Re: [asterisk-users] Callback / Camp / Extention Free notify?
Funnily enough, most people install phones with BLF lamps, on install something
like hudlite/FOP/etc so you know if the person is on the phone before you call
them..
PaulH
Daniel Johnson da...@scanningsystems.com.au wrote:
Hi,
I am trying to implement the callback feature of our old phone system.
This feature may go by a different name in asterisk?
It worked as follows. If phone A called phone B and it was BUSY, you
press a button to enable a callback.
User A is free to continue work or make other calls.
What this meant is that when both phones became free, phone A would
ring, on answer it would call phone B automatically.
Here is how I have tried to achieve this with asterisk.
When A calls B and B is busy you get a menu, 1 to set callback and 2 to
leave voicemail.
The set of the callback is done via AGI call to PHP script to set the
details in MySQL DB.
On all call hangups I check to see if there is a pending Callback via
another AGI script.
The script sets a couple of variables which I check in my Dialplan. If
there is a callback pending for the
phone that just hung up, I need to check that the other phone involved
is FREE. This is what I can not get to work.
I have tried ChanIsAvail which does not appear to work.
I have hints setup for each SIP phone. Which would be perfect, however
it does not appear that you can check the HINT STATE in the dialplan.
I have done plenty of googling and have found this
http://bugs.digium.com/view.php?id=10635 which appears that this kind of
functionality was placed in 1.4.11. (DEVICE_STATE(), EXTENSION_STATE())
I have 1.4.21.2 and do not have these features.
To continue with the rest of the feature. If the ChanIsAvail says all is
good. I then launch another AGI to write a CALLFILE and remove the
pending callback request from the DB.
This all works if A is not busy when B finishes their call etc.
I am sure that others have implemented this kind of feature. If you
could share your implementation or give me some pointers or even the
correct asterisk name so I can google and get the help I need, that
would be great.
I am considering trying out 1.6 which should have these features,
however not sure if stability is going to be a problem.
I have based my implantation based of a previous message to the list:
On Tue, Jun 10, 2008 at 5:34 PM, Phil Knighton phil.knigh...@mjog.com
wrote:
Hello
I'm looking for a way to do the following using my Asterisk system
and Snom
SIP phones...
Scenario:
Caller on Internal Phone 1 calls internal phone2. Phone 2 is busy
(or more
accurately goes straight to voicemail).
Caller on internal phone 1 can press a button / dial a code
(explained in
next step) and hangup
When phone 2 is free, phone 1 rings and on answer dials phone 2
I was sure this was called camping - but all the camping stuff I
can find,
refers to the caller having to hang on the phone and wait. Am I
missing
something?
Anyone have a solution?
Quick solution that comes into mind:
Set(exten_copy = ${EXTEN});
Dial(SIP/${EXTEN})
if (${DIALSTATUS}=BUSY) {
// prompt for camp
Set(DB(camp/${EXTEN}/call_to)=${CALLERID(num));
}
h = {
Set(call_to=${DB(camp/${exten_copy}/call_to)});
if (${call_to}!=) {
Set(DB(camp/${exten_copy}/call_to)=);
System(call_to ${exten_copy} ${call_to});
}
}
So, in case if phone2 is busy, store callerid of phone1 in database,
so when phone2 will hangup it will triger a script call_to which
however can originate call trough manager or call-file.
Of course you will need some additional handling in case if multiple
callers decide to camp, or diferent protocols are used, etc.
Regards,
Atis
Thanks in Advance for any help.
Regards
*Daniel Johnson*
Systems Administrator / Systems Development
Scanning Systems Australia
Scanning Systems Australia
*Office:* +61 7 3387
*Facsimile:* +61 7 3387 5588
*E-mail:* da...@scanningsystems.com.au
mailto:da...@scanningsystems.com.au
*Website:* http://www.scanningsystems.com.au
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Re: [asterisk-users] Callback / Camp / Extention Free notify?
Daniel Johnson da...@scanningsystems.com.au wrote: pdha...@optusnet.com.au wrote: Funnily enough, most people install phones with BLF lamps, on install something like hudlite/FOP/etc so you know if the person is on the phone before you call them.. PaulH Hi Paul, Yes I have seen these tools. However it is a manual process (simple, I know) and is not close to being as user friendly as the feature we are trying to achieve. Understood completely - I was simply saying that you might not find as much information as you were looking for because people use other tools. PaulH ___ -- 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] Callback / Camp / Extention Free notify?
Daniel Johnson da...@scanningsystems.com.au wrote: pdha...@optusnet.com.au wrote: Funnily enough, most people install phones with BLF lamps, on install something like hudlite/FOP/etc so you know if the person is on the phone before you call them.. PaulH Hi Paul, Yes I have seen these tools. However it is a manual process (simple, I know) and is not close to being as user friendly as the feature we are trying to achieve. Do the phones you are using support BLF? PaulH ___ -- 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] Callback / Camp / Extention Free notify?
Daniel Johnson da...@scanningsystems.com.au wrote: Jeff LaCoursiere wrote: I think you are looking to use a campon feature. Try this: http://www.voip-info.org/wiki/view/Asterisk+tips+campon j Hi Jeff, Yes I have seen this feature. Its a half implementation of what we require. The difference being that you must wait on the phone until the dialed party becomes available (be it on hold or continuous dial). From my original email, the description of out old systems callback feature: It worked as follows. If phone A called phone B and it was BUSY, you press a button to enable a callback. User A is free to continue work or make other calls. What this meant is that when both phones became free, phone A would ring, on answer it would call phone B automatically and the call connected as per normal. To be honest, I actually remember using a system like this - at the NAB 20 years ago. And it behaved exactly as you describe it...memories PaulH ___ -- 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] soft ATA on linux with zaptel?
If machine 1 has a zaptel card in it...and not running asterisk...and another machine with no card, but running Asterisk... Me confused! Move the card? Move the Asterisk? PaulH Brian J. Murrell br...@interlinx.bc.ca wrote: Slightly OT, but I'm wondering if anyone here has come across a soft ATA. That is, software that will perform the functions of a basic POTS line ATA on Linux with a zaptel driven card. I have a Linux machine with a zaptel card in it and I want to have another Linux machine running Asterisk utilize the zaptel card in the first Linux machine to make outgoing and receive incoming calls. I realize I could make Asterisk do this job, but it seems pretty heavy- weight for just that purpose -- of bridging a POTS line to a SIP (or IAX) connection. Ideas? b. ___ -- 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] Basic Possiblity Question.
Sure - read up on IAX for a few good points. PaulH rupak shrestha [EMAIL PROTECTED] wrote: Hi all, i have a basic question on asterisk.The below is my scenerao. I have my sales offices around the globe.Theyare all connected with Speed Internet connection.I don't mind installing 1 asterisk box in each site.i don't mind using IP phone.i just wanted to call them for free at the cost of existing internet connectionwe have at each site.All the asterisk box will be connected with TCP/IP with one of it's NIC card having a WAN connectivity.is it possible with asterisk.Please let me know.Thanx _ Going green? See the top 12 foods to eat organic. http://green.msn.com/galleries/photos/photos.aspx?gid=164ocid=T003MSN51N 1653A ___ -- 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] queuemetrics
We are looking for a site running Queumetrics in Sydney, Australia. We have been contacted by a company in Sydney, as a few staff members of a company that are currently running Queuemetrics would like to see a fully running installation for training and decision making purposes. Their trial licence has run out and they did not test the system to the level they would have liked. Please respond to me in person if you can help. We are happy to pay for someones time on this matter. Kind regards, PaulH ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quad BRI card
I thought the price of E1 in Australia was quite reasonable, at least compared to analog. Paul Hales Technical Manager AsteriskIT www.asteriskit.com.au - Original Message - From: Craig Guy [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, May 20, 2006 1:51 AM Subject: Re: [Asterisk-Users] Quad BRI card By the last sentence I mean that only the person or company holding the A-tick can put the sticker on the cards. Paralell importation refers to 'grey' imports that don't come through the vendors sanctioned distribution channels. For example I know that the fritz! has passed approval because this guy has gone through the approval process. The Australian distributor sells them for $400, I can get them off eBay in Europe for $20 per card - the exact same card. $400 is just pure extortion and is going a hell of a long way to prevent the adoption of Asterisk in this country where BRI is the norm and PRI is outrageously expensive. If I had a spare $20k or so then I'd approve the card myself and sell them at a more realistic price. Craig - Original Message - From: Andrew Furey [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, May 19, 2006 8:54 AM Subject: Re: [Asterisk-Users] Quad BRI card On 5/18/06, Craig Guy [EMAIL PROTECTED] wrote: Any device to legally connect to the PSTN in Australia must be approved by the regulatory body. A process that usually costs at least $20,000 and only allows the permit holder to sell the product for conneciton to the pstn. It is a very high barrier to entry for the Australian market. There is a guy in Victoria who certified the Fritz! card and charges $400 each for them. Paralell imports are not allowed to be connected. Ah, so that's why they're so expensive :( Sorry, what do you mean by that last sentence? Andrew -- Linux supports the notion of a command line or a shell for the same reason that only children read books with only pictures in them. Language, be it English or something else, is the only tool flexible enough to accomplish a sufficiently broad range of tasks. -- Bill Garrett ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Any IP phones with pro-audio connections?
The new grandstream video phone has rca-style audio jacks Paul Hales Technical Manager AsteriskIT www.asteriskit.com.au - Original Message - From: Julien Goodwin [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, May 20, 2006 9:30 PM Subject: [Asterisk-Users] Any IP phones with pro-audio connections? Does anybody know of any IP phones (ideally SIP based) that have interfaces to plug into a pro audio system (eg for phone interviews). Something can probably be hacked up with a headset connector or the 1/8 jacks on a 7970 but I'm wondering if there's something better out there. Thanks, Julien ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Hangs the whole system
In our case, it was cpuspeed (a daemon) interfering with the zaptel drivers. Paul Hales Technical Manager AsteriskIT - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, April 27, 2006 10:14 PM Subject: Re: [Asterisk-Users] Asterisk Hangs the whole system A.R. Nasir Qureshi wrote: Is it possible for asterisk to hang the whole system ?? My Linux box is acting up, and I want to be sure which way to look. Asterisk or some hardware. Both are possible. If you watched the cvs/svn commits over the last year or so, several asterisk issues have been identified and corrected relating to mem allocation, dereferencing, etc, etc. I don't know that anyone has actually kept track of bugs vs versions to know which versions might be suspect, but it might help if you'd include which distro/kernel you're running, asterisk version, types of cards installed, etc. You might also try running memtest just to rule out memory failures or issues. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Variables
Each call has a unique callid - I used that for a dialplan a short while ago, to do a very similar job to what you are doing... Paul Hales Technical Manager AsteriskIT - Original Message - From: Shaun [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, April 16, 2006 5:06 PM Subject: [Asterisk-Users] Variables I have a call screening system setup, caller calls in runs a macro and sets a far to track the recording that was taken of the callers name... then the callee runs a macro also that plays him that recording (pulled from that var that was set) This works fine until i use a queue in the middle of it all... it appears that with queues that the file name stored in a var called SCREEN_FILE is lost once the caller is taken out of the queue.. Is their a uniq ID or somthing thats set to each call that i can use as the file name so i can always play back that file that was recorded or is their a way to to not loose the value of SCREEN_FILE once the caller is put into the queue? I though about setting SCREEN_FILE as global but i think that will cause problems with multiple calls and SCREEN_FILE being overwritten by other callers and the screening macro running... If each call had a uniq session id i could easily just use that -- ~Shaun ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] FreePBX in Production systems?
We had an issue at an install of [EMAIL PROTECTED] - where if you use the external extensions the machine is unable to start Asterisk after a reboot. Which in the end begged a question - it was nice have customers who could edit their box, but was it worth it for the angry calls when their PABX would not start up? PaulH Rob Terhaar [EMAIL PROTECTED] wrote: I'm currently using it at 2 offices- each one is about 40 phones On 4/15/06, Min Hwan Chang [EMAIL PROTECTED] wrote: Is anyone using FreePBX in production level systems because I'm just wondering if its stable enough to use. Currently I'm editing my own *.conf scripts but it sure would be nice if there were some sort of web interface for other people to use. The only thing holding me back is the stability of the FreePBX package... Any comments on this? Thanks in advance. Regards, Min Chang ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk credit card processing
We have written something, so the answer is yes. regards, Paul Hales Technical Manager AsteriskIT Joseph [EMAIL PROTECTED] wrote: Is there a way somehow to implement Asterisk with Credit Card Processing (IVR system)? -- #Joseph ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Force codec
What about different extensions using different connections? Paul Hales Technical Manager AsteriskIT - Original Message - From: Brian Capouch [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, April 10, 2006 4:26 AM Subject: Re: [Asterisk-Users] Force codec Kerry Garrison wrote: Disallow=all allow=ulaw N.B. the problem is depending on extension, not context or protocol. . . B. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Michael Strelnikov *Sent:* Saturday, April 08, 2006 7:25 PM *To:* asterisk-users@lists.digium.com *Subject:* [Asterisk-Users] Force codec Hi, Is it possible to force using codec depends on extension? For example, voice codec is ILBC and with some prefix fax code should be ulaw. Thanks. Best regards, Michael http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Force codec
I meant dialled extension, not originating
extension.
like :
exten =
_37X,1,Dial(IAX2/FAX/${EXTEN})
exten =
_38X,1,Dial(IAX2/NOTFAX/${EXTEN})
Paul HalesTechnical
ManagerAsteriskIT
- Original Message -
From:
Michael
Strelnikov
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Monday, April 10, 2006 11:13
AM
Subject: Re: [Asterisk-Users] Force
codec
I want to make it global.
On 4/10/06, [EMAIL PROTECTED] [EMAIL PROTECTED]
wrote:
What
about different extensions using different connections? Paul
HalesTechnical ManagerAsteriskIT- Original Message
-From: "Brian Capouch" [EMAIL PROTECTED]To: "Asterisk
Users Mailing List - Non-Commercial Discussion" asterisk-users@lists.digium.comSent:
Monday, April 10, 2006 4:26 AMSubject: Re: [Asterisk-Users] Force
codec Kerry Garrison wrote:
Disallow=all allow=ulaw N.B. the
problem is "depending on extension," not context or protocol. .
. B.
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
] *On Behalf Of *Michael
Strelnikov *Sent:* Saturday, April 08,
2006 7:25 PM *To:* asterisk-users@lists.digium.com
*Subject:* [Asterisk-Users] Force
codec Hi,
Is it
possible to force using codec depends on extension? For
example, voice codec is ILBC and with some
prefix fax code should be
ulaw. Thanks.
Best regards,
Michael http://lists.digium.com/mailman/listinfo/asterisk-users
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-- Best
regards,Michael Strelnikov
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Re: [Asterisk-Users] Config TE110P and TDM400 with 2 FXS modules
WaitExten perhaps?
Paul Hales
Technical Manager
AsteriskIT
Dennis [EMAIL PROTECTED] wrote:
Hi,
Let me explain a little about our system here first.
We have a Digium TE110P card hooked up to an isdn 30 line. (Australia -
EuroISDN) This works fine.
We also have a Digium TDM400P with 2 FXS modules installed. (The green
modules)
The /etc/zaptel.conf file has nothing but the following in it.
-=-=-=-
span=1,1,1,ccs,hdb3,crc4
dchan=16
bchan=1-15,17-31
defaultzone=au
loadzone=au
##tdm card
fxoks=33-34
-=-=-=-
The /etc/asterisk/zapata.conf has been stripped down to include only the
following.
-=-=-=-
[trunkgroups]
[channels]
language=en
rxwink=300
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=yes
faxdetect=both
jitterbuffers=24
signalling=pri_cpe
switchtype=euroisdn
echocancel=yes
echocancelwhenbridged=yes
echotraining=400
context=default
channel = 1-15,17-31 ; Set this to 1-15,17-31 for E1
signalling=fxs_ks
context=uniware_sendfax
channel =33-34
-=-=-=-
What we want to be able to do is plug a standard telephone into an fxs port
on the tdm card, and have it get a line from the E1 when the handset is
picked up.
The problem is that when the handset is picked up, the phone automatically
gets picked up by asterisk as an incoming call, and the call is managed by
the incoming call part of the extensions.conf.
IE: the part of the extensions.conf with the following line.
Exten = s,1,MakeReceptionPhoneRing.
This is fantastic if I needed an instant dial to chat with our receptionist.
What I was expecting was that it would go to the part in the extensions.conf
where it detects the numbers dialed.
Ie:
exten = _NXXX,1,Dial(Zap/g1/${EXTEN}|20,t)
Thus allowing us to dial a number and have asterisk direct it to the
appropriate outgoing line.
I could possibly be going about this completely the wrong way. Ideally I
would have thought that a bridge between cards was possible outside of
asterisk.
Any ideas or thoughts on this would be appreciated.
--Dennis
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Re: [Asterisk-Users] Problems with wcte11xp module
It means that you are loading the digium card up with incorrect values. I had it happen to me recently. Paul Hales Technical Manager AsteriskIT - Original Message - From: Jon Farmer [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, March 28, 2006 9:31 PM Subject: [Asterisk-Users] Problems with wcte11xp module Hi I am in the process of commissioning a new * box for our sister company. Unlike us they want their incoming calls delivered on a ISDN 30 not SIP. I have got a TE110P for this project and have compiled the zaptel stuff. However when I modprobe wcte11xp it loads ok but all audio on SIP channels is lost. If I rmmod the driver then audio returns. What is going on? Any ideas? Regards Jon Jon Farmer Telford, Shropshire, UK ___ Yahoo! Messenger - NEW crystal clear PC to PC calling worldwide with voicemail http://uk.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Receptionist Phones
Using a Snom phone, you can monitor a lot more extensions, so I figure it's got to be a Polycom issue. Paul Hales Technical Manager AsteriskIT - Original Message - From: Daniel Hazelbaker [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, March 28, 2006 6:28 AM Subject: Re: [Asterisk-Users] Receptionist Phones Yes, I keep reading on the mailing list archives and the wikis that (wether or not it is indeed a Asterisk issue) Polycom keeps saying that an issue with Asterisk prevents you from monitoring more than 7 total (not per sidecar) extensions. Daniel On Mar 27, 2006, at 12:08 PM, Justin Moore wrote: On 3/27/06, Daniel Hazelbaker [EMAIL PROTECTED] wrote: I have seen that the polycom setup (601+sidecar) works but only for up to 7 phones From what I've seen, each sidecar supports up to 14 additional stations. Three of those along with the 5 buttons on the 601 comes up to 47 on my calculator. Is there a known problem with the 601+sidecars and * that prevents the user from being able to monitor more than 7 extensions? Just curious as I've been leaning toward this for our receptionist as well (only 12 extensions to monitor...) -- Justin Moore aka wantmoore --- www.wantmoore.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Receptionist Phones
Yes - set up about 10 of them at a business last year. Monitoring is fine - picking up calls is a bit iffy at the best of times. (that is, picking up a ringing call by pushing the extension button. *8 works fine) Paul Hales Technical Manager AsteriskIT - Original Message - From: Darrell Long [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, March 28, 2006 7:21 AM Subject: Re: [Asterisk-Users] Receptionist Phones Do you have experience with the Snom phones? We have not had much success getting them to work under Asterisk as a receptionist phone. Specifically, the ability to monitor and pick up calls ringing on other extensions has been a problem. Darrell S. Long BestWeb Corporation [EMAIL PROTECTED] wrote: Using a Snom phone, you can monitor a lot more extensions, so I figure it's got to be a Polycom issue. Paul Hales Technical Manager AsteriskIT - Original Message - From: Daniel Hazelbaker [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, March 28, 2006 6:28 AM Subject: Re: [Asterisk-Users] Receptionist Phones Yes, I keep reading on the mailing list archives and the wikis that (wether or not it is indeed a Asterisk issue) Polycom keeps saying that an issue with Asterisk prevents you from monitoring more than 7 total (not per sidecar) extensions. Daniel On Mar 27, 2006, at 12:08 PM, Justin Moore wrote: On 3/27/06, Daniel Hazelbaker [EMAIL PROTECTED] wrote: I have seen that the polycom setup (601+sidecar) works but only for up to 7 phones From what I've seen, each sidecar supports up to 14 additional stations. Three of those along with the 5 buttons on the 601 comes up to 47 on my calculator. Is there a known problem with the 601+sidecars and * that prevents the user from being able to monitor more than 7 extensions? Just curious as I've been leaning toward this for our receptionist as well (only 12 extensions to monitor...) -- Justin Moore aka wantmoore --- www.wantmoore.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Receptionist Phones
You can get an extension module that adds another 42 buttons. Paul Hales Technical Manager AsteriskIT - Original Message - From: Daniel Hazelbaker [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, March 28, 2006 8:22 AM Subject: Re: [Asterisk-Users] Receptionist Phones Hmm, which phone from Snom are you using for this? I've looked around their website and I can only find 3 VoIP phones, the 300, 320 and 360. The 360 by the looks of it only has 12 buttons you can assign to different extensions; am I missing something or is that the phone and you just do 12 per phone? Daniel On Mar 27, 2006, at 2:28 PM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Yes - set up about 10 of them at a business last year. Monitoring is fine - picking up calls is a bit iffy at the best of times. (that is, picking up a ringing call by pushing the extension button. *8 works fine) Paul Hales Technical Manager AsteriskIT ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 codec problems
What sort of call path are you trying to get working? Paul Hales Technical Manager AsteriskIT - Original Message - From: Rudolf Ladyzhenskii [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, March 26, 2006 10:18 AM Subject: [Asterisk-Users] G729 codec problems Hi, all I have a license for G.729A codec from Digium. When asterisk starts it shows: Jun 17 21:13:59 NOTICE[4040]: codec_g729.c:460 load_module: G.729 transcoding module Copyright (C) 1999-2005 Digium, Inc. Jun 17 21:13:59 NOTICE[4040]: codec_g729.c:461 load_module: This module is supplied under a commercial license granted by Digium, Inc. Jun 17 21:13:59 NOTICE[4040]: codec_g729.c:462 load_module: Please see the full license text supplied by the accompanying Jun 17 21:13:59 NOTICE[4040]: codec_g729.c:463 load_module: register utility, or ask for a copy from Digium. == G.729 Host-ID: cc:20:a3:86:01:93:53:92:2c:37:ae:e7:ad:16:6e:f0:39:f6:88:4e == Found license 'G729-190B962C' providing 1 channels == Found total of 1 G.729 licenses == Registered translator 'g729tolin' from format g729 to slin, cost 20 == Registered translator 'lintog729' from format slin to g729, cost 115 All is fine, however when trying to make a call I am getting: WARNING[4063]: codec_g729.c:170 g729tolin_framein: Out of G.729 Decoder Licenses! No other calls are active. Any ideas what is going on? Thanks, Rudolf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3Com Phones
If you can find yourself a local Asterisk consultant, they should be able to let you see some phones and maybe even try them out. Paul Hales Technical Manager AsteriskIT - Original Message - From: Daniel Hazelbaker [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 27, 2006 3:55 AM Subject: Re: [Asterisk-Users] 3Com Phones Drat, because the 3Com phones looked pretty good for the price. :) Is there somewhere that has a compatibility list for Asterisk with all the phones that are known to work/not work with Asterisk; since apparently VoIP phone companies incorrectly state that they support the SIP protocol (I don't consider, we support SIP as long as it only talks to our server because we tweaked it just a bit to be supported). I am looking for a good 60 phones. We are upgrading our entire phone system (and *old* NEC PBX). We don't need anything fancy on most of the phones, just the usual mid-size business features. Speakerphone, Hold, Park, Transfer, Voicemail; and we need at least 2 attendant stations that can see all in-use phone lines. We are trying to keep the costs (relatively) down, hence using Asterisk instead of a full commercial solution. It is very disconcerting to know the providers are essentially lying about what their phones support. (3Com states their phones are SIP compatible, not 3Com's version of SIP compatibile). Thanks for the info, hopefully somebody will have some recommendations for a good phone brand that actually IS Asterisk compatible. Daniel On Mar 26, 2006, at 12:01 AM, Jared Valentine wrote: I would not recommend the 3Com phones for use with Asterisk. 3Com 3100 series phones do not support SIP with non-3Com systems. They have a basic boot loader which must download code from a 3Com NBX or a 3Com VCX system. If you don't have either of these, then you won't get runtime code on the phone, thereby making it impossible to use the thing with Asterisk. I've heard rumors that the 3103 phones have enough storage space on the phone to store a SIP image, but I don't have any more information than that. As far as 3Com licensing is concerned, it's not per year, it's per- seat (one-time charge), just like any other commercial VoIP PBX vendor (Cisco, Avaya, Shoretel, etc.) Jared Valentine [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM/DECT handsets (was gsm picocells)
If you find anything out, I would like to know. I have tried to find a gsm/wifi phone in the past (in melbourne) and failed. later, Paul Hales Technical Manager AsteriskIT - Original Message - From: James Harper [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, March 25, 2006 11:21 AM Subject: [Asterisk-Users] GSM/DECT handsets (was gsm picocells) Now that I actually try and google for it, I can't find any dual mode GSM/DECT handsets, only pages telling me that they exist without any actual information!!! Does anyone know of any such handsets? (and even better, ones that are available in Australia) I've searched a few of the major gsm manufacturers (nokia, Panasonic, sonyericsson) but their web sites are absolutely pathetic to the point being useless (or maybe I'm just in a bad mood today :) Thanks James -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of James Harper Sent: Friday, 24 March 2006 13:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Re: gsm picocells Steve, Excellent explanation. In a nutshell, it might be better to just use a phone that can automatically switch between GSM and WiFi. Of course, that's limited to handful of handsets. I haven't done any sort of research, but I've been told that GSM+DECT phones are available, and while having them seamlessly switch network types during a call probably isn't possible, they can function as a cordless handset. Can anyone confirm or deny this? James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hopefully a Simple Question?
What about using system(echo) to push stuff into a text file, or the mysql plugin to push stuff over to a database? Paul Hales Technical Manager AsteriskIT - Original Message - From: Clint Tevlin [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, March 26, 2006 9:36 PM Subject: [Asterisk-Users] Hopefully a Simple Question? Hi Guys, I'm writing an app that receives a call on an incoming channel (A), the caller negotiates through a series of prompts and is transferred to an outgoing channel (B) using the Dial cmd. That part works perfectly! For billing I'd like to be able to charge for the time that the first caller is connected to the callee on channel (B) so I can pass on my own outgoing voip costs. How do I do this? I can get the DIALTIME and END time of the call from the cdr but there doesn't seem to be a way of capturing the ANSWERTIME of channel (B) from the dialplan. Any suggestions would be greatly appreciated. clint_in_sydney ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM/DECT handsets (was gsm picocells)
I think the main issue for James and myself is that we can't buy anything in Australia. Paul Hales Technical Manager AsteriskIT - Original Message - From: AR Tarzi [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 27, 2006 10:21 AM Subject: Re: [Asterisk-Users] GSM/DECT handsets (was gsm picocells) Not GSM/DECT but GSM/Wifi phones are available - This is not a recommendation, I don't like what I've seen. try www.imate.com (to start with) .. they have at least three types of GSM phones that do Wifi .. They run windows so there are several sip softwares and one IAX software that work with these - Also Nokia has a GSM phone that does Wifi but that's a symbian (OS) phone (don't know of sip software that works with it). - Original Message - From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 27, 2006 00:48 Subject: Re: [Asterisk-Users] GSM/DECT handsets (was gsm picocells) If you find anything out, I would like to know. I have tried to find a gsm/wifi phone in the past (in melbourne) and failed. later, Paul Hales Technical Manager AsteriskIT - Original Message - From: James Harper [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, March 25, 2006 11:21 AM Subject: [Asterisk-Users] GSM/DECT handsets (was gsm picocells) Now that I actually try and google for it, I can't find any dual mode GSM/DECT handsets, only pages telling me that they exist without any actual information!!! Does anyone know of any such handsets? (and even better, ones that are available in Australia) I've searched a few of the major gsm manufacturers (nokia, Panasonic, sonyericsson) but their web sites are absolutely pathetic to the point being useless (or maybe I'm just in a bad mood today :) Thanks James -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of James Harper Sent: Friday, 24 March 2006 13:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Re: gsm picocells Steve, Excellent explanation. In a nutshell, it might be better to just use a phone that can automatically switch between GSM and WiFi. Of course, that's limited to handful of handsets. I haven't done any sort of research, but I've been told that GSM+DECT phones are available, and while having them seamlessly switch network types during a call probably isn't possible, they can function as a cordless handset. Can anyone confirm or deny this? James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP 301 is slow
And the fact that rebooting a phone is a fairly rare occurence. Paul Hales Technical Manager AsteriskIT - Original Message - From: Avi Miller [EMAIL PROTECTED] To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED]; Kevin P. Fleming [EMAIL PROTECTED] Sent: Monday, March 27, 2006 10:17 AM Subject: Re: [Asterisk-Users] Polycom IP 301 is slow Nick Hoffman wrote: Hrm, well that's disappointing. If they're so slow, why are they so popular? They may be slow to startup, but they're great phones. :) Once the phone has started up, it works like a charm and the sound/call quality is fantastic. -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 - Multiple Server BLF Indications
I have a bad feeling that getting a phone with 160 lights is not going to happen anytime soon. From memory, the snom360 is limited to way less than that. Paul Hales Technical Manager AsteriskIT - Original Message - From: Stuart Elvish - Dallas Delta Corporation Pty Ltd [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, March 27, 2006 10:57 AM Subject: [Asterisk-Users] Snom 360 - Multiple Server BLF Indications Hi, This is a weird request, but does anyone have a Snom 360 monitoring extensions for BLF on several Asterisk servers accross a network? Alternatively, can anyone give me a pointer as to how to setup a Snom 360 to monitor an extension not on it's own server? My scenario is that I have a main site which will have its own server (for storage of call recording data etc because the remote sites don't have the appropriate facilities) and each site has its own embedded system (to ensure that if the network goes down we can still use a normal telephone line). We need an operator telephone with expansion modules (hence the Snom 360) to monitor approximately 180 extensions on approximately 60 asterisk systems (about three extensions per site) so the operator can immediately see any extensions that successfully initiate a call. Any information would be greatly appreciated. Kind Regards Stuart ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 - Multiple Server BLF Indications
I installed 2 Snom360's a few months ago, and 'at the time' only 1 expansion module could be added. (also the fact that the modules draw so much current that it got the POE switch upset!) Have you tested a snom360? I should have one in the lab soon enough. Paul Hales Technical Manager AsteriskIT - Original Message - From: Stuart Elvish - Dallas Delta Corporation Pty Ltd [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 27, 2006 11:41 AM Subject: Re: [Asterisk-Users] Snom 360 - Multiple Server BLF Indications There is an add on module for this phone and according to a source that distributes them here, the modules can be daisy chained until you reach the required number of extensions. I didn't think you could, but that is the information that we have at hand... [EMAIL PROTECTED] wrote: I have a bad feeling that getting a phone with 160 lights is not going to happen anytime soon. From memory, the snom360 is limited to way less than that. Paul Hales Technical Manager AsteriskIT - Original Message - From: Stuart Elvish - Dallas Delta Corporation Pty Ltd [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, March 27, 2006 10:57 AM Subject: [Asterisk-Users] Snom 360 - Multiple Server BLF Indications Hi, This is a weird request, but does anyone have a Snom 360 monitoring extensions for BLF on several Asterisk servers accross a network? Alternatively, can anyone give me a pointer as to how to setup a Snom 360 to monitor an extension not on it's own server? My scenario is that I have a main site which will have its own server (for storage of call recording data etc because the remote sites don't have the appropriate facilities) and each site has its own embedded system (to ensure that if the network goes down we can still use a normal telephone line). We need an operator telephone with expansion modules (hence the Snom 360) to monitor approximately 180 extensions on approximately 60 asterisk systems (about three extensions per site) so the operator can immediately see any extensions that successfully initiate a call. Any information would be greatly appreciated. Kind Regards Stuart -- -- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 - Multiple Server BLF Indications
I had a look at the snom website - and the manual for the expansion module read that only one module can be attached 'currently'. So maybe this has changed. Any ideas? Personally, I like snom phones a lot. I used a snom 200 at my desk at a previous job for almost 2 years. Paul Hales Technical Manager AsteriskIT - Original Message - From: Stuart Elvish - Dallas Delta Corporation Pty Ltd [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 27, 2006 11:41 AM Subject: Re: [Asterisk-Users] Snom 360 - Multiple Server BLF Indications There is an add on module for this phone and according to a source that distributes them here, the modules can be daisy chained until you reach the required number of extensions. I didn't think you could, but that is the information that we have at hand... [EMAIL PROTECTED] wrote: I have a bad feeling that getting a phone with 160 lights is not going to happen anytime soon. From memory, the snom360 is limited to way less than that. Paul Hales Technical Manager AsteriskIT - Original Message - From: Stuart Elvish - Dallas Delta Corporation Pty Ltd [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, March 27, 2006 10:57 AM Subject: [Asterisk-Users] Snom 360 - Multiple Server BLF Indications Hi, This is a weird request, but does anyone have a Snom 360 monitoring extensions for BLF on several Asterisk servers accross a network? Alternatively, can anyone give me a pointer as to how to setup a Snom 360 to monitor an extension not on it's own server? My scenario is that I have a main site which will have its own server (for storage of call recording data etc because the remote sites don't have the appropriate facilities) and each site has its own embedded system (to ensure that if the network goes down we can still use a normal telephone line). We need an operator telephone with expansion modules (hence the Snom 360) to monitor approximately 180 extensions on approximately 60 asterisk systems (about three extensions per site) so the operator can immediately see any extensions that successfully initiate a call. Any information would be greatly appreciated. Kind Regards Stuart -- -- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP 301 is slow
Now that's an interesting comment - most people think the speakerphone on the Polycom is quite good. Paul Hales Technical Manager AsteriskIT - Original Message - From: Denis Galvão - iSolve [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED]; Kevin P. Fleming [EMAIL PROTECTED] Sent: Monday, March 27, 2006 11:47 AM Subject: Re: [Asterisk-Users] Polycom IP 301 is slow The worst thing on all Polycom IP phones is the speaker phone's poor quality. You could not have a conference call using the speakers, only the head phone. Denis. On 26 de mar de 2006, at 21:17, Avi Miller wrote: Nick Hoffman wrote: Hrm, well that's disappointing. If they're so slow, why are they so popular? They may be slow to startup, but they're great phones. :) Once the phone has started up, it works like a charm and the sound/call quality is fantastic. -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM/DECT handsets (was gsm picocells)
Understanding..is not required. ;) PaulH - Original Message - From: James Harper [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 27, 2006 12:23 PM Subject: RE: [Asterisk-Users] GSM/DECT handsets (was gsm picocells) Not GSM/DECT but GSM/Wifi phones are available - This is not a recommendation, I don't like what I've seen. It strikes me as really strange that GSM/Wifi would be available while GSM/DECT is not so much. DECT is a voice technology, while wifi isn't. Still... there's a lot about the world I don't understand :) James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Stability of Asterisk with 2 x TDM400P cards (6 analogue lines)
We ran a system at one site with 2 TDM400's in it to hook up to 8 analog mobile phone gateways. Asterisk was much more reliable than the analog phone gateways, but we still rebooted it once a week. Running on a dual athlon 1800 we picked up very cheaply. regards, Paul Hales Technical Manager AsteriskIT Jared Davison [EMAIL PROTECTED] wrote: I would like to hear from anyone good or bad as what their experience has been in recent times with STABILITY of current builds of Asterisk and drivers for TDM400P. The sort of configuration is: 6 incoming POTS lines. ie. 2 TDM400P cards. I am not concerned with: price points, or the advantages or disadvantages of using POTS vs ISDN technology, but simply RELIABILITY stability of the Asterisk system associated interface hardware and drivers. Do people need to reboot their systems regularly? Thanks in advance. Jared ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] connecting Avaya Partnet with asterisk , TE205P
You will most likely need an E1 crossover cable. With regards to the dialplan programming, E1 connections (internal and external) are much the same - so dial(ZAP/G1) and so on will work fine. regards, Paul HalesTechnical ManagerAsteriskIT - Original Message - From: rnacharya To: asterisk-users@lists.digium.com Sent: Thursday, March 23, 2006 10:01 AM Subject: [Asterisk-Users] connecting Avaya Partnet with asterisk , TE205P Hi ..., I've a TE205P card installed in my asterisk box.Port 1 of my card is connected to service provider.From port 2 I want to connect Avay Partner system.what type of cable I require to connect the partner system (straight/cross over). How the call routing from outside will be done to epbx.Thanks inadvanceregardsrudra ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to make extension groups ???
You will have to break up the card into 4 Zap channels, and them use some clever dialplan work to make sure people are using the right channels. Paul Hales Technical Manager AsteriskIT - Original Message - From: Faisal Inam To: asterisk-users@lists.digium.com Sent: Tuesday, March 21, 2006 8:50 PM Subject: [Asterisk-Users] How to make extension groups ??? Hello All, i am repeating this question for the sixth time but i think i was not explaining the problem correctly. . Now i will try to explain it.. I have 4 telephone lines(PSTN) in my PBX. Now I want to makegroups of the extensions to use that lines. e.g. extensions 12,13 31 are in groupA extensions 14 - 20 are in groupB extensions 21 - 30 are in groupC groupA has access on lines 2,3,4 (Try line 2, if busy try line 3 ,if busy try line 4) groupB has access on lines 3,4 (Try line 3 ,if busy try line 4) groupC has access on line 4 only. (Try line 4 only, and if busy give busy tone) Line 1 is reserved for one extension only. i.e. 11 I will be grateful for an early and complete response. Thanks a lot Faisal Yahoo! MailUse Photomail to share photos without annoying attachments. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime / SIP Peers etc
User - sends you calls Peer - you send calls Friend - Both ways later, Paul HalesTechnical ManagerAsteriskIT - Original Message - From: Douglas Garstang To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, March 22, 2006 5:07 AM Subject: [Asterisk-Users] Realtime / SIP Peers etc Ready to scream here.. 1. After 6 months with Asterisk I'm STILL trying to understand the difference between a SIP user, friend and peer. 2. Exactly what resource does Asterisk use to send MWI to registered phones? I thought it was astdb? 3. It looks like it isn't astdb. It looks like it will only send MWI to a phone if it shows up in 'sip show peers'. 4. WHY then does a reload clear this list? Doesn't this list come from the astdb file? 5. Why is this such a damn mess? Doug. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Native MOH - Convert mp3 to ulaw
Audacity? PaulH - Original Message - From: Douglas Garstang [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, March 22, 2006 8:25 AM Subject: RE: [Asterisk-Users] Native MOH - Convert mp3 to ulaw Good grief. Considering all the libraries and requirements, it'd almost be easier to find some windows software to do this. -Original Message- From: Doug Lytle [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 21, 2006 1:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Native MOH - Convert mp3 to ulaw Douglas Garstang wrote: I tried that earlier today... found it somewhere online... This is what I get... [EMAIL PROTECTED] mp3]# sox -V fpm-calm-river.mp3 -t au -r 8000 -U -b -c 1 fpm-calm-river.ulaw resample -ql sox: resample opts: Kaiser window, cutoff 0.94, beta 16.00 sox: Failed reading fpm-calm-river.mp3: Do not understand format type: mp3 I believe you also need libmp3. Under Mandrake it's called libgmp3. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Users Mailing List Traffic
I was also thinking a list for newbies... PaulH - Original Message - From: Robert La Ferla [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, March 18, 2006 2:33 PM Subject: [Asterisk-Users] Asterisk Users Mailing List Traffic The volume/traffic on this list has been getting pretty heavy. I find it hard to follow certain discussions and there are some that I am not interested in. Perhaps, we could split the list into two: One for discussing hardware (client phones and cards) and one for the software (configuration, problems, etc...) Or some other better scheme that someone can propose. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] gsm picocells
But it looks like a step in the right direction. PaulH - Original Message - From: James Harper [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, March 18, 2006 4:49 PM Subject: RE: [Asterisk-Users] gsm picocells I believe the OP wants to use GSM handsets as extensions, like running your own localized GSM network. That's not the same as using a GSM terminal to connect Asterisk to the cellular network. Correct! IP Access makes such products. http://www.ipaccess.com/products/nanoBTS.htm That looks about right. All problems of spectrum licensing etc aside, the product claims to use Ethernet as the wired access medium, but appears to need to connect to a much meatier box as part of a packaged solution. The site doesn't seem to give much away, including price. Thanks James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE: [Asterisk-Users] MultiBRI in Australia - found one - maybe
Any idea where I can get some of these units in Melbourne? Paul Hales AsteriskIT Faxing received by SpanDSP seems to work fine with these units. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Phelan Sent: Tuesday, 14 March 2006 9:00 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] MultiBRI in Australia - found one - maybe HI Craig and all that is following this. I am running a Vanilla 2.6.11 From cli, misdn show config Misdn General-Config: - VERSION: 0.2.1 - DEBUG_LEVEL: 1 - TRACEFILE: not set - TRACE_CALLS: false - TRACE_DIR: /var/log/ - BRIDGING: no- STOP_TONE_AFTER_FIRST_DIGIT: yes - APPEND_DIGITS2EXTEN: yes- L1_INFO_OK: yes - CLEAR_L3: no- DYNAMIC_CRYPT: no - CRYPT_PREFIX: **- CRYPT_KEYS: test,muh So Far, no dropped calls etc Todays testing will be faxing. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Craig Guy Sent: Monday, 13 March 2006 6:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] MultiBRI in Australia - found one - maybe We went to kernel 2.6.15, and at the same time went to mISDN 0.3.0 rc25, we haven't had any lockups but users are reporting dropped calls. Unfortunately for us this means dropping chan_mISDN in favour of the Cisco router containing BRI cards and then SIP from the Cisco to Asterisk. It may still be possible to use chan_capi with the mISDN drivers for the Drayteks but for us we've run out of time which is a bit of a bummer. I believe the problem is in chan_mISDN which is admittedly still an experimental driver at this stage with release candidates every few days for the past couple weeks. I'm still interested to know how you guys get along with these adapters. As I said, I think the problem is within chan_mISDN at this stage rather than in the USB adapters, so maybe using chan_CAPI on top of mISDN hardware drivers or using chan_vISDN would be the way to go until chan_mISDN matures. Craig - Original Message - From: James Harper [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 13, 2006 3:16 PM Subject: RE: [Asterisk-Users] MultiBRI in Australia - found one - maybe Got my 2 dreytek adapters today... Dropped them on to my test system. After wadding thru my Memory of how to setup mISDN, I had it up and running within about 2 hours. You might be receiving an email from me shortly then if I get stuck. If it wasn't for these annoying public holidays (Labour day in Victoria) mine would probably have arrived today too :) Both of them operating in ptmp with no echo cancel turned on at this ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need help implementing call center features ofAsterisk
Where are you located? We are in Melbourne, Australia. regards PaulH - Original Message - From: Naren Koka [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, March 13, 2006 11:25 PM Subject: [Asterisk-Users] Need help implementing call center features ofAsterisk I am looking for help in implementing call center on Asterisk server. How can we implement predictive dialing? How does it communicate with a CRM system? Are there consultants who can help us setup the system? Thank you, Naren ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IVR woes
WaitExten is also a great item to put at the bottom of your menus.
PaulH
- Original Message -
From: Sean Cook [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, March 10, 2006 8:13 AM
Subject: Re: [Asterisk-Users] IVR woes
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
If memory servers me correctly DigitTimeout and ResponseTimeout are
depricated...
try:
exten = s,13,Set(TIMEOUT(digit)=5)
exten = s,14,Set(TIMEOUT(response)=30)
Sean
Robert P. McKenzie wrote:
Hello all. I'm having a problem debugging an IVR I'm building. I
can't see any reason this shouldn't be working. Firstly the
asterisk version is:
Asterisk SVN-trunk-r7230 built by root @ localhost.localdomain on a
i686 running Linux on 2006-02-17 22:44:48 UTC
Basically the problem is this. While the playbacks are happening
you can push any one of the options and to happily goes off and
does it. However, if you wait until the messages stop playing back
it just hangs up with the error at the bottome of this message.
Any help in finding a solution to this werid problem would be
greatly appreciated.
The IVR context and console logs are:
[lcl-ivr-main]
;;
; ; This is the main number IVR menu system ;
;;
exten = s,1,Answer exten = s,2,NoOp exten = s,3,NoOp exten =
s,4,NoOp exten = s,5,Wait(1) exten =
s,6,Background(LCL/prompt-00) exten =
s,7,Background(LCL/prompt-01) exten =
s,8,Background(LCL/prompt-02) exten =
s,9,Background(LCL/prompt-03) exten =
s,10,Background(LCL/prompt-04) exten =
s,11,Background(LCL/prompt-05) exten =
s,12,Background(LCL/prompt-09) exten = s,13,DigitTimeout,5 exten
= s,14,ResponseTimeout,30
; exten = _1,1,Background(LCL/prompt-20) ; Sales exten =
_1,2,Dial(${SALES}|40|trwo) exten = _1,3,Voicemail([EMAIL PROTECTED])
exten = _1,103,Voicemail([EMAIL PROTECTED]) exten = _1,4,Hangup
; exten = _2,1,Background(LCL/prompt-30) ; Support exten
= _2,2,Dial(${SUPPORT}|40|trwo) exten =
_2,3,Voicemail([EMAIL PROTECTED]) exten =
_2,103,Voicemail([EMAIL PROTECTED]) exten = _2,4,Hangup
; exten = _3,1,Background(LCL/prompt-40) ; Accounts exten
= _3,2,Dial(${ACCOUNTS}|40|trwo) exten =
_3,3,Voicemail([EMAIL PROTECTED]) exten =
_3,103,Voicemail([EMAIL PROTECTED]) exten = _3,4,Hangup
; exten = _4,1,Background(LCL/prompt-50) ; Reception exten
= _4,2,Dial(${RECEPTION}|40|trwo) exten =
_4,3,Voicemail([EMAIL PROTECTED]) exten =
_4,103,Voicemail([EMAIL PROTECTED]) exten = _4,4,Hangup
; exten = _5,1,NoOp ; Dial Extension
; exten = _6,1,Goto(lcl-ivr-menu,s,7) ; Play menu again
; exten = i,1,Goto(lcl-ivr-menu,s,7) ; Return to menu
after a time out exten = t,1,Goto(lcl-ivr-menu,s,7) ;
Return to menu after a time out
Here is he asterisk console output:
-- Accepting AUTHENTICATED call from xx.xx.xx.xx:
requested format = unknown, requested prefs = (), actual format =
ulaw, host prefs = (ulaw|alaw|gsm), priority = mine
-- Executing Goto(IAX2/rob-5, lcl-ivr-main|s|1) in new stack --
Goto (lcl-ivr-main,s,1) -- Executing Answer(IAX2/rob-5, ) in
new stack -- Executing NoOp(IAX2/rob-5, ) in new stack --
Executing NoOp(IAX2/rob-5, ) in new stack -- Executing
NoOp(IAX2/rob-5, ) in new stack -- Executing Wait(IAX2/rob-5,
1) in new stack -- Executing BackGround(IAX2/rob-5,
LCL/prompt-00) in new stack -- Playing 'LCL/prompt-00' (language
'en') -- Executing BackGround(IAX2/rob-5, LCL/prompt-01) in new
stack -- Playing 'LCL/prompt-01' (language 'en') -- Executing
BackGround(IAX2/rob-5, LCL/prompt-02) in new stack -- Playing
'LCL/prompt-02' (language 'en') -- Executing
BackGround(IAX2/rob-5, LCL/prompt-03) in new stack -- Playing
'LCL/prompt-03' (language 'en') -- Executing
BackGround(IAX2/rob-5, LCL/prompt-04) in new stack -- Playing
'LCL/prompt-04' (language 'en') -- Executing
BackGround(IAX2/rob-5, LCL/prompt-05) in new stack -- Playing
'LCL/prompt-05' (language 'en') -- Executing
BackGround(IAX2/rob-5, LCL/prompt-09) in new stack -- Playing
'LCL/prompt-09' (language 'en') -- Executing
DigitTimeout(IAX2/rob-5, 5) in new stack -- Set Digit Timeout
to 5 -- Executing ResponseTimeout(IAX2/rob-5, 30) in new stack
-- Set Response Timeout to 30 == Auto fallthrough, channel
'IAX2/rob-5' status is 'UNKNOWN' -- Hungup 'IAX2/rob-5'
That hangup is Asterisk just dumping out..
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Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
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Re: [Asterisk-Users] IVR woes
Shove a WaitExten at the end of the menu.
PaulH
- Original Message -
From: Robert P. McKenzie [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, March 10, 2006 8:21 AM
Subject: Re: [Asterisk-Users] IVR woes
Sean,
Thanks I've made those changes but still the same problem. The call falls
through if nothing is pushed.
-- Executing Set(IAX2/rob-6, TIMEOUT(digit)=5) in new stack
-- Digit timeout set to 5
-- Executing Set(IAX2/rob-6, TIMEOUT(response)=30) in new stack
-- Response timeout set to 30
== Auto fallthrough, channel 'IAX2/rob-6' status is 'UNKNOWN'
-- Hungup 'IAX2/rob-6'
The hangup is still asterisk dropping the call.
Sean Cook wrote:
If memory servers me correctly DigitTimeout and ResponseTimeout are
depricated...
try:
exten = s,13,Set(TIMEOUT(digit)=5)
exten = s,14,Set(TIMEOUT(response)=30)
Sean
Robert P. McKenzie wrote:
Hello all. I'm having a problem debugging an IVR I'm building. I
can't see any reason this shouldn't be working. Firstly the
asterisk version is:
Asterisk SVN-trunk-r7230 built by root @ localhost.localdomain on a
i686 running Linux on 2006-02-17 22:44:48 UTC
Basically the problem is this. While the playbacks are happening
you can push any one of the options and to happily goes off and
does it. However, if you wait until the messages stop playing back
it just hangs up with the error at the bottome of this message.
Any help in finding a solution to this werid problem would be
greatly appreciated.
The IVR context and console logs are:
[lcl-ivr-main]
;;
; ; This is the main number IVR menu system ;
;;
exten = s,1,Answer exten = s,2,NoOp exten = s,3,NoOp exten =
s,4,NoOp exten = s,5,Wait(1) exten =
s,6,Background(LCL/prompt-00) exten =
s,7,Background(LCL/prompt-01) exten =
s,8,Background(LCL/prompt-02) exten =
s,9,Background(LCL/prompt-03) exten =
s,10,Background(LCL/prompt-04) exten =
s,11,Background(LCL/prompt-05) exten =
s,12,Background(LCL/prompt-09) exten = s,13,DigitTimeout,5 exten
= s,14,ResponseTimeout,30
; exten = _1,1,Background(LCL/prompt-20) ; Sales exten =
_1,2,Dial(${SALES}|40|trwo) exten = _1,3,Voicemail([EMAIL PROTECTED])
exten = _1,103,Voicemail([EMAIL PROTECTED]) exten = _1,4,Hangup
; exten = _2,1,Background(LCL/prompt-30) ; Support exten
= _2,2,Dial(${SUPPORT}|40|trwo) exten =
_2,3,Voicemail([EMAIL PROTECTED]) exten =
_2,103,Voicemail([EMAIL PROTECTED]) exten = _2,4,Hangup
; exten = _3,1,Background(LCL/prompt-40) ; Accounts exten
= _3,2,Dial(${ACCOUNTS}|40|trwo) exten =
_3,3,Voicemail([EMAIL PROTECTED]) exten =
_3,103,Voicemail([EMAIL PROTECTED]) exten = _3,4,Hangup
; exten = _4,1,Background(LCL/prompt-50) ; Reception exten
= _4,2,Dial(${RECEPTION}|40|trwo) exten =
_4,3,Voicemail([EMAIL PROTECTED]) exten =
_4,103,Voicemail([EMAIL PROTECTED]) exten = _4,4,Hangup
; exten = _5,1,NoOp ; Dial Extension
; exten = _6,1,Goto(lcl-ivr-menu,s,7) ; Play menu again
; exten = i,1,Goto(lcl-ivr-menu,s,7) ; Return to menu
after a time out exten = t,1,Goto(lcl-ivr-menu,s,7) ;
Return to menu after a time out
Here is he asterisk console output:
-- Accepting AUTHENTICATED call from xx.xx.xx.xx:
requested format = unknown, requested prefs = (), actual format =
ulaw, host prefs = (ulaw|alaw|gsm), priority = mine
-- Executing Goto(IAX2/rob-5, lcl-ivr-main|s|1) in new stack --
Goto (lcl-ivr-main,s,1) -- Executing Answer(IAX2/rob-5, ) in
new stack -- Executing NoOp(IAX2/rob-5, ) in new stack --
Executing NoOp(IAX2/rob-5, ) in new stack -- Executing
NoOp(IAX2/rob-5, ) in new stack -- Executing Wait(IAX2/rob-5,
1) in new stack -- Executing BackGround(IAX2/rob-5,
LCL/prompt-00) in new stack -- Playing 'LCL/prompt-00' (language
'en') -- Executing BackGround(IAX2/rob-5, LCL/prompt-01) in new
stack -- Playing 'LCL/prompt-01' (language 'en') -- Executing
BackGround(IAX2/rob-5, LCL/prompt-02) in new stack -- Playing
'LCL/prompt-02' (language 'en') -- Executing
BackGround(IAX2/rob-5, LCL/prompt-03) in new stack -- Playing
'LCL/prompt-03' (language 'en') -- Executing
BackGround(IAX2/rob-5, LCL/prompt-04) in new stack -- Playing
'LCL/prompt-04' (language 'en') -- Executing
BackGround(IAX2/rob-5, LCL/prompt-05) in new stack -- Playing
'LCL/prompt-05' (language 'en') -- Executing
BackGround(IAX2/rob-5, LCL/prompt-09) in new stack -- Playing
'LCL/prompt-09' (language 'en') -- Executing
DigitTimeout(IAX2/rob-5, 5) in new stack -- Set Digit Timeout
to 5 -- Executing ResponseTimeout(IAX2/rob-5, 30) in new stack
-- Set Response Timeout to 30 == Auto fallthrough, channel
'IAX2/rob-5' status is 'UNKNOWN' -- Hungup 'IAX2/rob-5'
That hangup is Asterisk just dumping out..
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Re: [Asterisk-Users] MultiBRI in Australia - found one - maybe
Noted - I may need to grab one for an install coming up. regards, PaulH - Original Message - From: Avi Miller [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, March 09, 2006 10:11 PM Subject: Re: [Asterisk-Users] MultiBRI in Australia - found one - maybe James Harper wrote: One use for the multi BRI card though, especially one that can do NT mode, is that you can use it to trunk to a legacy BRI PBX, which is why I'm still interested in finding one for use in Australia. I'm using the Eicon Diva Server V-4BRI (~$2,200 each). They are awesome. Onboard hardware echo cancellation, native CAPI drivers for Linux (source available) and chan_capi compatibility. They can do both NT and TE mode, so you can use them to connect to legacy BRI PBXs as well. cYa, Avi -- National Manager - Special Projects Melbourne / Sydney / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom 501 power over ethernet
I have installed several hundred polycom's, and I have never seen a 500/501 with a power jack. All with the inline cable, as you mention. Of course, if someone can provide photo evidence I will stand corrected. PaulH - Original Message - From: The VoIP Connection [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Tuesday, March 07, 2006 4:26 AM Subject: RE: [Asterisk-Users] Polycom 501 power over ethernet I've seen a lot of IP501 and I've never seen one with a power jack. According to Polycom they all use the cable. Possibly it was an IP500? -Mike Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Douglas Garstang [mailto:[EMAIL PROTECTED] Sent: Monday, March 06, 2006 10:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom 501 power over ethernet No, some IP 501's have the inline cable and some have the power jack. -Original Message- From: Paul Hales [mailto:[EMAIL PROTECTED] Sent: Sunday, March 05, 2006 8:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom 501 power over ethernet The IP300/301 has the power jack, the IP500/501 the inline cable. PaulH On Sun, 2006-03-05 at 20:56 -0700, Douglas Garstang wrote: Not true. Some do and some don't. Some have a place to plug a separate DC adapter, and some have the inline power, where the adapter plugs into the ethernet cable. Not sure which ones are newer, and which are older. -Original Message- From: Michael Welter [mailto:[EMAIL PROTECTED] Sent: Sun 3/5/2006 6:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Polycom 501 power over ethernet The IP501 does not have a power jack. You'll need one of the Polycom cables. William M Conlon wrote: My recollection of the marketing fluff was that we would just use our legacy network (cables) and the devices at both ends would figure out whether they were sourcing, sinking, or neither. In the case of the 501, it's the special Polycom cable, either with or without provision for an AC power adapter, that powers the phone. That's what I meant by saying the '501' itself is not compliant with 802.3af -- it needs a separate thingamajig [tech jargon :)]to be powered. Anyway I had hoped that I could just plug a CAT-5 patch cable from my RJ45 wall outlet into the phone. On Mar 5, 2006, at 5:17 PM, Michael Welter wrote: As I understand 802.3af, the phones go through a negotiation with the unit supplying the power. I don't think it's a matter of -48VDC on a particular pair. I remember a schematic from years ago--it had each of the receive pair and the transmit pair going into a transformer winding, and that winding had a center tap for PoE. This is not something that *I* am going to screw with. The IP501 telephone set is the same for both PoE and local power. With the PoE cable, the 802.3af electronics (the negotiator) is a plastic thing in the cable. For the local power, there is a plastic thingie toward the wall end of the cable, and you plug the wall wart into the plastic thingie. Notice the advanced technical jargon here With local power, there is still only one cable one the desk--the power plugs into the cable towards the wall. Except for a power interruption, this has all the advantages of PoE. William M Conlon wrote: I saw that Polycom offered a cable (not stocked anywhere), at $40 a pop for 802.3af connections. That's what made me think the phone itself is NOT 802.3af compliant. Presumably, for $40, there's more than a fuse in that special cable. On Mar 5, 2006, at 4:31 PM, Paul Hales wrote: For Polycom IP500/501's and IP300/301's you need a special polycom POE cable. When you buy Polycom phones you can usually specify POE or powerpack. PaulH On Sun, 2006-03-05 at 16:23 -0800, William M Conlon wrote: When I bought two Polycom 501 SIP phones, I naively thought they were Power-over-Ethernet (IEEE 802.3af) because they were powered over ethernet. Silly me. Polycom must have some odd voltage or funny way of injecting the power, because the POE switch I bought for them (Netgear [EMAIL PROTECTED]) won't power them, though if I use the Polycom-supplied AC adapter and ethernet power injector cable, they work with the switch in either its powered or unpowered ports. Anyhow, I hadn't seen any mention of how people power these phones, as I had planned on centralizing phone power on a UPS to supply my
Re: [Asterisk-Users] Polycom 501 power over ethernet
Can you provide a photo of this? I am interested in seeing it! PaulH - Original Message - From: Douglas Garstang [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, March 07, 2006 2:13 AM Subject: RE: [Asterisk-Users] Polycom 501 power over ethernet No, some IP 501's have the inline cable and some have the power jack. -Original Message- From: Paul Hales [mailto:[EMAIL PROTECTED] Sent: Sunday, March 05, 2006 8:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom 501 power over ethernet The IP300/301 has the power jack, the IP500/501 the inline cable. PaulH On Sun, 2006-03-05 at 20:56 -0700, Douglas Garstang wrote: Not true. Some do and some don't. Some have a place to plug a separate DC adapter, and some have the inline power, where the adapter plugs into the ethernet cable. Not sure which ones are newer, and which are older. -Original Message- From: Michael Welter [mailto:[EMAIL PROTECTED] Sent: Sun 3/5/2006 6:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Polycom 501 power over ethernet The IP501 does not have a power jack. You'll need one of the Polycom cables. William M Conlon wrote: My recollection of the marketing fluff was that we would just use our legacy network (cables) and the devices at both ends would figure out whether they were sourcing, sinking, or neither. In the case of the 501, it's the special Polycom cable, either with or without provision for an AC power adapter, that powers the phone. That's what I meant by saying the '501' itself is not compliant with 802.3af -- it needs a separate thingamajig [tech jargon :)]to be powered. Anyway I had hoped that I could just plug a CAT-5 patch cable from my RJ45 wall outlet into the phone. On Mar 5, 2006, at 5:17 PM, Michael Welter wrote: As I understand 802.3af, the phones go through a negotiation with the unit supplying the power. I don't think it's a matter of -48VDC on a particular pair. I remember a schematic from years ago--it had each of the receive pair and the transmit pair going into a transformer winding, and that winding had a center tap for PoE. This is not something that *I* am going to screw with. The IP501 telephone set is the same for both PoE and local power. With the PoE cable, the 802.3af electronics (the negotiator) is a plastic thing in the cable. For the local power, there is a plastic thingie toward the wall end of the cable, and you plug the wall wart into the plastic thingie. Notice the advanced technical jargon here With local power, there is still only one cable one the desk--the power plugs into the cable towards the wall. Except for a power interruption, this has all the advantages of PoE. William M Conlon wrote: I saw that Polycom offered a cable (not stocked anywhere), at $40 a pop for 802.3af connections. That's what made me think the phone itself is NOT 802.3af compliant. Presumably, for $40, there's more than a fuse in that special cable. On Mar 5, 2006, at 4:31 PM, Paul Hales wrote: For Polycom IP500/501's and IP300/301's you need a special polycom POE cable. When you buy Polycom phones you can usually specify POE or powerpack. PaulH On Sun, 2006-03-05 at 16:23 -0800, William M Conlon wrote: When I bought two Polycom 501 SIP phones, I naively thought they were Power-over-Ethernet (IEEE 802.3af) because they were powered over ethernet. Silly me. Polycom must have some odd voltage or funny way of injecting the power, because the POE switch I bought for them (Netgear [EMAIL PROTECTED]) won't power them, though if I use the Polycom-supplied AC adapter and ethernet power injector cable, they work with the switch in either its powered or unpowered ports. Anyhow, I hadn't seen any mention of how people power these phones, as I had planned on centralizing phone power on a UPS to supply my Asterisk server and POE switch. Now the question is: Can the Polycom AC-powered injector be used with a standard ethernet patch cable: switch :: Polycom injector cable :: RJ45 coupler :: patch cable :: Polycom 501 which would allow me to power the Polycom AC adapters by my UPS. Or do I need to provide a UPS at each phone and run the ethernet like switch :: patch cable :: RJ45 coupler :: Polycom injector cable :: Polycom 501 thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and
Re: [Asterisk-Users] Polycom 501 power over ethernet
Totally correct - according to me at least. PaulH - Original Message - From: Ken D'Ambrosio [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, March 07, 2006 8:25 AM Subject: Re: [Asterisk-Users] Polycom 501 power over ethernet On Mon, March 6, 2006 4:19 pm, [EMAIL PROTECTED] wrote: I have installed several hundred polycom's, and I have never seen a 500/501 with a power jack. All with the inline cable, as you mention. Of course, if someone can provide photo evidence I will stand corrected. I think the confusion here is the different *ways* the 300/500/600 do PoE: 301 has a power brick, just like (say) a Grandstream. 501 has _almost_ PoE: the cable is (as noted above) in-line, but this might confuse someone differentiating with the 301. 601 has true PoE, where you've got your PoE switch, a stock Ethernet cable, and the phone -- nothing else, and no special cabling required. -Ken (purveyor of fine differentiations) PaulH - Original Message - From: The VoIP Connection [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Tuesday, March 07, 2006 4:26 AM Subject: RE: [Asterisk-Users] Polycom 501 power over ethernet I've seen a lot of IP501 and I've never seen one with a power jack. According to Polycom they all use the cable. Possibly it was an IP500? -Mike Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Douglas Garstang [mailto:[EMAIL PROTECTED] Sent: Monday, March 06, 2006 10:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom 501 power over ethernet No, some IP 501's have the inline cable and some have the power jack. -Original Message- From: Paul Hales [mailto:[EMAIL PROTECTED] Sent: Sunday, March 05, 2006 8:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom 501 power over ethernet The IP300/301 has the power jack, the IP500/501 the inline cable. PaulH On Sun, 2006-03-05 at 20:56 -0700, Douglas Garstang wrote: Not true. Some do and some don't. Some have a place to plug a separate DC adapter, and some have the inline power, where the adapter plugs into the ethernet cable. Not sure which ones are newer, and which are older. -Original Message- From: Michael Welter [mailto:[EMAIL PROTECTED] Sent: Sun 3/5/2006 6:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Polycom 501 power over ethernet The IP501 does not have a power jack. You'll need one of the Polycom cables. William M Conlon wrote: My recollection of the marketing fluff was that we would just use our legacy network (cables) and the devices at both ends would figure out whether they were sourcing, sinking, or neither. In the case of the 501, it's the special Polycom cable, either with or without provision for an AC power adapter, that powers the phone. That's what I meant by saying the '501' itself is not compliant with 802.3af -- it needs a separate thingamajig [tech jargon :)]to be powered. Anyway I had hoped that I could just plug a CAT-5 patch cable from my RJ45 wall outlet into the phone. On Mar 5, 2006, at 5:17 PM, Michael Welter wrote: As I understand 802.3af, the phones go through a negotiation with the unit supplying the power. I don't think it's a matter of -48VDC on a particular pair. I remember a schematic from years ago--it had each of the receive pair and the transmit pair going into a transformer winding, and that winding had a center tap for PoE. This is not something that *I* am going to screw with. The IP501 telephone set is the same for both PoE and local power. With the PoE cable, the 802.3af electronics (the negotiator) is a plastic thing in the cable. For the local power, there is a plastic thingie toward the wall end of the cable, and you plug the wall wart into the plastic thingie. Notice the advanced technical jargon here With local power, there is still only one cable one the desk--the power plugs into the cable towards the wall. Except for a power interruption, this has all the advantages of PoE. William M Conlon wrote: I saw that Polycom offered a cable (not stocked anywhere), at $40 a pop for 802.3af connections. That's what made me think the phone itself is NOT 802.3af compliant. Presumably, for $40, there's more than a fuse in that special cable. On Mar 5, 2006, at 4:31 PM, Paul
Re: [Asterisk-Users] incoming calls dropout on PRI over TE110p
Just trying to think - are you using the standard E1 setup from ATP? I have found that the settings on their website work pretty well. Also - have you tried to put an answer in your dialplan? That might keep the dialplan open.. later, PaulH - Original Message - From: Paul C [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, March 01, 2006 5:15 PM Subject: Re: [Asterisk-Users] incoming calls dropout on PRI over TE110p Paul C wrote: I am running Asterisk 1.0.9 and have been running all my calls through a VSP over a IAX2 trunk however we have recently purchased and connected a TE110p to a PRI ( E1 with 16 voice channels ) through Optus. I can make outgoing calls via it fine, however incoming calls are dropped after a few seconds ( or as soon as a command like Playback, or the call is picked up if forwarded to a SIP extensions ). SNIP overlapdial should usually be no in my experience. Okay I've turned that to no with no change. I've just got off the phone to Optus and apparently they had a client in melbourne last week and they fixed the problem by turning crc checking off at the optus end. I don't suppose that was anybody on here ? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] incoming calls dropout on PRI over TE110p
Funnily - I have set up 2 or 3 pri's over the last few weeks on 1.2x and haven't had any issues. (and one of those is a high load situation - passthru at an outbound call centre) PaulH Melbourne - Original Message - From: James Sturges [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Sunday, March 05, 2006 7:52 AM Subject: RE: [Asterisk-Users] incoming calls dropout on PRI over TE110p I would not upgrade to 1.2.x yet, I did and now have taken asterisk out of the site. It is sending CRC errors )to Telsta, drops all calls once a day for 1 second, calls getting stuck, quite unpleasant! I was advised to roll back to 1.0.9 Asterisk, Zaptel and Libpri. James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul C Sent: Wednesday, 1 March 2006 4:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] incoming calls dropout on PRI over TE110p Paul C wrote: I am running Asterisk 1.0.9 and have been running all my calls through a VSP over a IAX2 trunk however we have recently purchased and connected a TE110p to a PRI ( E1 with 16 voice channels ) through Optus. I can make outgoing calls via it fine, however incoming calls are dropped after a few seconds ( or as soon as a command like Playback, or the call is picked up if forwarded to a SIP extensions ). SNIP overlapdial should usually be no in my experience. Okay I've turned that to no with no change. I've just got off the phone to Optus and apparently they had a client in melbourne last week and they fixed the problem by turning crc checking off at the optus end. I don't suppose that was anybody on here ? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk Question
I actually got it all working - but it's great to see where we did the same
thing, and where we differ.
I ended up using the 'pop' perl command - inside a loop to go back one item
at a time through my list
PaulH
- Original Message -
From: Michael Collins [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, March 02, 2006 6:21 AM
Subject: RE: [Asterisk-Users] Re: Asterisk Question
Thanks for this example - it has really got me started!
Paul,
I did some tinkering and I think I found something that might be
helpful. If not, I did at least learn quite a bit about AGI scripting
and dialplan writing! :)
Okay, first I created a pretend file with numbers: /tmp/numbers.txt
It is simply a list of numbers from 001 to 020, like so:
001
002
003
...
019
020
Since you have to check the file in reverse order, I have the AGI script
read in the values, assign them to an array, @NUMS, then I perform a
reverse on @NUMS:
@NUMS = reverse @NUMS;
Now 020 is the first value in the array, or $NUMS[0].
019 is the second value, or $NUMS[1].
If the number of previous messages that have been read (which I put in
the dialplan variable NUM_MSGS_READ) is 0, then the next number is
referenced by grabbing $NUMS[0]. If the number of previous messages
that have been read is 1, then the next number is referenced by
grabbing $NUMS[1], and so on.
I created two different dialplan extensions so I could tinker. The
first one hard codes the value of NUM_MSGS_READ to 3. Thus the return
value of NEXT_NUM is $NUMS[3], which is 017. The second one lets the
caller enter in the value for NEXT_NUM, which is good for testing
purposes. I tried to add some validation logic to handle exceptions. I
didn't test it thoroughly but hopefully it illustrates the point.
Here are the dialplan extensions:
; AGI test2.1
; hard coded NUM_MSGS_READ
;
exten = 555,1,Noop(Starting AGI test)
exten = 555,n,Answer
exten = 555,n,Wait(1)
exten = 555,n,SetVar(NUM_MSGS_READ=3)
exten = 555,n,Playback(beep)
exten = 555,n,AGI(agi_var_test2.pl)
exten = 555,n,GotoIf($[${NEXT_NUM:1} = INVALID ]?invalid:ok)
exten = 555,n(ok),SayDigits(${NEXT_NUM})
exten = 555,n,Wait(1)
exten = 555,n,Playback(beep)
exten = 555,n,Hangup
exten = 555,n(invalid),Playback(invalid)
exten = 555,n,Hangup
; AGI test2.2
; For kicks, let the caller set the value of NUM_MSGS_READ
;
exten = 556,1,Noop(Starting AGI test)
exten = 556,n,Answer
;exten = 556,n,SetVar(NUM_MSGS_READ=X)
exten = 556,n,Wait(1)
exten = 556,n,Playback(please-enter-the)
exten = 556,n,Playback(digits)
exten = 556,n,Read(NUM_MSGS_READ,,2,,,10)
exten = 556,n,Playback(beep)
exten = 556,n,AGI(agi_var_test2.pl)
exten = 556,n,GotoIf($[${NEXT_NUM:1} = INVALID ]?invalid:ok)
exten = 556,n(ok),SayDigits(${NEXT_NUM})
exten = 556,n,Wait(1)
exten = 556,n,Playback(beep)
exten = 556,n,Hangup
exten = 556,n(invalid),Playback(invalid)
exten = 556,n,Hangup
exten = 556,t,Playback(vm-goodbye)
exten = 556,t,Hangup
The Read() application in exten 556 is configured to accept up to two
digits with a 10 second digit timeout. On a timeout it plays a goodbye
message. In both extensions, there is a test to make sure that NEXT_NUM
doesn't contain the string INVALID - if it does then we go to a
priority and voice that to the caller. The AGI script will return
1INVALID or 2INVALID depending upon why the script failed. See the
notes in the code for details.
Here's the AGI script, agi_var_test2.pl:
#!/usr/bin/perl
#
# agi_var_test2.pl
#
# Reads in info from file /tmp/numbers.txt into array @NUMS
# Uses Asterisk var NUM_MSGS_READ to count from end of @NUMS
# Assigns appropriate value from @NUMS to Asterisk var NEXT_NUM
#
use strict;
use warnings;
use Asterisk::AGI;
# the AGI object
my $agi = new Asterisk::AGI;
# pull AGI variables into %input
my %input = $agi-ReadParse();
my @NUMS = (); # Array to hold data from file
my $number;# Value to be sent back to Asterisk
my $infile = '/tmp/numbers.txt';
open(FILEIN,,$infile) or die $infile - $!\n;
while(FILEIN) {
chomp;
# Perform some data validation, just to be safe
s/^\s|\s+$|^\s+$//;# remove whitespace, just in case
next unless length;# skip if nothing left...
next unless m/^\d+$/; # skip if line contains any non-digits
push @NUMS,$_;
} # while(FILEIN)
close(FILEIN);
# reverse array to make it easier to find our value
# e.g. last line of file becomes first item in @NUMS:
# $NUMS[0] = last item of file
# $NUMS[1] = next to last item of file, etc.
@NUMS = reverse @NUMS;
my $arraypos = $agi-get_variable('NUM_MSGS_READ');
if ( ! $arraypos ) {
# if we get to this point, something is wrong...
$agi-set_variable('NEXT_NUM','1INVALID');
exit(1);# 1INVALID = AGI script could not read NUM_MSGS_READ
}
# since @NUMS is zero based, $arraypos = next
Re: [Asterisk-Users] Polycom 501
Ummm- from memory the sequence is TRANSFER - NUMBER - SEND - chat to other person - TRANSFER. PaulH - Original Message - From: MBIT Technologies To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Thursday, March 02, 2006 11:02 PM Subject: RE: [Asterisk-Users] Polycom 501 AMP is being run but it seems the transfer needs to be configured in the phone somewhere so when you press the transfer button its like hitting #. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark AufflickSent: Thursday, 2 March 2006 10:51 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Polycom 501 One thing to keep in mind when someone says "Asterisk does that by default" is that a lot of people have AMP installed, and an AMP installation includes extra configuration and features as well as the web interface. It may be that there is phone-specific config installed with AMP that is not installed in a base Asterisk installation. Cheers,Mark.-- Mark Aufflicke: [EMAIL PROTECTED]w: www.pumptheory.com (business)w: mark.aufflick.com (personal)p: +61 438 700 647f: +61 2 9436 4737 On 3/2/06, MBIT Technologies [EMAIL PROTECTED] wrote: I guess it doesn't work by default on my phone. You still need to press hash to transfer calls. The transfer button doesn't work. Where do I set it? Regards Mark Brooker T: 02 4959 8670 M: 0415 846 865 F: 02 9882 0947 E: [EMAIL PROTECTED] W: http://www.mbit.com.au -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Anton KrallSent: Thursday, 2 March 2006 3:47 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Polycom 501 Those buyttons do work with asterisk by default... what kind of problems are you having? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of MBIT TechnologiesSent: Wednesday, March 01, 2006 7:56 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: [Asterisk-Users] Polycom 501 Hi Guys Just a quick question regarding on the 501, has anyone been able to configure the transfer button and messaging buttons to work with asterisk? Can you share a configuration to do this? Thanks in advance. iBurst Wireless Broadband from $34.95/month - Platform Networks Spam Virus Filtering by Mail SecurityTo report SPAM forward the spam message to: [EMAIL PROTECTED] ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk Question
Thanks to everyone who helped me with this!
I will post my code next week (when I am back at the workplace where I did
this)
later,
PaulH
- Original Message -
From: Paul Hales [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, March 01, 2006 4:36 PM
Subject: Re: [Asterisk-Users] Re: Asterisk Question
Perfection!
PaulH
On Tue, 2006-02-28 at 22:24 -0700, Darren Wiebe wrote:
my ( $var1, $var2, $var3 ) =
@ARGV;
and so on and so forth.
Good Luck
Darren Wiebe
[EMAIL PROTECTED]
Paul Hales wrote:
Thanks for this example - it has really got me started!
Short question - how can I put a variable into my perl script?
I imagine it's something like
exten = 780,1,AGI(agi_ret_val2.pl|${back})
But how can I get my perl script to pick this value up?
Again - thanks to everyone who has helped me with this.
later,
PaulH
On Tue, 2006-02-28 at 11:25 -0800, Michael Collins wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Monday, February 27, 2006 7:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: Asterisk Question
I was going to see if I can execute a bash script as an AGI - just
looking
around the internet for examples at the moment.
Anybody got an example spare?
I'm just a bit stuck on how to start this, but I am quite comfortable
writing asterisk dialplan stuff and bash scripts
later,
PaulH
Paul,
I'm a Perl guy myself. Here's a simple dialplan extension and AGI
script written in Perl and using the very cool Asterisk::AGI module:
; AGI test
exten = 555,1,Noop(Starting AGI test)
exten = 555,n,Answer
exten = 555,n,Wait(1)
exten = 555,n,Playback(beep)
exten = 555,n,AGI(agi_var_test.pl)
exten = 555,n,SayDigits(${EXTERN_VAR})
exten = 555,n,Wait(1)
exten = 555,n,Playback(beep)
exten = 555,n,Hangup
Here's the Perl script:
#!/usr/bin/perl
#
# agi_var_test.pl
#
# Reads in info from file /etc/group
# assigns asterisk GID to Asterisk variable EXTERN_VAR
#
use strict;
use warnings;
use Asterisk::AGI;
# the AGI object
my $agi = new Asterisk::AGI;
# pull AGI variables into %input
my %input = $agi-ReadParse();
my $infile = '/etc/group';
open(FILEIN,,$infile) or die $infile - $!\n;
while(FILEIN) {
chomp;
next unless m/^asterisk/;
my @REC = split :,$_;
print STDERR agi_var_test.pl: Setting EXTERN_VAR to $REC[2]\n;
$agi-set_variable(EXTERN_VAR, $REC[2]);
last;
} # while(FILEIN)
close(FILEIN);
Basically the script just parses /etc/group until it finds the
asterisk
entry. It then parses the data line and extracts the GID. Finally,
it
prints the value to STDERR (for debugging purposes) and then assigns
the
value to EXTERN_VAR.
This is more a proof-of-concept than anything else, but it does show
the
value of AGI and Asterisk::AGI.
-MC
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Re: [Asterisk-Users] Re: Asterisk Question
Numbers from a text file - just need to read them back in one at a time.
PaulH
- Original Message -
From: Michael Collins [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, March 01, 2006 5:14 AM
Subject: RE: [Asterisk-Users] Re: Asterisk Question
Paul,
Just curious - what kind of stuff are you reading from the file?
-MC
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Monday, February 27, 2006 7:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: Asterisk Question
I was going to see if I can execute a bash script as an AGI - just
looking
around the internet for examples at the moment.
Anybody got an example spare?
I'm just a bit stuck on how to start this, but I am quite comfortable
writing asterisk dialplan stuff and bash scripts
later,
PaulH
steve [EMAIL PROTECTED] wrote:
From: Paul Hales [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk question
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain
Any idea how to read an external file, grab some stuff and push it
back
into an Asterisk variable?
I can do it the other way with:
system(echo ${UNIQUEID} = /home/ast/curr_calls)
but I'm a bit stumped on how to go the other way around
much thanks,
Paul Hales
I'll go out on a limb here and take a guess that it could be done as
an
AGI script that incorporates SED (http://www.gnu.org/software/sed/)
and
AWK (http://www.gnu.org/software/gawk/gawk.html). I've used both
for
some bash scripting in the past. . .
Regards,
Steve Cayona
Super Technologies, Inc.
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Re: [Asterisk-Users] Re: Asterisk Question
That's getting pretty close - thanks for that.
I just couldn't find any decent info on the web about working with AGI.
regards,
PaulH
- Original Message -
From: Michael Collins [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, March 01, 2006 6:25 AM
Subject: RE: [Asterisk-Users] Re: Asterisk Question
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Monday, February 27, 2006 7:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: Asterisk Question
I was going to see if I can execute a bash script as an AGI - just
looking
around the internet for examples at the moment.
Anybody got an example spare?
I'm just a bit stuck on how to start this, but I am quite comfortable
writing asterisk dialplan stuff and bash scripts
later,
PaulH
Paul,
I'm a Perl guy myself. Here's a simple dialplan extension and AGI
script written in Perl and using the very cool Asterisk::AGI module:
; AGI test
exten = 555,1,Noop(Starting AGI test)
exten = 555,n,Answer
exten = 555,n,Wait(1)
exten = 555,n,Playback(beep)
exten = 555,n,AGI(agi_var_test.pl)
exten = 555,n,SayDigits(${EXTERN_VAR})
exten = 555,n,Wait(1)
exten = 555,n,Playback(beep)
exten = 555,n,Hangup
Here's the Perl script:
#!/usr/bin/perl
#
# agi_var_test.pl
#
# Reads in info from file /etc/group
# assigns asterisk GID to Asterisk variable EXTERN_VAR
#
use strict;
use warnings;
use Asterisk::AGI;
# the AGI object
my $agi = new Asterisk::AGI;
# pull AGI variables into %input
my %input = $agi-ReadParse();
my $infile = '/etc/group';
open(FILEIN,,$infile) or die $infile - $!\n;
while(FILEIN) {
chomp;
next unless m/^asterisk/;
my @REC = split :,$_;
print STDERR agi_var_test.pl: Setting EXTERN_VAR to $REC[2]\n;
$agi-set_variable(EXTERN_VAR, $REC[2]);
last;
} # while(FILEIN)
close(FILEIN);
Basically the script just parses /etc/group until it finds the asterisk
entry. It then parses the data line and extracts the GID. Finally, it
prints the value to STDERR (for debugging purposes) and then assigns the
value to EXTERN_VAR.
This is more a proof-of-concept than anything else, but it does show the
value of AGI and Asterisk::AGI.
-MC
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Re: [Asterisk-Users] Re: Asterisk Question
I need to read through the numbers in reverse order, so I can decide which messages to play to people. I was using a variable to mark how many messages they had read, and each time read a number further back in the list. PaulH - Original Message - From: Michael Collins [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, March 01, 2006 9:22 AM Subject: RE: [Asterisk-Users] Re: Asterisk Question That's getting pretty close - thanks for that. I just couldn't find any decent info on the web about working with AGI. Ditto. However, I pieced some stuff together by sifting through my well-worn copy of TFOT and bouncing around between the wiki, the sample AGI scripts and asterisk.gnuinter.net. I had never sat down to write an AGI script before - I hadn't needed one - but I thought, How hard can it be? Ugh. The Asterisk::AGI module is very handy, and I highly recommend it. I've only written one AGI script in my life (up to now) but I've written 10's of thousands of lines of Perl and I know a good module when I see one. Now, as far as reading one number at a time, are you reading one line from the file at a time? In other words, does your file look something like this: 1237 2340 5434 123 9173 ... I was wondering what you were planning on doing with each number. Were you looking at multiple passes to your AGI script, or were you going to run the script once and collect all of the data in one pass? -MC ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk Question
I was going to see if I can execute a bash script as an AGI - just looking
around the internet for examples at the moment.
Anybody got an example spare?
I'm just a bit stuck on how to start this, but I am quite comfortable writing
asterisk dialplan stuff and bash scripts
later,
PaulH
steve [EMAIL PROTECTED] wrote:
From: Paul Hales [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk question
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain
Any idea how to read an external file, grab some stuff and push it back
into an Asterisk variable?
I can do it the other way with:
system(echo ${UNIQUEID} = /home/ast/curr_calls)
but I'm a bit stumped on how to go the other way around
much thanks,
Paul Hales
I'll go out on a limb here and take a guess that it could be done as an
AGI script that incorporates SED (http://www.gnu.org/software/sed/) and
AWK (http://www.gnu.org/software/gawk/gawk.html). I've used both for
some bash scripting in the past. . .
Regards,
Steve Cayona
Super Technologies, Inc.
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Re: [Asterisk-Users] $ for an hr of asterisk support
Where are you located? Paul Hales Melbourne, Australia - Original Message - From: Sam Tam [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Tuesday, February 21, 2006 8:52 PM Subject: [Asterisk-Users] $ for an hr of asterisk support Hello I need some asterisk expert on setting up incoming DID on asterisk Please email me back or msn me on sam__tam AT hotmail DOT com $£ waiting.. Sam ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] immediate pick up in s
This sounds more like a dialplan issue - and what has got to do with anything? PaulH - Original Message - From: Alejandro Vargas [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, February 21, 2006 8:16 PM Subject: [Asterisk-Users] immediate pick up in s I'm configuring a sip trunk. My problem is if I configure the sip device to dial to a sip phone, it works ok but when I dials to s or , asterisk picks up the call immediatly and places it's own ring tone instead of waiting until one of the extension configured for answer the call picks up. Is there a way to avoid it? Is it a problem of the sip trunk? Should I post this question to devel list? -- Alejandro Vargas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Download Asterisk: The Future Of Telephony[More Info]
Jenn bought hers at Borders(jenn @ ains from this list) PaulH - Original Message - From: Bob McDowell [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, February 22, 2006 5:32 AM Subject: RE: [Asterisk-Users] Download Asterisk: The Future Of Telephony[More Info] Speaking of this book, where can I get it? Obviously I can read the pdf, but I lack the facility to print it in any usable fashion. The labor and materials that I have spent on trying to print it thus far probably outweighs the cost of the silly thing. Is it only available online, or do you think Barnes and Noble, Borders, etc might have it? Bob McDowell -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Burke Sent: Monday, February 20, 2006 6:22 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Download Asterisk: The Future Of Telephony [More Info] One thing I forgot to mention: I also cropped the registration and cut marks off the sides of the pages. If you want the uncropped version, get: http://www.alexburke.ca/asterisk-tfot-uncropped.pdf Sorry about the excessive noise, but I figured I should mention this. Date: Mon, 20 Feb 2006 18:55:50 -0500 To: asterisk-users@lists.digium.com From: Alexander Burke [EMAIL PROTECTED] Subject: Download Asterisk: The Future Of Telephony Hello, list! I'm hosting a mirror of the book Asterisk: The Future Of Telephony by O'Reilly Press, published under the Creative Commons license; I believe this license allows me to do this, but if I'm mistaken, please let me know. I've taken the liberty of fixing the page numbers so Acrobat is now aware of the correct number of each page, and shrinking the filesize with Acrobat's Reduce File Size tool (while still maintaining compatibility with Acrobat 4.0, apparently). I bought a paper copy before I knew the book was available online, but it's good enough that even had I known it was available online, I still would have bought it on paper. You're welcome to download it and keep it on hand -- it makes for EXCELLENT reading: http://www.alexburke.ca/asterisk-tfot.pdf -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What business IP phone to use
It's funny, but I found it more challening to buy a second hand car than to buy phones. PaulH - Original Message - From: mustardman29 [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, February 22, 2006 4:57 AM Subject: [Asterisk-Users] What business IP phone to use I have been struggling with this issue for about a year now. There were just too many IP phones to choose from at all sorts of price points and not enough information about any of them. Now I am looking at the situation again and if anything it has gotten worse. There are even more phones and all sorts of opinions. For every person that says phone x is great there is someone else complaining about it. I ended up buying a Grandstream GXP2000 and an Aastra 9133i to test so I pretty much know what those two phones are about. Lot's of people talking about Polycom phones but they still seem to have their problems and since they don't officially support Asterisk I have my concerns. I really don't want to have to keep buying phones to find out for myself as it get's expensive real fast. Is there any unbiased comparison of various phones and features anywhere. If someone wrote a book I'd buy it but it would probably be obsolete before it was published with the rate of new IP phone introductions and firmware revisons. I hear some people praising the GXP2000 phones and I gotta wonder what they are smokin (regardless of firmware revison) so I just don't know who to believe anymore. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail 0 for operator call routing
It's the 'o' extension in your context that hits the voicemail. (thats a lower case o not a zero) PaulH - Original Message - From: Paul Tinsley [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, February 22, 2006 3:19 AM Subject: [Asterisk-Users] Voicemail 0 for operator call routing Does anyone know of a way to specify what extension is dialed when 0 is pressed in the voicemail system. I have a situation where there is more than one secretary and they want the 0 to redirect to the appropriate secretary for the two groups of people. So an example would be: 555-1234 - voicemail - Secretary 1 555-1235 - voicemail - Secretary 2 Any help would be greatly appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Call centre - * hang's up
- Original Message - From: Tomislav Parčina [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, February 20, 2006 10:46 PM Subject: [Asterisk-Users] Re: Call centre - * hang's up In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I think it's a bit of a known fault - the attended transfer function does not work from the queue system. It would be nice if it did, though. Hi Paul! Is there any explanation about this? Is that something that will change? It probably should change - somebody different asks the question on the list here every month or so. Has anyone logged this onto bugs.digium.com??? PaulH ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] good voip
Where are you located? That makes a big difference! PaulH Melbourne, Australia - Original Message - From: CyberSource [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, February 21, 2006 7:37 AM Subject: [Asterisk-Users] good voip Can anyone recommend a good voip provider? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Call centre - * hang's up
But using the native transfer on the phone causes the system to think the agent is still on the call Is this still correct? It was last time I tested it, 6 odd months ago. PaulH - Original Message - From: Eric ManxPower Wieling [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, February 21, 2006 10:44 AM Subject: Re: [Asterisk-Users] Re: Call centre - * hang's up [EMAIL PROTECTED] wrote: - Original Message - From: Tomislav Parčina [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, February 20, 2006 10:46 PM Subject: [Asterisk-Users] Re: Call centre - * hang's up In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I think it's a bit of a known fault - the attended transfer function does not work from the queue system. It would be nice if it did, though. Hi Paul! Is there any explanation about this? Is that something that will change? It probably should change - somebody different asks the question on the list here every month or so. Has anyone logged this onto bugs.digium.com??? As I understand it, only DTMF TRANSFER has a problem. If you use the native transfer support of the device you are using it should work just fine. We do it all the time. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple TDM400P's in a single machine
Depending on the setup, you might actually need 8 x FXO. PaulH - Original Message - From: Marc Archer To: asterisk-users@lists.digium.com Sent: Tuesday, February 21, 2006 10:24 AM Subject: [Asterisk-Users] Multiple TDM400P's in a single machine Hi All, Can someone give me a definite answer as to wether or not you can reliably run multiple TDM400Ps in the same machine? I need 4 x FXO and 4 x FXS to connect to both the PSTN and existing key system, but I have seen several threads suggesting that this is not a supported configuration Thanks, Marc. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wildfire messsaging server
On an Asterisk server- yes. [EMAIL PROTECTED] - not me. PaulH - Original Message - From: Dean Collins To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, February 20, 2006 1:17 AM Subject: [Asterisk-Users] Wildfire messsaging server http://www.jivesoftware.org/ Is anyone running a wildfire messaging server on the same pc as their asterisk server? Is anyone specifically running it on an [EMAIL PROTECTED] installation? TIA, Dean ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Connecting two phones with different codecs
To translate between g729 and g711 you need to buy some licences. PaulH - Original Message - From: Lisa Wolf [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, February 16, 2006 11:38 AM Subject: Re: [Asterisk-Users] Connecting two phones with different codecs Paul Hales wrote: From memory, it's really down to making the right selections in sip.conf We did a large installation, with phones at the Head Office using g711 and phones at remote sites using g729. Asterisk happily transcoded for us. Which was great. It keeps telling me (for g711 to G729) that there's no translation path. I'll look and see if I've forgotten to load a module. Thanks for the reply. Lisa PalH ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] configure TE205P on [EMAIL PROTECTED]
The genzaptelconf doesn't work with E1/T1 cards in my experience. You will have to configure it by hand. PsulH - Original Message - From: nik600 [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, February 11, 2006 11:09 PM Subject: [Asterisk-Users] configure TE205P on [EMAIL PROTECTED] hi i'm trying to configure a TE205P on [EMAIL PROTECTED] i've edited /etc/sysconfig/zaptel adding this line: MODULES=$MODULES wct2xxp now, when the system is loading, i can see that the wct2xxp module is loaded correctly but if i try the command: /usr/local/sbin/genzaptelconf i get: STOPPING ASTERISK STOPPING FOP SERVER Generating '/etc/zaptel.conf' Generating '/etc/asterisk/zapata-auto.conf' STOPPING ASTERISK STOPPING FOP SERVER Unloading zaptel hardware drivers:. Removing zaptel module: ERROR: Module zaptel is in use by wct4xxp [FAILED] Loading zaptel framework: [ OK ] Waiting for zap to come online...OK Loading zaptel hardware modules: wct2xxpRunning ztcfg: [ OK ] SETTING FILE PERMISSIONS Permissions OK STARTING ASTERISK Asterisk ended with exit status 1 Asterisk died with code 1. Automatically restarting Asterisk. Asterisk ended with exit status 1 Asterisk died with code 1. Automatically restarting Asterisk. - Asterisk could not start! Use 'tail /var/log/asterisk/full' to find out why. - Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) the file asterisk.ctl esists... [EMAIL PROTECTED] /]# ls -la /var/run/asterisk/asterisk.ctl srwxr-xr-x 1 asterisk asterisk 0 Feb 11 07:06 /var/run/asterisk/asterisk.ctl and this is what is reported in the logs: [EMAIL PROTECTED] /]# tail /var/log/asterisk/full Feb 11 07:06:05 VERBOSE[4808] logger.c: [chan_zap.so]Feb 11 07:06:05 VERBOSE[4808] logger.c: [chan_zap.so] = (Zapata Telephony w/PRI) Feb 11 07:06:05 VERBOSE[4808] logger.c: == Parsing '/etc/asterisk/zapata.conf': Feb 11 07:06:05 VERBOSE[4808] logger.c: == Parsing '/etc/asterisk/zapata.conf': Found Feb 11 07:06:05 VERBOSE[4808] logger.c: == Parsing '/etc/asterisk/zapata-auto.conf': Feb 11 07:06:05 VERBOSE[4808] logger.c: == Parsing '/etc/asterisk/zapata-auto.conf': Found Feb 11 07:06:05 VERBOSE[4808] logger.c: == Parsing '/etc/asterisk/zapata_additional.conf': Feb 11 07:06:05 VERBOSE[4808] logger.c: == Parsing '/etc/asterisk/zapata_additional.conf': Found Feb 11 07:06:05 WARNING[4808] chan_zap.c: Unable to specify channel 1: No such device or address Feb 11 07:06:05 ERROR[4808] chan_zap.c: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Feb 11 07:06:05 ERROR[4808] chan_zap.c: Unable to register channel '1-23' Feb 11 07:06:05 WARNING[4808] loader.c: chan_zap.so: load_module failed, returning -1 Feb 11 07:06:05 WARNING[4808] loader.c: Loading module chan_zap.so failed! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Issues in Australia? Ringing, iaxy etc
There are quite a few Asterisk systems running in Australia, so you should be fine PaulH Melbourne - Original Message - From: Chris Earle (CBL) [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, February 10, 2006 3:31 AM Subject: [Asterisk-Users] Issues in Australia? Ringing, iaxy etc Hi all, getting a server going wiht a few TDM400's and some phones, and some IAXys too I haven't heard any issues about AU phones being able to RING in Australia, like the problem in the UK with ring capacitors on the BT system. Are there any problems like that? Also, with the iaxy's -- they should work (and ring) in Australia right? The only hint I'm seeing around is the use of notransfer=yes in the iax.conf for the iaxy entry Basically, just hoping for a smooth transition over to the asterisk system Cheers -- Chris Earle System Solutions Specialist -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Two Lines, Two Businesses
Alexander Lopez [EMAIL PROTECTED] wrote: You can do all that you want and MORE with 'hand-knitted' configuraton files. That's what I figured; I was really wondering whether AMP would specifically *prevent* that kind of configuration. I'm presuming that a hand-knitted extensions.conf, with separate contexts for the two trunks and two businesses is the right approach - would someone stop me, please, if I'm wrong? ;) Sounds right - set different context in the zap file, and incoming calls will arrive at different contexts nicely. PaulH ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP PAX gateway to PSTN
Thanks for the answer. Is this PSTN gateway is something for a VoIP company to setup in order to connect their VoIP calls to the Telco's PSTN then to the end phone? I don't think Australia treat this as illegal. But I m not sure how much the Telco will charge from IP PAX (or PSTN) gateway to end phone. Assuming that there are 1000 VoIP calls thru Telco's PSTN to end phones, how will these calls get calculated? is the charge will be per-call basis? There are lots of Asterisk users here in Australia...and it's not illegal. You will probably have to discuss charging with your Telco. PaulH ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP PAX gateway to PSTN
How much your telco is going to charge you for the PSTN calls depends on your arrangement with the telco... Usually, with proper volume interconnects (say you order a PRI line), these calls are charged per second. Do I really need PRI T1 line when I initially setup VoIP network? How do you want to send calls out onto the public phone network? (and here in Australia, we run E1, not T1) PaulH ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP PAX gateway to PSTN
There are lots of Asterisk users here in Australia...and it's not illegal. You will probably have to discuss charging with your Telco. PaulH _ Thanks for the answers. I really appreciate that. It may be better for me to talk to local Telco for further price negotiation. Going through a VOIP termination service is also good for testing. There are quite a few here in Australia. PaulH ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Do we need a QOS switch ?
Hi, We have 10 people on our network and each person will have a SIP phone connected to our Asterisk server. All phones, Asterisk, other servers and users workstations will be using the same network. The question is: would I need a QOS device to give SIP traffic a chance? Our internal network is 100M. We will have a ISDN30 for outgoing calls. No calls will be made over the internet. As long as the current infrastructure is decent, you should be fine without a separate voice switch. PaulH ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] IP PAX gateway to PSTN
Hi, I m still not quite understand why I need E1/T1 PRI line with VoIP calls to normal phone line. I thought I can send calls out thru a normal home telphone line. If this is the case, I will just need to pay each VoIP call to phone line at 20 - 30 cents / call. Then you will need to buy an analogue phone card. PaulH ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] IP PAX gateway to PSTN
Thanks for the answers. I really appreciate that. It may be better for me to talk to local Telco for further price negotiation. Going through a VOIP termination service is also good for testing. When going thru a VoIP termination service, do I also need to have a IP PAX gateway? To not answer your question, but send you in the right direction: If you use a terminaton service, you do not need a phone card. PaulH ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Blocked Callerid
Do you mean 1-800 number? I don't really know the answer - I will have to ask next time I visit. PaulH - Original Message - From: Joe Pukepail To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, February 02, 2006 7:47 AM Subject: Re: [Asterisk-Users] Blocked Callerid Do they have an 800 number? If so perhaps their 800 number provider is doing it via DTMF. Search around on the internet, I believe the standard format for the DTMF is *CALLERID*CALLEDNUMBER* (or perhaps reversed). On 2/1/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I have been discussing an asterisk solution with a company that has a custom written dialogic based solution. The issue is that their dialogic solution can read callerid from incoming calls, even if the callerid is blocked. I have read before that Asterisk can do this, and they want me to make sure that their new system will be able to do this. A quick poke around inside the zaptel source code wasunproductive... Any ideas? PaulH ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fax possibilities
h - using hylafax or asterisk? PaulH - Original Message - From: James Harper [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, February 02, 2006 10:13 AM Subject: [Asterisk-Users] fax possibilities I am trying to set up a linux based faxing solution for a client, and have found that the modem they have (ancient dataplex external unit) just isn't up to the job. It talks to some remote fax machines but not others. A new external modem ranges from AUD$75 to AUD$400, which got me thinking of other possibilities... #1 FXO PCI card (more expensive for 1 port, probably cheaper for 2+) #2 Sipura SPA3000 #3 Grandstream ATA488 I assume there will be no problem getting #1 working as a fax modem, but what about #2 and #3? Has anyone done this before? Some brief googling shows that it is possible, but not that it has been done... James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] limit sip sessions
Shouldn't all sip users have different usernames? (or am I missing some vital detail here?) PaulH - Original Message - From: Miguel [EMAIL PROTECTED] To: Asterisk User List asterisk-users@lists.digium.com Sent: Friday, February 03, 2006 3:21 AM Subject: [Asterisk-Users] limit sip sessions hi, is there a way to limit the sip session per username?. i mean, if i have a sip session with asterisk using xxx as username, nobody can register with that username until my session is terminated. thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Blocked Callerid
I have been discussing an asterisk solution with a company that has a custom written dialogic based solution. The issue is that their dialogic solution can read callerid from incoming calls, even if the callerid is blocked. I have read before that Asterisk can do this, and they want me to make sure that their new system will be able to do this. A quick poke around inside the zaptel source code wasunproductive... Any ideas? PaulH ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dumb Dialout Question
You are missing the G exten = 190,1,Dial(ZAP/g0/800111) Assumes that you have set up your zap card as group 0. (zap/g1 is probably more realistic) later, PaulH - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, February 02, 2006 7:11 AM Subject: [Asterisk-Users] Dumb Dialout Question I'm still trying to learn some parts of Asterisk, so sorry in advance for the dumb question! How do I set up an extension to dial out to the PSTN through my ZAP interfaces? I want the ability to have a ring group that will ring all of the phones in an office and then ring cell phones if nobody answers. I'm sure this is simple to do but I'm at a loss. I have tried the following configs in extensions.conf to no avail: exten = 190,1,Dial(ZAP/[EMAIL PROTECTED]) ; Cell Phone exten = 190,1,Dial(ZAP/800111) ; Cell Phone exten = 190,1,Dial(SIP/[EMAIL PROTECTED]) ; Cell Phone exten = 190,1,Dial(ZAP/800111) ; Cell Phone Thank you in advance! ___ Sent by ePrompter, the premier email notification software. Free download at http://www.ePrompter.com. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No Audio on Local Machine, Remote works fine
Is ztdummy loaded properly? I had a similar problem with a system recently. PaulH - Original Message - From: Hadar Pedhazur [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, February 02, 2006 8:16 AM Subject: [Asterisk-Users] No Audio on Local Machine, Remote works fine I don't even know where to begin. I run a lot of production Asterisk servers, for a couple of years now, with no real problems. We built a brand new box, CentOS 4.2, and installed Asterisk 1.2.4 from source tarball(s). Built fine, and started up fine. Any attempts to do local audio (e.g. a Playback(welcome)) results in complete silence. Worse, the Playback command will hang forever (even if the file is tiny), so it's not just not being heard, it's like the command is waiting to do something. In one specific case (and only in case), I'll hear a 1/2 second burst of audio, like it's about to start, and then dead air. The Record command creates a zero length file if the format is ulaw, and hangs forever after that, and a wav format is always 44 bytes before the hang. If I run the demo-echo-test, I don't hear the prompt, and it hangs on the Playback. OK, now for the weirdness ;-). If I connect this Asterisk to one of our other servers, and dial the echo test on the remote server through this same server, I hear the prompts, and can hear my voice echoed correctly, so this same Asterisk server will happily forward the audio in both directions, it just won't generate it. This is with notransfer=yes, so this Asterisk is staying in the audio stream. I'm stumped, and any help or pointers in the right direction will be greatly appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyway to do this?
If callerid is received, it will be displayed on the sip phones. My guess would be that it's not coming in on the analog line in the first place. PaulH Scott Geist [EMAIL PROTECTED] wrote: How do you retreive the caller id on incoming analog lines and display the id on the sip phones on the network? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE: [Asterisk-Users] Blocked Callerid
I think the customer is more interested in whether Asterisk can do what their current system does, rather than discuss the legalites of this. PaulH Chris Bagnall [EMAIL PROTECTED] wrote: The issue is that their dialogic solution can read callerid from incoming calls, even if the callerid is blocked. I don't know what the laws on such things are where you're located, but you might want to check into the legality of actually doing that. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE: [Asterisk-Users] Blocked Callerid
I think they have a 1-800 number so you might be right. But the important question is still - will Asterisk support this? PaulH Alexander Lopez [EMAIL PROTECTED] wrote: They are using ANI instead of CallerID. If they have an 800 number thya have the right to know who is calling them because they are paying for the call. the *ANI*DNIS* format is known as Feature Grooup D. Alex From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Pukepail Sent: Wednesday, February 01, 2006 3:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Blocked Callerid Do they have an 800 number? If so perhaps their 800 number provider is doing it via DTMF. Search around on the internet, I believe the standard format for the DTMF is *CALLERID*CALLEDNUMBER* (or perhaps reversed). On 2/1/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I have been discussing an asterisk solution with a company that has a custom written dialogic based solution. The issue is that their dialogic solution can read callerid from incoming calls, even if the callerid is blocked. I have read before that Asterisk can do this, and they want me to make sure that their new system will be able to do this. A quick poke around inside the zaptel source code was unproductive... Any ideas? PaulH ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/ -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to start a playback after the called partypicks up?
I hope the staff are not answering the question with their middle digit. That would be rude. PaulH - Original Message - From: Ronald Wiplinger [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, February 01, 2006 12:45 AM Subject: [Asterisk-Users] How to start a playback after the called partypicks up? 1. I want to call somebody and, as soon (and not before) a playback should be played. How can I do that? 2. How can I accept dtmf tones with such calls? Example: System calls all staff and ask them a question. The staff will answer with a digit! The playback should start when the staff picks up. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Evening in Melbourne: Feb 2!
Sounds great - be there and be square. PaulH jurgen [EMAIL PROTECTED] wrote: Hi all, Come one come all! We're having the next Asterisk evening at the Fujitsu Centre for Excellence! This is Fuji's state of the art show-off centre - they're promising lots of interesting toys to play with. As usual, we'll be discussing developments in Asterisk land over the past couple of months. If you've got some interesting toys yourself, please bring them along! Date: Thursday February 2nd (the day after tomorrow!) Time: 7pm - late Place: Fujitsu Centre for Excellence, 1 Southbank Boulevard, Southbank (Inside the Pacific Internet building) After Fujitsu makes us leave, we'll be heading to a local cafe or pub to continue our collective phone geekiness. If you have trouble finding us, please give me a call (my number is in the .sig below). Many thanks to Joseph Sirucka for securing the venue for this evening, and to Fujitsu for letting us use it. More about Fujitu's playground here: http://www.fujitsu.com/au/news/pr/archives/2005/20051109-01.html Best, ...jurgen -- [EMAIL PROTECTED] is jurgen's gmail address. Visit http://jurgen.ca/ for more yummy goodness. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How's the best way to set up agents...
Base your internal dialplan around Dial(Agent/xxx) rather than DIAL(SIP/xxx) for calling extensions. later, PaulH Ben Ferguson [EMAIL PROTECTED] wrote: So I'm trying to set up queues and agents and am trying to figure out the best way to set up what I need to do. And what I need to do is basically get Asterisk to mimic my company's current phone system. As close as possible of course. And my main problem is queues and agents. Currently, for our queues and agents, a person is assigned a hot-desk extension, which they use to login to any phone and then they can send and receive calls at that extension. There is no seperate extension and agent id--they are pretty much the same thing. But the extension moves around with them to wherever they log in. The advantage is that they always have the same extension. When no one is logged into a phone, the phone is assigned a catch all username called no user which has limited dialing capabilities. With Asterisk, when you log in an agent, they assume the extension of the phone that they have just logged in under. Yes, if they are a member of a queue, they will always receive calls from that queue regardless of what extension they are at, but for DID and internal calls, you would never know which extension to dial to reach a person setup in such a way. So here's what I've come up with (but I, of course, still have questions...): Match the agent ID to an extension. Assign an agent their ID and then assign a certain working area, and a assign certain phone to that working area and assign that phone an extension that is the same as their agent id. The pitfall here is that if you do it this way, only one person could utilize that working area and that phone. Agents working in shifts at the same work area and same phone would not work--unless multiple agents used the same extension, which again kills your DID and internal calling. Of course, you could assign the extension the phone based on time, but what if you want to have open seating and you want any agent to sit in any work area at any given time? You can see it starts to get messy when you try to work it out this way. Also, if you did do the matched extension and agent id, if a person was re-assigned to a new work area, they either have to take their phone with them, or you get to setup a new phone with their extension. Perhaps there is something in Asterisk that I don't know about that could benefit me here? I'm thinking another way to do something like what I need is via XML, but I'm not exactly sure how to do it this way. Can you assign a phone a certain extension and then give it an option for logging in using their agent id and then based on their agent id, push a new XML file out that assigns their specific extension. Can you re-assign a new extension to a phone this way? I believe this would be a decent way to set this up (if the XML files aren't too complicated) but I'm not exactly sure how to do it. Any suggestions, pointers, directions...? Thanks! Ben Ferguson ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium hardware
Move to PRI - it will be much more fun than working with analog. PaulH - Original Message - From: Cisco - Kameko To: asterisk-users@lists.digium.com Sent: Wednesday, January 25, 2006 6:17 PM Subject: [Asterisk-Users] Digium hardware Hello, I want to setup an asterisk pabx. I want to understand more on what hardware (PCI cards) i will need to do this. I have 5 xchange lines and 30 extensions within our offices. I have just finished installing Fedora Core and downloaded asterisk-1.2.3.tar.gzand zaptel-1.2.2.tar.gzwhich i want to install. In need or your advise ASAP Regards, SOUL ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] checking voicemail via trunk
Or put it on one of your DID's, I suppose. (if you have E1/T1) PaulH - Original Message - From: Sean Cook [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 27, 2006 8:16 AM Subject: Re: [Asterisk-Users] checking voicemail via trunk -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 just set up an extension that goes straight into voicemail main from your autoattendant... if you know your parties extension, please dial... exten = 770,1,VoiceMailMain() OK Computer wrote: Can someone point me to a link describing how to enable voicemail checking by dialing into asterisk from off-site? Gabe Herbert ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFD2Twvy9wPyZpnL2URAk3HAJ9u8WCXUsxUozSoQgYwtA/YAJqKeACgqSvW rycX/htYYKqUS6B6+QYtRm8= =YIKp -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with sip setup because can't receive calls
What error do you get when trying to call the SIP phones? PaulH - Original Message - From: abc def To: asterisk-users@lists.digium.com Sent: Wednesday, January 25, 2006 11:58 PM Subject: [Asterisk-Users] Help with sip setup because can't receive calls Hi all, I readmany posts on asterisk mail site and been trying many different thingsbut still I can't get my sip phones to work with asterisk. I have a full blown-up voip netwok with two asterisk servers connected to pstn networkwith iax phones and cisco sccp phones which all work fine. however, I have been struggeling to configure my sip phones (polycom 601, Aastra 480i and cisco 9760) to work with asterisk. I can call out from sip phones to anywhere else but not receive phone calls. I can see the phones on "sip show registry" and "sip show peers" but no track phone calls for sip. can you please shed some light on me how to go about solving this problem? thank you and best regards, Ama Do you Yahoo!?With a free 1 GB, there's more in store with Yahoo! Mail. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware recommendations
Why would you want to get line 16? PaulH - Original Message - From: Dane Reugger [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, January 25, 2006 4:09 AM Subject: Re: [Asterisk-Users] Hardware recommendations If you have 16 call appearances or lines - how do you get to line 16 - type in some code? Adam Goryachev wrote: On Mon, 2006-01-23 at 23:00 -0700, Douglas Garstang wrote: Polycom SoundPoint 601 has 4 'lines'. :) Actually, it has 6 'lines' :) Needing a 4 line phone is going to decrease your choices of phones. Why do you need 4 lines? He probably hasn't worked out the difference between 'call appearances' and lines yet Even a polycom 301 (with 2 'lines' can handle loads of calls, I think the limit is something like 16 per line, configurable in the xml file). Regards, Adam ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Announcing PodMail 1.0 (GPL)
Cute? But it can use LDAP... PaulH - Original Message - From: Ben Klang [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 24, 2006 3:58 AM Subject: [Asterisk-Users] Announcing PodMail 1.0 (GPL) Hello Asterisk Community. While sitting at lunch the other day I had a typical napkin-prototype idea: What if I could make my Asterisk Voicemail accessible as a Podcast in iTunes? Three hours later with the help of two friends I had a working proof of concept. Now we are releasing the polished version of this idea as PodMail 1.0 PodMail brings together open-source telephony and Podcasting to create a new, useful way of accessing voicemail and podcasting. PodMail integrates with Asterisk to provide a secure podcast of your voicemail. Supporting authentication directly against voicemail.conf or using an LDAP directory, PodMail allows you to subscribe to your own voicemail box. Each time you dock your iPod, your new voicemails will sync right along. Listen to your voicemail at your convenience and without using cell minutes. PodMail also allows for a brand new type of PodCasting. Unchain Podcasting from the computer! Configure PodMail for public access and you have a ready-to-run PodCast. Updating your Podcast is as easy as phone call. Moblogging has never been so easy or flexible. Live Demo: Do not miss out our live demo at http://podmail.alkaloid.net/ Leave us a message in one of our mailboxes, subscribe to one of the PodMail Podcasts, then see and hear your message immediately! Check out the PodMail Documentation and Installation Notes at http://projects.alkaloid.net. PodMail is released under the terms of the GPL. Enjoy! /BAK/ -- Ben Klang Alkaloid Networks http://projects.alkaloid.net [EMAIL PROTECTED] 404.475.4850 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Announcing PodMail 1.0 (GPL)
Sorry about this - I hit send by accident while I was still writing the email. Pretend it never happened. PaulH - Original Message - From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 24, 2006 7:26 AM Subject: Re: [Asterisk-Users] Announcing PodMail 1.0 (GPL) Cute? But it can use LDAP... PaulH - Original Message - From: Ben Klang [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 24, 2006 3:58 AM Subject: [Asterisk-Users] Announcing PodMail 1.0 (GPL) Hello Asterisk Community. While sitting at lunch the other day I had a typical napkin-prototype idea: What if I could make my Asterisk Voicemail accessible as a Podcast in iTunes? Three hours later with the help of two friends I had a working proof of concept. Now we are releasing the polished version of this idea as PodMail 1.0 PodMail brings together open-source telephony and Podcasting to create a new, useful way of accessing voicemail and podcasting. PodMail integrates with Asterisk to provide a secure podcast of your voicemail. Supporting authentication directly against voicemail.conf or using an LDAP directory, PodMail allows you to subscribe to your own voicemail box. Each time you dock your iPod, your new voicemails will sync right along. Listen to your voicemail at your convenience and without using cell minutes. PodMail also allows for a brand new type of PodCasting. Unchain Podcasting from the computer! Configure PodMail for public access and you have a ready-to-run PodCast. Updating your Podcast is as easy as phone call. Moblogging has never been so easy or flexible. Live Demo: Do not miss out our live demo at http://podmail.alkaloid.net/ Leave us a message in one of our mailboxes, subscribe to one of the PodMail Podcasts, then see and hear your message immediately! Check out the PodMail Documentation and Installation Notes at http://projects.alkaloid.net. PodMail is released under the terms of the GPL. Enjoy! /BAK/ -- Ben Klang Alkaloid Networks http://projects.alkaloid.net [EMAIL PROTECTED] 404.475.4850 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware recommendations
Needing a 4 line phone is going to decrease your choices of phones. Why do you need 4 lines? PaulH - Original Message - From: Dane Reugger [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, January 24, 2006 2:45 PM Subject: [Asterisk-Users] Hardware recommendations We would like to test Asterisk in our small office - 5 users. We are a small computer shop in New Orleans and would like to offer VoIP and Asterisk to our clients but we are very new to VoIP and Asterisk. We feel the best way to learn is to jump in. We've signed up w/ Teliax and setup a D-link phone that works OK - but our goal is an Asterisk PBX. We would like to avoid as many costly mistakes as possible. We plan on keeping 2 analog lines for emergencies, VoIP down, 911, credit card machine, and Fax machine as we understand Fax and CC machines are very unreliable w/ VoIP but plan on integrating them in to the Asterisk with an FXO card We are looking for recommendations for VoIP phones and a 1 or 2 Line FXO(?) card. I suspect the first is kinda vague and the latter is a Digium card. Just looking for solutions, brands, and even vendors that are known to work well. Phone needs 4 lines, Hold, VM, Caller ID Any advice appreciated Thanks, Dane ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fw: setting outgoing caller ID by the queue anextension is logged into
Use different prefixes for different outgoing calls? (I know that's a nuisance though) PaulH - Original Message - From: Franklin Webb To: asterisk-users@lists.digium.com Sent: Tuesday, January 24, 2006 7:34 AM Subject: [Asterisk-Users] Fw: setting outgoing caller ID by the queue anextension is logged into Greetings fellow list members, I am trying to add some tricky functionality to Asterisk dialplan and I was curious if anyone else has come up with a solution to something like this. Basically I have phone representatives that log into one of several queues (not using chan Agent, welog inby the extension), and frequently these agents have to make attended transfer calls to outside numbers. This transfer basically amounts to a new outgoing call. I have been asked to set the caller ID for these outgoing calls based on the queue the phone representative is currently logged in to. Unfortunetly I cannot think of a way to do this. The incomming and outgoing calls are two different calls. I have considered using DBPut and DBGet to store this information in a database. This might work, but I am also concerned about the overhead involved. I cannot think of a way to do this using global variables since I need to store a seperate value for each extension. Has anyone run into an issue like this and come up with a solution? Any thoughts are much appreciated. Thank you, Franklin Webb Assistant IT Project Leader Inter Media Marketing Solutions ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware recommendations
Hmmmdo you mean that the system needs 4 lines? Or that you need a phone that can make 4 concurrent calls? PaulH - Original Message - From: Dane Reugger [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 24, 2006 4:32 PM Subject: Re: [Asterisk-Users] Hardware recommendations I need 2 concurrent connections but prefer 4 - we spend a lot of time on the phone here. Once things recover in New Orleans we will probably build our staff up to 7 or 8 quickly. -Dane [EMAIL PROTECTED] wrote: Needing a 4 line phone is going to decrease your choices of phones. Why do you need 4 lines? PaulH - Original Message - From: Dane Reugger [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, January 24, 2006 2:45 PM Subject: [Asterisk-Users] Hardware recommendations We would like to test Asterisk in our small office - 5 users. We are a small computer shop in New Orleans and would like to offer VoIP and Asterisk to our clients but we are very new to VoIP and Asterisk. We feel the best way to learn is to jump in. We've signed up w/ Teliax and setup a D-link phone that works OK - but our goal is an Asterisk PBX. We would like to avoid as many costly mistakes as possible. We plan on keeping 2 analog lines for emergencies, VoIP down, 911, credit card machine, and Fax machine as we understand Fax and CC machines are very unreliable w/ VoIP but plan on integrating them in to the Asterisk with an FXO card We are looking for recommendations for VoIP phones and a 1 or 2 Line FXO(?) card. I suspect the first is kinda vague and the latter is a Digium card. Just looking for solutions, brands, and even vendors that are known to work well. Phone needs 4 lines, Hold, VM, Caller ID Any advice appreciated Thanks, Dane ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNOM 190 Daylight Savings
Sadly, most of the phone manufacturers do not understand Southern Hemisphere daylight savings. Don't know why, but they just don't. PaulH Rod Bacon [EMAIL PROTECTED] wrote: I've posted this to SNOM, but was wondering wheter anyone here has issues with SNOM 190 phones not showing the correct DST adjusted time (using the latest firmware). -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600 Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SMS to fixed phone line
This would be great to have working! PaulH Blackburn, Vic - Original Message - From: James Harper [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, January 19, 2006 3:08 PM Subject: [Asterisk-Users] SMS to fixed phone line Telstra (Australian Telco) has recently introduced a feature to allow the sending of SMS direct to fixed analogue lines, with an appropriate handset. As best as I can figure out, this appears to use CID type signalling, or at least on a line that otherwise has no CID on it, CID is sent, but with a standard modem I can only receive the date, time, and phone number (eg normal CID info). After that the phone rings, but Telstra will just call the number and use 'Text to Speech' to read the message out when a user answers. Does anyone know anything more about this in Australia or, failing that, if they do the same thing anywhere else in the world? My guess is that either: 1. the whole message is transmitted in the CID period, but my modem doesn't hear it, but then I don't know how Telstra would know that the message has been received. 2. Some indicative signalling takes place in the CID, which then triggers the handset to hide rings from the user and use normal modem signalling to transfer the message. If it has been around for a while outside Australia, is there an SMS module for Asterisk which would make use of it? I think that being able to receive (and probably send - haven't even started looking at that yet but it is supported in the same way) SMS messages would be a really nifty thing to be able to do from a phoneline, and would save me buying a $600 GSM modem to do the same thing! Thanks James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I need feed back on how an Aastra VentureIP 4FXO
Point taken. At $1300 per month it really isn't worth it. PaulH - Original Message - From: Tim Litwiller [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 17, 2006 4:41 PM Subject: Re: [Asterisk-Users] I need feed back on how an Aastra VentureIP 4FXO We'd love that - but our business is in Kansas - 6 miles from town - and a T1 is over $1300 per month - so it's not an option. 8 lines will be around $360 + long distance. Luckily we can get DSL because our phone company is using some Canadian technology that allows dsl to work up to 12 miles from the CO. Instead of 2 mile like Sprint does in the next county over where I live. [EMAIL PROTECTED] wrote: Point taken! Then an T1/E1 is the way to go. PaulH - Original Message - From: Tim Litwiller [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, January 16, 2006 9:18 AM Subject: Re: [Asterisk-Users] I need feed back on how an Aastra VentureIP 4FXO Well we have 3 sales people that are out visiting customers 50% or more of the time and it will get more as we got them laptops now. And if we can forward the calls to their cell phones with our phone system instead of giving the customers their cell numbers and then hanging up on the customer it will provide a better experience for the customer and better control for us. It may still be overkill but 4 lines aren't enough in the busy season and if we have 3 calling in and getting forwarded thru another to cells that is six already. And business is growing so we want room to expand before having to upgrade again. [EMAIL PROTECTED] wrote: 8 lines for 10 phones is overkillreally PaulH - Original Message - From: Tim Litwiller [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, January 15, 2006 2:38 PM Subject: Re: [Asterisk-Users] I need feed back on how an Aastra VentureIP 4FXO Thanks for the heads up - I didn't see anything that said it did work with asterisk so I thought I better ask. So if you where setting up a 6 - 8 telephone line system with 10 - 12 phones and trying to stay under $3000 for the system and phones what would you suggest. It sounds like if I can't do it for $3000 or under we will just stay with our old - outdated - partially functional phone system. I can probably reuse a workstation machine. And use AAH to make install and configuration easy. But that leaves some device ( suggest one to me) for * 8 fxo ports -And **12 voip phones * I think I'll just pass the fax/dsl line directly to the fax machine and dsl modem since we don't use it for anything else anyways and that means we don't have to worry about receiving faxes thru asterisk. ** I'd like to use the new Sipura 941's but may have to go with grandstream 2000's because of cost. We were supposed to have this done in November but cost issues have pushed it back this far already - so I'm not sure when this will happen. Cory Andrews wrote: The Aastra VentureIP system used a semi proprietary, non SIP protocol. I do not think it would integrate with Asterisk very well. Cory J Andrews VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 ++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] AIM - B2CORY - Original Message - From: Carlos Chavez [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, January 14, 2006 6:04 PM Subject: Re: [Asterisk-Users] I need feed back on how an Aastra VentureIP 4FXO On Sat, 14 Jan 2006 11:22:51 -0600, Tim Litwiller wrote works with Asterisk. I'm thinking I'd need 2 to support 6-8 lines - Or suggest some other equipment that will provide up to 8 fxo ports and connect to asterisk. for future projects I'd also like something with 2 fxo ports and 4 - 5 fxs ports - I suppose a digium card would do fine for 2 fxo and 2fxs and I could do a sipura 2002 for 2 more. I do not think that the Venture IP will work with Asterisk at all. As far as I know it is a self contained system. The gateway unit will autoconfigure the phones so they work together. The firmware for the phones is not the same as the one used for SIP and Asterisk. -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] List
The list is very quiet today - almost too quiet PaulH ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
