaround for this problem Thanks, Vivek On 12/2/07, Mike Clark <[EMAIL PROTECTED]> wrote: > > I submiited to the list last night, but it never showed up. Here we go > again. > > I've tried building Asterisk 1.4.15 on Solaris based on instuctions > here, http://forums.digium.com/viewtopic.php?t=5888. However, this is > the message I get. This is Solaris on X86. Any ideas? > > [CC] stdtime/localtime.c -> stdtime/localtime.o > stdtime/localtime.c: In function `localsub': > stdtime/localtime.c:1136: error: structure has no member named `tm_gmtoff' > gmake[1]: *** [stdtime/localtime.o] Error 1 > gmake: *** [main] Error 2 > > Thanks, > > Mike Clark > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Answering Machine Detection
carlos,
you got further than I did... AMD didn't work at all on my release.. I
think I was using 1.4.11 at the time...
I ended up using the below
daveC
;---< amdtest (ext 13) starts here >
;
; restructure this for the following conditions:
; 13 using waitforsilence(variable set) then play message
; only works when there is an answering machine picking up and it doesn't
; cut off (hangup) before SILENCEDURATION ms
exten => 13,1,NoOp( Starting exten 13 AMD stuff)
exten => 13,n,Wait(1)
exten => 13,n,Set(SILENCEDURATION=4300)
exten => 13,n,Set(SILENCEOCCURANCES="")
exten => 13,n,Set(SILENCETIMEOUT=38)
exten => 13,n,NoOp( SILENCEDURATION=${SILENCEDURATION} )
exten => 13,n,NoOp( SILENCEOCCURANCES=${SILENCEOCCURANCES} )
exten => 13,n,NoOp( SILENCETIMEOUT=${SILENCETIMEOUT} )
;exten => 13,n,Answer
exten =>
13,n,WaitForSilence(${SILENCEDURATION},${SILENCEOCCURANCES},${SILENCETIMEOUT})
exten => 13,n,NoOp(Retnd WAITSTATUS=${WAITSTATUS} )
exten => 13,n,Playback(lax/lax-important-msg-from)
exten => 13,n,PlayBack(lax/to-hear-msg-press-1)
exten => 13,n,Read(CALL_ACK,beep,1,,,3)
exten => 13,n,NoOp(CALL_ACK is >${CALL_ACK}<)
exten => 13,n,GotoIf([${CALL_ACK} = ""]?nak13)
exten => 13,n(ack13),NoOp( Ack )
exten => 13,n,NoOp( log the ACK acknowlegement here calling the AGI script)
;exten => 13,n,AGI(lax/track-laxcalls.sh,${EXTEN},${CALL_ACK})
exten => 13,n,GotoIf([${CALL_ACK} = ""]?play13)
exten => 13,n(nak13),NoOp( Nak )
exten => 13,n,NoOp( log the NAK acknowlegement here calling the AGI script)
;exten => 13,n,AGI(lax/track-laxcalls.sh,${EXTEN},${CALL_ACK})
exten => 13,n(play13),Playback(lax/lax-important-msg-from)
exten => 13,n,Playback(tt-weasels)
exten => 13,n,Playback(tt-monkeysintro)
exten => 13,n,Wait(1)
exten => 13,n,Hangup
Carlos Chavez wrote:
> I am having a bit of a problem getting AMD to work on a new server. On
> my regular office server it works like a charm. I am running Asterisk
> 1.4.13, Zaptel 1.4.5.1 on both machines. Both servers run CentOS 5 and
> I am using a SIP trunk to send out calls (the same one on both servers).
>
> Here is the output of a call on my office server:
>
> -- Attempting call on Local/[EMAIL PROTECTED] for [EMAIL PROTECTED]:1
> (Retry
> 1)
> -- Executing [EMAIL PROTECTED]:1]
> Set("Local/[EMAIL PROTECTED],2", "CIDTEMP="1" <5540881644>") in new
> stack
> -- Executing [EMAIL PROTECTED]:2]
> Dial("Local/[EMAIL PROTECTED],2", "SIP/protel-out/0445540881644|
> 25") in new stack
> -- Called protel-out/0445540881644
> -- SIP/protel-out-0934bb28 is making progress passing it to
> Local/[EMAIL PROTECTED],2
> -- SIP/protel-out-0934bb28 answered Local/[EMAIL PROTECTED],2
> -- Executing [EMAIL PROTECTED]:1] NoOp("Local/[EMAIL PROTECTED],1", ""1"
> <5540881644>") in new stack
> -- Executing [EMAIL PROTECTED]:2] AMD("Local/[EMAIL PROTECTED],1", "")
> in new stack
> -- AMD: Local/[EMAIL PROTECTED],1 5540881644 (null) (Fmt: 64)
> -- AMD: initialSilence [2500] greeting [1500] afterGreetingSilence
> [800] totalAnalysisTime [5000] minimumWordLength [100]
> betweenWordsSilence [50] maximumNumberOfWords [5] silenceThreshold
> [256]
> == Spawn extension (CC2, 0445540881644, 2) exited non-zero on
> 'Local/[EMAIL PROTECTED],2'
> -- Executing [EMAIL PROTECTED]:1] DeadAGI("Local/[EMAIL PROTECTED],2",
> "agi://localhost/updateCallStatus.agi?callStatus=hangupcc2") in new
> stack
> -- AGI Script
> agi://localhost/updateCallStatus.agi?callStatus=hangupcc2 completed,
> returning 0
> -- AMD: Word detected. iWordsCount:1
> -- AMD: Changed state to STATE_IN_SILENCE
> -- AMD: HUMAN: silenceDuration:800 afterGreetingSilence:800
> -- Executing [EMAIL PROTECTED]:3] GotoIf("SIP/protel-out-0934bb28",
> "1?7:4")
> in new stack
> -- Goto (CC,2001,7)
> -- Executing [EMAIL PROTECTED]:7] AGI("SIP/protel-out-0934bb28",
> "agi://localhost/updateCallStatus.agi?callStatus=answered") in new stack
> -- AGI Script
> agi://localhost/updateCallStatus.agi?callStatus=answered completed,
> returning 0
> -- Executing [EMAIL PROTECTED]:8] Set("SIP/protel-out-0934bb28",
> "CALLERID(all)=") in new stack
> -- Executing [EMAIL PROTECTED]:9] MixMonitor("SIP/protel-out-0934bb28",
> "1192468625.7.wav|b") in new stack
> -- Executing [EMAIL PROTECTED]:10] Dial("SIP/protel-out-0934bb28",
> "SIP/2001|
> 20") in new stack
> -- Called 2001
> == Begin MixMonitor Recording SIP/protel-out-0934bb28
>
>
> And here is the output on the new server:
>
> -- Attempting call on Local/[EMAIL PROTECTED] for [EMAIL PROTECTED]:1 (Retry
> 1)
> -- Executing [EMAIL PROTECTED]:1]
> Set("Local/[EMAIL PROTECTED],2", "CIDTEMP="1" <5540881644>") in new
> stack
> -- Executing [EMAIL PROTECTED]:2]
> Dial("Local/[EMAIL PROTECTED],2", "SIP/protel-out/0445540881644|
> 25") in new stack
> -- Called protel-out/0445540881644
> -- SIP/protel-out-09ce0358 is making progress passing it to
> Local/[EMAIL PROTECTED],2
>
Re: [asterisk-users] get SIP extension status without calling it
vieri,
you can get sip status with the following shell script... I named it
'sipshowpeer'... to execute, chmod 755 sipshowpeers
daveC
--< cut here >-
#!/bin/sh
# sipshowpeers
#
# show current asterisk SIP peers
asterisk -r -x 'sip show peers' | awk '
BEGIN{
#Name/username HostDyn Nat ACL Port Status
# $1$2 $3 $4 $5 $6 $7
}
{
name=$1
host=$2
dyn=$3
nat=$4
acl=$5
port=$6
status=$7
printf("%14.14s %18.18s %14.14s %14.14s %s\n",$1,$2,$3,$4,$5,$6,$7)
}
END{
printf("Done...\n")
}'
#502(Unspecified)D 0Unmonitored
#501(Unspecified)D 0Unmonitored
#40310.10.15.43 5060 Unmonitored
#40210.10.15.42 5060 Unmonitored
#401/401192.168.15.100 D 5062 Unmonitored
#301/301192.168.15.31D 5060 Unmonitored
#300/300192.168.15.31D 5060 Unmonitored
--< cut here
>---
Vieri wrote:
> Hi,
>
> I am trying to get a SIP extension's status without
> actually making a call.
>
> I am using sofia-sip's "options" example utility and
> the sip clients are SJphone softphones.
>
> >From Asterisk I run the "options" utility and query a
> sip extension at 10.215.147.240. I get:
>
> # ./options -1 --all sip:10.215.147.240
> SIP/2.0 501 Not Implemented
> Via: SIP/2.0/UDP
> 10.215.144.27:38098;branch=z9hG4bKUKS02S3F8H8ZS;received=10.215.144.27
> From: ;tag=U3DKgF7HgFKXH
> To: "unknown" ;tag=614733430
> Call-ID: b6968197-1b7d-122b-0ab0-00c09f10e472
> CSeq: 92182805 OPTIONS
> Content-Length: 0
> Server: SJphone/1.65.377a (SJ Labs)
>
> I guess that the softphone should be answering with a
> 2xx code followed by a status description?
> So I tried with the INVITE method and set DND on the
> SIP extension:
>
> # ./options -1 --all --method INVITE
> sip:10.215.147.240
> SIP/2.0 486 Busy Here
> Via: SIP/2.0/UDP
> 10.215.144.27:38098;branch=z9hG4bK5Z3BS8F737t0e;received=10.215.144.27
> From: ;tag=590Z1ND8B6XpN
> To: "unknown" ;tag=1a2d77b524
> Call-ID: 668ad4fa-1b7e-122b-fcb6-00c09f10e472
> CSeq: 92182952 INVITE
> Content-Length: 0
> Server: SJphone/1.65.377a (SJ Labs)
>
> The above would suit me fine because I get a "486 Busy
> Here" response.
> However, if DND is off then I get:
>
> # ./options -1 --all --method INVITE
> sip:10.215.147.240
> SIP/2.0 180 Ringing
>
> and the SIP extension actually "rings", as
> expected.(but this is undesireable)
>
> Now, does someone know another way to get the status
> (ie. does it accept calls or not?) without making the
> extension "ring"?
>
> Thanks
>
> Vieri
>
>
>
>
>
> Be a better pen pal.
> Text or chat with friends inside Yahoo! Mail. See how.
> http://overview.mail.yahoo.com/
>
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
--
My wife's sister is in California.
I should buy her a Videophone2008!
Truly, The Next Best Thing to Being There!
--
WorldWideVideoPhones.com
856.380.0894
___
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Re: [asterisk-users] Outgoing PSTN calls , unusable voice quality
Dear Salvatore and Joanna, Thank you much for both your detailed explanations. I will surely check my firewall configuration and logs to make sure the VoIP traffic is passing correctly. However, I'm a bit confused as the problems that I'm experiencing are with calls made via the Sangoma analog card. So the voice goes from the SIP phone through asterisk through the sangoma card then directly into the PSTN and vice versa. There are no firewalls in the way. Furthermore incoming calls are OK, the problem is only with outgoing calls when I can hear the other party well but they barely understand me. Is there any major difference in the way that Incoming/Outgoing calls are processed in the above scenario? Any way that I could trace those processes for faults? Thank you again. Regards, Veselin Salvatore Giudice wrote: When you take your packet capture, you'll need to look at the sip messages with SDP attached to get the ip's and ports used for both media streams. Make sure that the ips are correct and that the port used can traverse between those ip's without being blocked by a packet filter or firewall. A lot of times, administrators will set a range of UDP ports that are allowed to pass their packet filter for media and your pbx or phones may be using a different range. This can cause audio loss. You'll need to eliminate that possibility. Sometimes checking your firewall/packet filters for blocks may also prove helpful in identifying problems. You should be aware that the logs from certain firewall products may not be comprehensive. For example, in the past I have seen packets dropped going through netscreens because of invalid headers and no entries appeared in the logs. If you ultimately believe a firewall may be blocking your traffic make sure you setup a capture port or a span on each side of the device and verify the traffic going to and leaving from the firewall using ethereal on a laptop or maybe a Nixon box if you are in a large distributed environment. Never trust a potentially broken device to report accurate information about it's function. TDM = Time Division Multiplexing TDM describes how channels are separated on T1's, etc. It's common to refer to those types of connections as TDM. http://en.wikipedia.org/wiki/Time-division_multiplexing -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Veselin Kantsev Sent: Friday, November 30, 2007 8:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Outgoing PSTN calls , unusable voice quality Thank you much for the prompt reply Salvatore. Would you have the time to explain further how should I go for verifying that SDP and RTP are OK. Also what is reffered to as the TDM site. Veselin On Fri, Nov 30, 2007 at 05:01:17PM -0500, Salvatore Giudice wrote: Take a packet capture of your VoIP segment and verify that the SDP is correct and that the RTP is making it to the correct places. If all that looks good and this is a straight out quality problem, then you need to figure out if it's happening on the voip side or on the TDM side. You should make calls (with captures) VoIP to Voip passing the media through your asterisk and also try routing a tdm call in and back out. If you have the equipment, take a mos score of the TDM loop. Without any of the above, you will not be able to isolate the issue. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Veselin Kantsev Sent: Friday, November 30, 2007 2:47 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Outgoing PSTN calls , unusable voice quality Hello, I have an Asterisk running with a Sangoma A200 card with Hardware Echo cancelling connected to the UK PSTN. If a PSTN call comes in, voice both ways is OK, however if an outgoing call over the PSTN is made I can hear the other party OK but they can not, they can barely understand what I am saying, my voice is unclear fading and skipping. Internal SIP and IAX2 calls are OK, incoming/outgoing calls over IAX2 are OK too. I've tried gsm/ulaw/alaw codecs so far. Tried disabling the echo cancelling as well. Any suggestions will be greatly appreciated. Regards, Veselin ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _
[asterisk-users] Asterisk on Solaris
I submiited to the list last night, but it never showed up. Here we go again. I've tried building Asterisk 1.4.15 on Solaris based on instuctions here, http://forums.digium.com/viewtopic.php?t=5888. However, this is the message I get. This is Solaris on X86. Any ideas? [CC] stdtime/localtime.c -> stdtime/localtime.o stdtime/localtime.c: In function `localsub': stdtime/localtime.c:1136: error: structure has no member named `tm_gmtoff' gmake[1]: *** [stdtime/localtime.o] Error 1 gmake: *** [main] Error 2 Thanks, Mike Clark ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] get SIP extension status without calling it
Hi, I am trying to get a SIP extension's status without actually making a call. I am using sofia-sip's "options" example utility and the sip clients are SJphone softphones. >From Asterisk I run the "options" utility and query a sip extension at 10.215.147.240. I get: # ./options -1 --all sip:10.215.147.240 SIP/2.0 501 Not Implemented Via: SIP/2.0/UDP 10.215.144.27:38098;branch=z9hG4bKUKS02S3F8H8ZS;received=10.215.144.27 From: ;tag=U3DKgF7HgFKXH To: "unknown" ;tag=614733430 Call-ID: b6968197-1b7d-122b-0ab0-00c09f10e472 CSeq: 92182805 OPTIONS Content-Length: 0 Server: SJphone/1.65.377a (SJ Labs) I guess that the softphone should be answering with a 2xx code followed by a status description? So I tried with the INVITE method and set DND on the SIP extension: # ./options -1 --all --method INVITE sip:10.215.147.240 SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 10.215.144.27:38098;branch=z9hG4bK5Z3BS8F737t0e;received=10.215.144.27 From: ;tag=590Z1ND8B6XpN To: "unknown" ;tag=1a2d77b524 Call-ID: 668ad4fa-1b7e-122b-fcb6-00c09f10e472 CSeq: 92182952 INVITE Content-Length: 0 Server: SJphone/1.65.377a (SJ Labs) The above would suit me fine because I get a "486 Busy Here" response. However, if DND is off then I get: # ./options -1 --all --method INVITE sip:10.215.147.240 SIP/2.0 180 Ringing and the SIP extension actually "rings", as expected.(but this is undesireable) Now, does someone know another way to get the status (ie. does it accept calls or not?) without making the extension "ring"? Thanks Vieri Be a better pen pal. Text or chat with friends inside Yahoo! Mail. See how. http://overview.mail.yahoo.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Pcengines Alix board
> I guess /tmp can live in RAM, but what about eg. recording ten-twenty > WAV files to /var a day, and logs into /var/log? Do I have to worry > about the card wearing out in six months? This is nothing really. Just make sure your using an industrial compact flash card. These support 1-2 million cycles where many of the retail cards only support 100,000 cycles. We also greatly limit the logs being generated. Writing logs files creates many times more write cycles than voicemail ever could. If your concerned about logs use syslog to send them to an external system. > >I'm not sure I understand the need for the PCI card to be > perpendicular to > >the board. > > So I can use a flatter box. Then you don't want it perpendicular you want it parallel. The systems I'm using and even the e140 do this. There is a short riser card that plugs into the PCI slot on the board. The PCI card then plugs into this riser which allows the PCI card to be parallel to the motherboard. The cases I use are 12.5" by 10.5" by 2" so you can see that is a pretty thin box. John ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DID provider
Hello: Some time ago, in this list, I asked for a DID provider in Spain. Somebody answers me detailing his company DID services. I has an email problem and lost his response and his mail address. Please, if you (the DID provider) are reading this, let me know again your mail address because we are going to contract your DID service. Regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Answer Machine/Fax/modem detection
Maybe this can help: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AMD Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tong Sent: Sunday, December 02, 2007 7:36 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Answer Machine/Fax/modem detection Has anyone sucessfully implimented a fax or modem detection dial plan? I'm originating calls from asterisk using a list of numbers and dropping the destination into an IVR menu but need to do something different if a modem or fax answers. I tried to use the NVBackgroundDetect() application but i think that is for receiving faxes only. Any help would be appreciated. Thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
