[Asterisk-Users] Re: Asterisk-Users Digest, Vol 10, Issue 39
> > Message: 24 > Date: Thu, 5 May 2005 10:06:17 -0300 > From: "itamar" <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] softphone for ipaq h4350 > To: > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; format=flowed; charset="Windows-1252"; > reply-type=original > > there are softphones for ipaq h4350 > > I am using ms windows pocketpc 2003 > > Itamar Reis Peixoto > +55 (34) 3238 3845 > e-mail : [EMAIL PROTECTED] > http://vps.ispbrasil.com.br --->>> servidores linux > Yes there is, I have personally used SJphone from www.sjlabs.com with the same iPaq. I dont have that iPaq anymore but when I did it worked great. They have a free and a paid for version. Xten also have one, www.xten.com. Not free and I have no experience with it. Cheers Nick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Flash Timings
Hi, I am trying to integrate Asterisk with a very old PABX I have here for test purposes. I have it linked with and FXO module. Now the test scenario I am building goes like this: Incoming call on Legacy PABX --> Call Transferred to Asterisk --> Announcement Played --> Call Transferred to SIP Xtn --> If call is unanswered perform a hook flash on active zap channel and return it to the legacy pabx. Now whenever I perform that hook flash, the calling party is disconnected, which after much playing I have come to the conclusion that the Asterisk default flash time is too long. I'm based in Australia and as far as I know we use much shorter flash timings than in the US. I have tried changing the Default Flash Time in zaptel.h, but this does not make any difference, so I am just a little confused if I am making the change in the correct place? For what its worth the Legacy PABX is an Imagineering Ultra 8, just a cheap one I picked up a while back! Any inspiration would be appreciated! Cheers Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Flashing Active ZAP Channels
My problem is that I'm trying to do a flash on an active ZAP channel
to transfer a call, but every time the flash is performed the caller
that im trying to transfer gets disconnected. Here is a longer
explanation of whats going on.
I have a situation where I am linking asterisk upto a PABX via FXO
modules. Calls will come in via the traditional PABX and be transfered
to the asterisk system where a couple of VoIP phones are attached to
take the calls.
Now I need to set this up so that calls return to reception on the
pabx when they are not answered on Asterisk. The problem is we have to
do an announcement to the caller on Asterisk to say they are going to
be recorded. I wanted to have the announcement perfomed when the call
is answered by the destination extension, but I can only get it to
play the announcement to the destination extension not the caller.
So the problem is I have to answer the call when it first hits
Asterisk in order to play the announcement, so I cannot have the pabx
return the call after x amount of rings etc so the return will have to
happen on Asterisk.
I tried having the following in the dialplan for when the dial command
times out:
exten => s,1,Answer
exten => s,2,Dial(${ALL})
exten => s,3,Flash
exten => s,4,SendDTMF(22)
exten => s,5,Flash tried leaving out this stage but makes no difference.
exten => s,6,Hangup
Problem is every time the flash is performed the original caller gets
disconnected, and an empty call gets transferred... then Hangup
terminates all the calls.
This is just one of many ways I have been trying to achieve my goal,
but every one has had some kind of block I could not get round!
Any help would be appreciated as I'm all out of ideas and have to get
this working in the next day or two :(
TIA!
Nick
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[Asterisk-Users] Playing announcement when call is answered
Okay, I posted a question earlier asking how to start the monitor
command when the call is answered not when it is dialed. I managed to
fix the problem myself by using a macro in the Dial string like this:
exten => 21060,1,Dial(${Nick},20,M(monitor))
Next problem though! What I need to do is play a legal announcement to
the caller when the the call is answered. Now I thought this would be
easy, but it seems I am missing something. I simply added the playback
line to the same macro that I used for my monitor command.
I made sure the call was answered before commencing playback along
with having a pause and removed the r option from the dial string so
as to not block playback on the line. But when I call in the audio is
played on the destination extension but not to the caller. Caller just
hears the ring until the macro is complete then conversation
commences. My macro I am using is below.
[macro-monitor]
exten => s,1,Answer
exten => s,2,Wait(2)
exten => s,3,Playback(legal-announcement)
exten => s,4,Monitor(wav49,${UNIQUEID},m)
I have tried lots of variations of this, including using Background
not Playback.
Anyone have any ideas?
TIA for any help anyone can offer.
Oh yea, I also tried answering the call in the dial string rather than
the macro but this had the same effect. I also want to avoid answering
the call before the call has been picked up, as this will be an
extension of an existing PABX, so we want the call to return to
reception if the call goes unanswered. Not sure how I would go abouts
doing this if astrisk answers the call?
The test system I am doing this on has calls incoming via IAX and
terminating on SIP extensions. But the final production system will be
using Zaptel hardware for the incoming call and SIP extensions.
Cheers
Nick
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[Asterisk-Users] Monitor Command
Hi,
I have asked this question before, but only just got round to testing
the solution, and it does not work as expected.
What I need to do is record all calls from the time the call is
answered not from the moment the call starts to ring.
The solution suggested previously was to answer the call in my
dialplan first then dial the phone. I just tried this but it still
records the ringing (while previously I would just get silence rather
than ringing).
Is there any way to do this?
Currenlty all I have in my dialplan for this is
exten => 21060,1,Monitor(wav49,${UNIQUEID},m)
exten => 21060,2,Dial(${Nick},20,t)
Also, what would be even better, though not essential, is if I could
put the caller on hold then when the call is answered the legal
announcement will be played played before the call commences. I looked
at using the A variable within the dial command to do this, but if I
understand it correctly this will play a message to destination rather
than the caller.
Hope all this makes sense, and TIA for any advice.
Regards
Nick
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Re: [Asterisk-Users] Connecting to Exicom GSX 418/816
Excellent news :) We will be hooking up our system to a number of PABX's over the next few months, of which I expect a lot of them to be pre stone age. Is hooking it up this way preferable to using an FXO module? I can see there is a cost advantage as a lot of these systems are maxed out or not able to provide the 2 wire extensions, but in a lot of cases they seem to have CO lines spare. But just wondering if there is actually any technical advantage to hooking it up this way? On Wed, 10 Nov 2004 21:58:41 -0600, Eric Wieling <[EMAIL PROTECTED]> wrote: > Nick Cobley wrote: > > > > I have a need to connect up asterisk to an Exicom GSX 418/816, this > > will be a very simple setup, just one extension on the Asterisk box so > > only one line to the PABX required. > > > > Problem lies in the Exicom being a Key system and and we cannot source > > any Single Line Modules for this system to allow me to interface this > > with and FXO module. So I was thinking, would it be technically > > possible to use one of the CO and hook that upto an FXS module on > > Asterisk? > > > > I'm kinda desperate to make this work! > > This is the best way to do it. Use a Digium FXS module and plug it into > the CO port on your PBX. > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Connecting to Exicom GSX 418/816
I have a need to connect up asterisk to an Exicom GSX 418/816, this will be a very simple setup, just one extension on the Asterisk box so only one line to the PABX required. Problem lies in the Exicom being a Key system and and we cannot source any Single Line Modules for this system to allow me to interface this with and FXO module. So I was thinking, would it be technically possible to use one of the CO and hook that upto an FXS module on Asterisk? I'm kinda desperate to make this work! TIA Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Connecting to Commander NT132
On Fri, 22 Oct 2004 16:52:27 +1000, Christopher Lee <[EMAIL PROTECTED]> wrote: > Hi Nick, > > The Commander NT132 is essentially a rebadged Nortel Norstar MICS - I > believe with a slightly different firmware for Australian conditions. I > have the smaller version - Commander NT40 (which is a rebadged Norstar > CICS) connected up to Asterisk via the Norstar's built in Analog port. > > You can get analog ports for the Commander NT132 if you don't have > any... I don't know how much they cost, but you can often find second > hand parts for these systems relatively easily. > > Actually I wish I had a Commander NT132 for my setup as I ended up > tagging 3 analog adapters off the NT40 system (which take up digital > extension ports) and still need more analog ports. The external ATA's > are so bulky and awkard, at least with the NT132 you can buy a proper > analog card - I belive they have about 8 ports or so to plug directly > into the system, great if you have a need for quite a few analog > extensions. > > There was some good doco written up by David Gomillion on intergrating > the Norstar MICS with Asterisk, but it's based on using PRI (Primary > Rate Interface) cards. > > http://www.voip-info.org/wiki-Asterisk+Interop+Nortel+Norstar+MICS > > Integration with analog FXS port (on the NT40 anyway) is relatively > straightforward - but a big downside of these smaller Nortel systems is > they don't offer up any disconnect supervision on the FXS ports (and in > my case I also have a 4port FXO trunk card in the system which also > lacks disconnect supervision). > > This means when someone calls from the NT40 into my asterisk system > (through a X101P card), if they hangup first the X101P doesn't get any > signal that the line has hungup. It was pretty annoying to start with, > but through ensuring my dialplans all correctly hangup the line from my > end when they're supposed to, and watching for silence in voicemails, > then much of the trouble is solved and I find it works quite well for > what I need. > > Another problem with integrating via analog is that you don't receive > any CallerID info from the NT40... So if someone calls from the NT40 and > I miss the call, I have to do a bit of guesswork to figure out which > extension to try calling back to find the person who was looking for me. > > As for integrating via ISDN with a Fritz card, I'm not sure, someone > else might be able to answer that better... I asked about this quite > some time ago about interfacing the NT40 via BRI - I believe you need a > card with a HFC chipset like this one - > > http://www.junghanns.net/asterisk/page17.html > > My previous question - "Norstar Integration with Asterisk via FXO or BRI > ISDN" > http://lists.digium.com/pipermail/asterisk-users/2004-February/035729.ht > ml > > It's not cheap, and hence why I haven't really pursued integration of my > system via ISDN... from what I've read it seems PRI really is the best > way to integrate, which is a downside if you only have BRI interfaces & > lines. > > Hope this helps some... The most important thing to know is that what > you have is a Nortel Norstar MICS system, which makes searching for > information much easier as they are very common systems in the states. > > Regards, > Chris Lee > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of > > Nick Cobley > > Sent: Friday, 22 October 2004 3:52 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: [Asterisk-Users] Connecting to Commander NT132 > > > > Hi, > > > > I am looking at connecting Asterisk upto a Commander NT132. I > > need 2 lines, and initially was going to connect it up to > > some analog ports, which I have since discovered they don't have. > > > > So I am taking another look at other options rather than > > getting a couple single line cards. > > > > Now firstly, would I be able to install something like a > > Fritz ISDN card and hook that upto the PABX? I know this can > > be done with say an > > E1 card but was not sure if the same applies with an ISDN > > card connecting to the PABX rather than the carrier. > > > > If someone could perhaps summarise my options I would appreciate it. > > We will be installing a number of systems in exactly the same > > configuration, so I need to try and understand these areas a > > little more. > > > > BTW I am in Australia if that makes a difference. > > > > Kind regards > > Nick Thanks guys, Well t
[Asterisk-Users] Connecting to Commander NT132
Hi, I am looking at connecting Asterisk upto a Commander NT132. I need 2 lines, and initially was going to connect it up to some analog ports, which I have since discovered they don't have. So I am taking another look at other options rather than getting a couple single line cards. Now firstly, would I be able to install something like a Fritz ISDN card and hook that upto the PABX? I know this can be done with say an E1 card but was not sure if the same applies with an ISDN card connecting to the PABX rather than the carrier. If someone could perhaps summarise my options I would appreciate it. We will be installing a number of systems in exactly the same configuration, so I need to try and understand these areas a little more. BTW I am in Australia if that makes a difference. Kind regards Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Silence on incoming monitored calls
I am currently recording incoming calls using the monitor application. Problem is it is recording silence for the ringing portion of the SIP call. We really would like it to commence from the beginning of the conversation. Is there any way to do this? Also, I tried setting monitor to record the files in GSM format but Windows Media Player will not play it back, does not recognise it as a valid file. I have played GSM files back with out any problems in the past with our old logging system. Any ideas? Cheers Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cdr record for recording location
Hopefully someone can help out here. I currently have cdr records being logged to mysql for each call, along with all calls being recorded with the monitor application. What I really need is for the path to the recorded file to be logged with the corresponding cdr record. Is this possibile? Had a good hunt on tikki but can't seem to find anything. Also, I am finding with incoming calls, I get silence at the begining of the call when the destination phone is ringing, is there a way to not record this? Or will I need to post process it out in some way. Cheers Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P Internal Extenion Config
Jason your a legend!!! I swear I tried include => internal in the sip
context, guess I managed to stuff it up somehow!!
Thanks so much for your help, sanity now saved :)
Regards
Nick
Jason Williams wrote:
On Mon, 19 Jul 2004 20:13:09 +0800, Nick Cobley <[EMAIL PROTECTED]> wrote:
Hopefully someone here can save my sanity. I have been trying to solve
this problem for days now, but just cant put my finger on it. Im new to
* so I have probably done something stupid!
Only a config issue I'm sure
[sip]
exten => 301,1,Dial(SIP/Nick,20,tr)
exten => 302,1,Dial(SIP/Sharon,20,tr)
exten => 1000,1,Dial(SIP/Nick&SIP/Sharon,20,tr)
exten => 302,2,VoiceMail,u302
exten => 301,2,VoiceMail,u301
exten => 1000,2,VoiceMail,u
exten => 1000,102,VoiceMail,b
exten => 1001,1,Ringing
exten => 1001,2,Wait(2)
exten => 1001,3,VoicemailMain
include => outgoing
add here
include => internal ; allow sip to dial 310
[incoming]
exten => s,1,Dial(SIP/Nick&SIP/Sharon,20,tr)
[outgoing]
exten => _7.,1,Dial(IAX2/login:[EMAIL PROTECTED]>XXX/${EXTEN:1})
exten => 5.,1,Dial,Zap/1/${EXTEN:1}
[9103]
exten => 21060,1,Dial(SIP/Nick)
exten => 21062,1,Dial(SIP/Sharon)
[internal]
exten => 310,1,Dial,Zap/2
include => sip ; allow internal to dial sip phone
Try those changes and see how you get on
Jason
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Re: [Asterisk-Users] TDM400P Internal Extenion Config
Thanks Steve, The SIP handsets are working find as I can make calls to other handsets as well as receive incoming calls via the FXO module. So all is good there. Cheers Nick Steven Critchfield wrote: On Mon, 2004-07-19 at 07:13, Nick Cobley wrote: If I dial the extension I just get a 404 error on the phone (Grandstream), but no errors at all on the console. I am using CVS-HEAD-07/14/04. Here is a snippet of what I have in the various config files. Welcome to SIP. Dialtone is local to your phone and is not dependent on proper config. Hope that helps put you on the correct step to fix that problem. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: TDM400P Internal Extenion Config
Forgot to mention, both modules are show in ztcfg fine, see below:
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXO Kewlstart (Default) (Slaves: 02)
2 channels configured.
and zap show channel 2 give the following:
Channel: 2
File Descriptor: 19
Span: 1
Extension:
Dialing: no
Context: internal
Caller ID string: "Fax" <310>
Destroy: 0
Signalling Type: FXO Kewlstart
Owner:
Real:
Callwait:
Threeway:
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: no
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: ulaw
Fax Handled: no
Pulse phone: no
Echo Cancellation: 128 taps, currently OFF
Actual Confinfo: Num/0, Mode/0x
Actual Confmute: No
Actual Hookstate: Onhook
Okay, so now I'm going to lie down in a dark room.
Cheers
Nick Cobley wrote:
Hopefully someone here can save my sanity. I have been trying to solve
this problem for days now, but just cant put my finger on it. Im new
to * so I have probably done something stupid!
I have a TDM400P with one FXO module and a FXS module. The main
problem I have is not being able to get the extension attached to the
FXS module to ring or be able to make calls. It gets a dialtone fine
but I guess this doesnt really mean all that much.
If I dial the extension I just get a 404 error on the phone
(Grandstream), but no errors at all on the console. I am using
CVS-HEAD-07/14/04. Here is a snippet of what I have in the various
config files.
zaptel.conf
loadzone=au
defaultzone=au
fxsks=1
fxoks=2
zapata.conf
[channels]
context=incoming
switchtype=national
signalling=fxs_ks
usecallerid=yes
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
txgain=0.0
rxgain=0.0
group=1
callgroup=1
pickupgroup=1-4
immediate=no
busydetect=yes
busycount=7
callerid=asreceived
channel => 1
context=internal
group=2
signalling=fxo_ks
callerid="Fax" <310>
channel => 2
extensions.conf
[general]
static=yes
writeprotect=no
[globals]
[sip]
exten => 301,1,Dial(SIP/Nick,20,tr)
exten => 302,1,Dial(SIP/Sharon,20,tr)
exten => 1000,1,Dial(SIP/Nick&SIP/Sharon,20,tr)
exten => 302,2,VoiceMail,u302
exten => 301,2,VoiceMail,u301
exten => 1000,2,VoiceMail,u
exten => 1000,102,VoiceMail,b
exten => 1001,1,Ringing
exten => 1001,2,Wait(2)
exten => 1001,3,VoicemailMain
include => outgoing
[incoming]
exten => s,1,Dial(SIP/Nick&SIP/Sharon,20,tr)
[outgoing]
exten => _7.,1,Dial(IAX2/login:[EMAIL PROTECTED]>XXX/${EXTEN:1})
exten => 5.,1,Dial,Zap/1/${EXTEN:1}
[9103]
exten => 21060,1,Dial(SIP/Nick)
exten => 21062,1,Dial(SIP/Sharon)
[internal]
exten => 310,1,Dial,Zap/2
If I try to make any calls from the extension connected to the fxs
module i just get what sounds like a busy tone. Looking at the console
it generally give the error "zt_set_hook: zt hook failed Device or
resource busy". It only gives this error when it goes off hook and
number dialed.
Only other information I can provide is a couple errors when asterisk
start up. I get the following:
"Unable to open /dev/dsp: No such device"
"Unable to get our IP address, Skinny disable"
"Ignoring switchtype"
Have not been able to dig out vast amounts of information on the
above, but what I have found didnt seem to point to my problem, but
then what do I know!
If anyone can help I would appreciate it! I'm going crazy here!
Kind regards
Nick
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[Asterisk-Users] TDM400P Internal Extenion Config
Hopefully someone here can save my sanity. I have been trying to solve
this problem for days now, but just cant put my finger on it. Im new to
* so I have probably done something stupid!
I have a TDM400P with one FXO module and a FXS module. The main problem
I have is not being able to get the extension attached to the FXS module
to ring or be able to make calls. It gets a dialtone fine but I guess
this doesnt really mean all that much.
If I dial the extension I just get a 404 error on the phone
(Grandstream), but no errors at all on the console. I am using
CVS-HEAD-07/14/04. Here is a snippet of what I have in the various
config files.
zaptel.conf
loadzone=au
defaultzone=au
fxsks=1
fxoks=2
zapata.conf
[channels]
context=incoming
switchtype=national
signalling=fxs_ks
usecallerid=yes
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
txgain=0.0
rxgain=0.0
group=1
callgroup=1
pickupgroup=1-4
immediate=no
busydetect=yes
busycount=7
callerid=asreceived
channel => 1
context=internal
group=2
signalling=fxo_ks
callerid="Fax" <310>
channel => 2
extensions.conf
[general]
static=yes
writeprotect=no
[globals]
[sip]
exten => 301,1,Dial(SIP/Nick,20,tr)
exten => 302,1,Dial(SIP/Sharon,20,tr)
exten => 1000,1,Dial(SIP/Nick&SIP/Sharon,20,tr)
exten => 302,2,VoiceMail,u302
exten => 301,2,VoiceMail,u301
exten => 1000,2,VoiceMail,u
exten => 1000,102,VoiceMail,b
exten => 1001,1,Ringing
exten => 1001,2,Wait(2)
exten => 1001,3,VoicemailMain
include => outgoing
[incoming]
exten => s,1,Dial(SIP/Nick&SIP/Sharon,20,tr)
[outgoing]
exten => _7.,1,Dial(IAX2/login:[EMAIL PROTECTED]>XXX/${EXTEN:1})
exten => 5.,1,Dial,Zap/1/${EXTEN:1}
[9103]
exten => 21060,1,Dial(SIP/Nick)
exten => 21062,1,Dial(SIP/Sharon)
[internal]
exten => 310,1,Dial,Zap/2
If I try to make any calls from the extension connected to the fxs
module i just get what sounds like a busy tone. Looking at the console
it generally give the error "zt_set_hook: zt hook failed Device or
resource busy". It only gives this error when it goes off hook and
number dialed.
Only other information I can provide is a couple errors when asterisk
start up. I get the following:
"Unable to open /dev/dsp: No such device"
"Unable to get our IP address, Skinny disable"
"Ignoring switchtype"
Have not been able to dig out vast amounts of information on the above,
but what I have found didnt seem to point to my problem, but then what
do I know!
If anyone can help I would appreciate it! I'm going crazy here!
Kind regards
Nick
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