RE: [Asterisk-Users] 'Asterisk for Small Office Setup'
Surely you mean "grammar"? Sorry, I just had to point that out :) Personally, I'd take issue with the title -- if you need to do a "small-office" setup by-the-book, then chances are you're not resourceful enough to find the required information online -- and if you can't even find the basics online, once something doesn't go quite-as-planned, you're stuck up the creek without a paddle. I propose to write a five-page pamphlet based on the excellent "Getting Started With Asterisk" guide that simply lists what you can do with Asterisk; what you need in terms of hardware and SKILLs; then walks you through a basic setup "from scratch" and lists where you can find more information online. This would be a worthwhile pursuit once 1.0 is done. > -Original Message- > From: John Vogel [mailto:[EMAIL PROTECTED] > Sent: Saturday, July 24, 2004 9:15 AM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] 'Asterisk for Small Office Setup' > > 1. Poor spelling, punctuation, grammer. A flat, hard-to-read ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with T1 PRI Configuration
[EMAIL PROTECTED] wrote: 1. Dial Tone - No, Yes - Precise, Yes - SCC Not relevant on a PRI. 2. Framing - SF, ESF ESF 3. Line Coding - AMI, B8ZS B8ZS 4. Signaling Type - Ground Start, E&M, Loop Start w/Ring, Loop Start w/o Ring The signalling type is "PRI", none of the rest of these are relevant. 5. Pulse Mode - DTMF, MF Not relevant on a PRI. 6. Outpulse Start - Wink, Immediate, Seizure Not relevant on a PRI. 7. If Seizure then - Origination, Digit Collection. Not relevant on a PRI. I'm not sure of the correct answers to any of these. I do know that I want to be able to get caller ID and the number dialed for the application that I'm building. Most of these questions are for trunks being delivered over a T-1, not for PRI. If you are truly ordering PRI service (which is the more powerful of the two) then these questions don't apply :-) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help with T1 PRI Configuration
I'm ordering a PRI T1 for use with Asterisk and a Digium Wildcard TE405P. The provider is asking me a number of questions about how I want to configure the line. Here goes: 1. Dial Tone - No, Yes - Precise, Yes - SCC 2. Framing - SF, ESF 3. Line Coding - AMI, B8ZS 4. Signaling Type - Ground Start, E&M, Loop Start w/Ring, Loop Start w/o Ring 5. Pulse Mode - DTMF, MF 6. Outpulse Start - Wink, Immediate, Seizure 7. If Seizure then - Origination, Digit Collection. I'm not sure of the correct answers to any of these. I do know that I want to be able to get caller ID and the number dialed for the application that I'm building. I think from other things I've read on the list that I want B8ZS and E&M with Wink. I'm not sure of those and I have no idea on the others. So, please help me out if you know. Thanks. -Scott
[Asterisk-Users] VoiceMail Group Broadcasting
Using the latest code from CVS. Has anyone figured out a way to setup any kind of Group or Broadcasting of Voicemail messages? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pseudo zap channel - how to get rid of it ?
Shahid wrote (on Jun 07):
> The result is, I cant use Zap/g1 in extensions, eg this doesnt work anymore:
Where in your channels section do you describe what group the channel is
in?
If it worked "before" then it probably shouldn't have, since there's no
group=X (X should be 1 in your example) in your channel configuration.
Chris
>
> exten => _BLAXXX,1,Dial(Zap/g1/${EXTEN})
>
> Instead I need to use:
>
> exten => _BLAXXX,1,Dial(Zap/1/${EXTEN})
>
> Here is my zaptel.conf
>
> fxsks=1
> loadzone = us
> defaultzone=us
>
> Here is my zapata.conf
> ---
>
> [channels]
> signalling=fxs_ks
> rxwink=300 ; Ausecallerid=yes
> hidecallerid=no
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> context=default
> channel => 1
>
>
>
>
>
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[Asterisk-Users] pseudo zap channel - how to get rid of it ?
Hello all,
Downloaded, compiled and installed Asterisk CVS-04/15/04-17:54:5. Everything
looks fine except I see a pseudo channel in the 'zap show channels'.
*CLI> zap show channels
Chan Extension Context Language MusicOnHold
pseudodefault
1default
The result is, I cant use Zap/g1 in extensions, eg this doesnt work anymore:
exten => _BLAXXX,1,Dial(Zap/g1/${EXTEN})
Instead I need to use:
exten => _BLAXXX,1,Dial(Zap/1/${EXTEN})
Here is my zaptel.conf
fxsks=1
loadzone = us
defaultzone=us
Here is my zapata.conf
---
[channels]
signalling=fxs_ks
rxwink=300 ; Ausecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
context=default
channel => 1
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[Asterisk-Users] Layer 3 VPN Question
I am trying to hook up my Cisco telephones to Asterisk using a Layer 3 switch and am having difficulties it getting it to work. I realize this may not be the proper forum for a discussion on VLAN architecture and configuration so I won’t post the question here. I though I had read all the requisite information regarding the configuration for this, but perhaps I am missing something simple. Is there anyone who is knowledgeable in VLAN’s and networking that could offer some assistance? Thanks, Kevin
Re: [Asterisk-Users] sip ua---------asterisk-------h323gw
mohammad mirzaee wrote: HI ALL; I have a sip ua (ATA) registered in my asterisk box.I want my astersik box to route all incoming calls from sip ua (A-Z prefix) to h323 GW. what syntax should I use in extension.conf for routing all prefixes to h323 GW. Set the context of the sip user to a specific context. context=sipata Then create in extensions.conf a context like that with the s exten. [sipata] exten => s,1,Dial(the h323 channel name and extension) -- respectfully, Joseph ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multi companies
Troy Settle wrote: Sure, you can assign different contexts to different zap channels, but how does this help? Normally, the telco will send each call on the first available channel in a given trunk group, sometimes, they will come in on random channels. When a call rings in on Zap/1-1, the only way to know what to do with it, is by the DNIS information. exten => 2200,1,NoOp,Company A - main line exten => 2201,1,NoOp,Company A - Fax exten => 2211,1,noOp,Company A - CEO Direct Line exten => 3000,1,NoOp,Company B - main line exten => 3001,1,NoOp,Company B - Fax exten => 3022,1,NoOp,Company B - Sales exten => 3023,1,NoOp,Company B - Customer Service Any of these calls might come in on any of your lines, so how does setting a different context for different zap channels help? Does each company have a separate dnis? exten => 2200,1,Goto(company1,s,1) exten => 2201,1,Goto(company2,s,1) [company1] ;Do stuff exten => s,1,Answer exten => s,2,Background(Welcome-msg) include => this-company-phones [company2] ;Do stuff exten => s,1,Answer exten => s,2,Background(Welcome-msg) include => this-company-phones If you know what channel pertains to who, than you coould use the zap contexts to control them. respectfully, Joseph ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Multi companies
Sure, you can assign different contexts to different zap channels, but how does this help? Normally, the telco will send each call on the first available channel in a given trunk group, sometimes, they will come in on random channels. When a call rings in on Zap/1-1, the only way to know what to do with it, is by the DNIS information. exten => 2200,1,NoOp,Company A - main line exten => 2201,1,NoOp,Company A - Fax exten => 2211,1,noOp,Company A - CEO Direct Line exten => 3000,1,NoOp,Company B - main line exten => 3001,1,NoOp,Company B - Fax exten => 3022,1,NoOp,Company B - Sales exten => 3023,1,NoOp,Company B - Customer Service Any of these calls might come in on any of your lines, so how does setting a different context for different zap channels help? -- Troy Settle Pulaski Networks http://www.psknet.com 866.477.5638 > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Chris Shaw > Sent: Friday, July 23, 2004 7:15 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Multi companies > > Can't you just assign a context to a Zap channel or Group of > Zap channels in > zapata.conf just like you do with SIP? (e.g. > Context=company1) If so, you > don't need any of that, just create separate IVR contexts for > each company > and assign those contexts to specific Zap channels or channel > groups you > want... > > I might be wrong but that would seem to be the logical way to do it... > > > - Original Message - > From: "Joshua McClintock" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Friday, July 23, 2004 3:50 PM > Subject: Re: [Asterisk-Users] Multi companies > > > > You don't need the _ on the front of those extensions since those > > particular examples aren't patterns. My mistake. > > > > Check out www.voip-info.org for MANY good examples. > > > > On Fri, 2004-07-23 at 15:21, Joshua McClintock wrote: > > > Depending on the context that your 'incoming' lines are > on, you can do > > > something like this: > > > > > > [incoming-lines] > > > exten => _1235551212,Macro(autoatt-company1) > > > exten => _1235551213,Macro(autoatt-company2) > > > > > > > > > [macro-autoatt-company1] > > > Do some junk, dial some peeps > > > > > > [macro-autoatt-company2] > > > Do some junk, dial some peeps > > > > > > > > > > > > On Fri, 2004-07-23 at 14:57, Martin Keding wrote: > > > > I am fairly new to Asterisk and I want to do some testing with > > > > multi-companies on the same box. I have two inbound lines and I > basically > > > > want one to trigger auto-att. for company 1, the other > line to trigger > > > > auto-attend for company 2. Could somebody point me to a > sample conf. > or > > > > documentation. > > > > > > > > Thanks > > > > Martin > > > > > > > > ___ > > > > Asterisk-Users mailing list > > > > [EMAIL PROTECTED] > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > To UNSUBSCRIBE or update options visit: > > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > To UNSUBSCRIBE or update options visit: > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Autologout of dynamic agents
Has anyone had any luck with dynamic agents in queues (addqueuemember) and autologoff? I have searched, but so far come up empty for a Sip related example.
RE: [Asterisk-Users] Please help I fear I have missed something very important! but what?
I had already tried that, however my register statement already specifies to
ring out on ext 1001 so the call isn't an unqualified one so should not look
up the s extension
register => 2:[EMAIL PROTECTED]/1001
However I put the s extension statements in and the results were the same
with or without the /1001 on the register statement.
When I run the SIP DEBUG command from the console, I can see the CLI of the
phone I am calling from in the sip statements however the asterisk box just
sends out an engaged tone.
A bit of background to the box, it is running a full install of Redhat 9
without the firewall enabled. (I did a full install as I thought it would be
good to learn about Linux as well)
It is not behind a NAT and configured with a static Public IP.
Does anyone know of problems that can occur with a full install of Redhat 9?
As I have read somewhere that someone recommends Redhat 8 and not 9 but he
wasn't specific as to why, does anyone know of any problems (or is it a wild
goose chase to read into it too much)
If anyone else has any ideas I would be much appreciated. Elman Thankyou
very much for taking the time out to respond.
Would any of you suggest blatting the PC and starting again with a basic
bare minimum config, i.e. no GUI
Regards
Stuart Buchanan
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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Elman Efendiyev
Sent: 24 July 2004 19:53
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Please help I fear I have missed something
very important! but what?
Looks like you missed 's' extension for incoming calls
You need something like this in extensions.conf
exten => s,1,Answer
exten => s,2,Dial(SIP/1001,20,t)
See sample of extensions.conf in asterisk distribution (make samples if
you didn't install samples)
--
Sincerely,
Elman Efendiyev
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sales
Sent: Saturday, July 24, 2004 9:03 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Please help I fear I have missed something
very important! but what?
Sorry about this, I have been struggling with the basics of my asterisk
config.
I set up two sip peers and two phones. And I set up lots of dial masks
for outgoing calls, all my outgoing calls were working great, however
incoming calls were a different matter altogether, I cannot get incoming
calls to work. So I have gone back to a very basic FWD config, with one
phone which as far as I am aware should work, but doesn't. I cannot find
info on how to fix this.
Below is my sip.conf
[general]
port = 5060
bindaddr = xxx.xxx.xxx.xxx
context = sip
register => 2:[EMAIL PROTECTED]/1001
[fwd]
type=friend
secret=xx
username=xx
host=fwd.pulver.com
;
;
[1001]
type=friend
username=xx
host=dynamic
secret=xxx
callerid=Home <1001>
dtmfmode=RFC2833
mailbox=1001
context=sip
and here is my extensions.conf:
[general]
static=yes
writeprotect=no
;
[globals]
HOME=SIP/1001
;
[sip]
exten => 1001,2,Dial(SIP/1001,20,t)
include => fwdnet
;
[fwdnet]
exten => _8.,1,Dial,SIP/${EXTEN:[EMAIL PROTECTED],t
Now as I said I can call out no probs by dialing 8 then the FWD number,
but incoming calls don't work, and as far as I can see that should ring
ext 1001 for 20 secs.
Could someone please help a complete Linux/Asterisk Newb, as apart from
this I have learnt a hell of a lot. But it's the last thing I need to
solve.
The linux box for this testing has a unfirewalled public IP address, so
there is no problems with NAT
Please please can someone help. If I have missed something important
then I aplogise, as I have been scouring the wiki and the archives to no
avail
Regards
Stuart Buchanan
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RE: [Asterisk-Users] Do not buy "Asterisk for Small Office Setup"!
You bet it's for sale! Make me an offer! BTW I have the other Asterisk book on order too - I'm keeping my fingers crossed it will be a good one. I'd really like to hand out a good book to each of my customers. Makes my life easier. And good luck with Asterisk. It's a lot of fun. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Remco Barende Sent: Saturday, July 24, 2004 11:41 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Do not buy "Asterisk for Small Office Setup"! But isn't this due to the 'intended audience' for the book? I just started setting up asterisk, I can dial the test message now (yea! :) ) but I have still to find out how to write an extensions.conf file. So far I found the book on asterisk.org easier to read than just browsing thru the wikis but a book for people starting from scratch could be useful too Are you putting it up for sale? :) Thanks! Remco On Sat, 24 Jul 2004, John Vogel wrote: > > See my previous email but the book is worse than I thought. In > addition, to the things I mentioned earlier. > > 1. The table of contents is not a table of contents. The only chapter > heading is "Chapter 1". It is impossible to tell what is in what chapter. > > 2. The index is not alphabetized! This renders it useless. Any decent > word processor will create a better index with a push of the button. > > 3. I'm trying to set up fax. Does the book discuss it? No way to tell > except leafing through every page. > > Summary - use the wiki, Luke. A Googol search of voip-info is faster > and provides much more complete, well written information. > > Well, I said in my previous email that a fair price for this book is > $14.95 not $40. I was wrong. This book does the entire Asterisk > community a disservice and should not be purchased at any price. Do > not encourage/fund the "author" to produce more such rip offs. > > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip ua---------asterisk-------h323gw
HI ALL; I have a sip ua (ATA) registered in my asterisk box.I want my astersik box to route all incoming calls from sip ua (A-Z prefix) to h323 GW. what syntax should I use in extension.conf for routing all prefixes to h323 GW. Regards mohammad
Re: [Asterisk-Users] Do not buy "Asterisk for Small Office Setup"!
On Sat, 24 Jul 2004 20:41:00 +0200 (CEST), Remco Barende <[EMAIL PROTECTED]> wrote: > But isn't this due to the 'intended audience' for the book? > > I just started setting up asterisk, I can dial the test message now (yea! :) > ) but I have still to find out how to write an extensions.conf file. This is entirely off-topic (well, maybe not), but since you were looking for extensions.conf information, check out chapter 4 of the http://www.asteriskdocs.org project. If you find anything that doesn't seem to make sense (or is wrong) please feel free to inform the asterisk-doc mailing list and we'll change it. The wiki is also an EXCELLENT resource. Thanks, Leif Madsen http://www.asteriskdocs.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Please help I fear I have missed something very important! but what?
Looks like you missed 's' extension for incoming calls
You need something like this in extensions.conf
exten => s,1,Answer
exten => s,2,Dial(SIP/1001,20,t)
See sample of extensions.conf in asterisk distribution (make samples if
you didn't install samples)
--
Sincerely,
Elman Efendiyev
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sales
Sent: Saturday, July 24, 2004 9:03 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Please help I fear I have missed something
very important! but what?
Sorry about this, I have been struggling with the basics of my asterisk
config.
I set up two sip peers and two phones. And I set up lots of dial masks
for outgoing calls, all my outgoing calls were working great, however
incoming calls were a different matter altogether, I cannot get incoming
calls to work. So I have gone back to a very basic FWD config, with one
phone which as far as I am aware should work, but doesn't. I cannot find
info on how to fix this.
Below is my sip.conf
[general]
port = 5060
bindaddr = xxx.xxx.xxx.xxx
context = sip
register => 2:[EMAIL PROTECTED]/1001
[fwd]
type=friend
secret=xx
username=xx
host=fwd.pulver.com
;
;
[1001]
type=friend
username=xx
host=dynamic
secret=xxx
callerid=Home <1001>
dtmfmode=RFC2833
mailbox=1001
context=sip
and here is my extensions.conf:
[general]
static=yes
writeprotect=no
;
[globals]
HOME=SIP/1001
;
[sip]
exten => 1001,2,Dial(SIP/1001,20,t)
include => fwdnet
;
[fwdnet]
exten => _8.,1,Dial,SIP/${EXTEN:[EMAIL PROTECTED],t
Now as I said I can call out no probs by dialing 8 then the FWD number,
but incoming calls don't work, and as far as I can see that should ring
ext 1001 for 20 secs.
Could someone please help a complete Linux/Asterisk Newb, as apart from
this I have learnt a hell of a lot. But it's the last thing I need to
solve.
The linux box for this testing has a unfirewalled public IP address, so
there is no problems with NAT
Please please can someone help. If I have missed something important
then I aplogise, as I have been scouring the wiki and the archives to no
avail
Regards
Stuart Buchanan
--
This email is only for the intended recipient. If you have received this
email
in error please notify the sender and delete the message immediately.
This email has been checked for viruses to ensure that any attachments
are free
from viruses. You should, however, carry out your own virus check before
opening any attachment. We accept no liability for loss or damage caused
by
software viruses.
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Re: [Asterisk-Users] Do not buy "Asterisk for Small Office Setup"!
But isn't this due to the 'intended audience' for the book? I just started setting up asterisk, I can dial the test message now (yea! :) ) but I have still to find out how to write an extensions.conf file. So far I found the book on asterisk.org easier to read than just browsing thru the wikis but a book for people starting from scratch could be useful too Are you putting it up for sale? :) Thanks! Remco On Sat, 24 Jul 2004, John Vogel wrote: > > See my previous email but the book is worse than I thought. In addition, to > the things I mentioned earlier. > > 1. The table of contents is not a table of contents. The only chapter > heading is "Chapter 1". It is impossible to tell what is in what chapter. > > 2. The index is not alphabetized! This renders it useless. Any decent word > processor will create a better index with a push of the button. > > 3. I'm trying to set up fax. Does the book discuss it? No way to tell except > leafing through every page. > > Summary - use the wiki, Luke. A Googol search of voip-info is faster and > provides much more complete, well written information. > > Well, I said in my previous email that a fair price for this book is $14.95 > not $40. I was wrong. This book does the entire Asterisk community a > disservice and should not be purchased at any price. Do not encourage/fund > the "author" to produce more such rip offs. > > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Do not buy "Asterisk for Small Office Setup"!
Title: Do not buy "Asterisk for Small Office Setup"! See my previous email but the book is worse than I thought. In addition, to the things I mentioned earlier. 1. The table of contents is not a table of contents. The only chapter heading is "Chapter 1". It is impossible to tell what is in what chapter. 2. The index is not alphabetized! This renders it useless. Any decent word processor will create a better index with a push of the button. 3. I'm trying to set up fax. Does the book discuss it? No way to tell except leafing through every page. Summary - use the wiki, Luke. A Googol search of voip-info is faster and provides much more complete, well written information. Well, I said in my previous email that a fair price for this book is $14.95 not $40. I was wrong. This book does the entire Asterisk community a disservice and should not be purchased at any price. Do not encourage/fund the "author" to produce more such rip offs.
[Asterisk-Users] Please help I fear I have missed something very important! but what?
Sorry about this, I have been struggling with the basics of my
asterisk config.
I set up two sip peers and two phones. And I set up lots of dial
masks for outgoing calls, all my outgoing calls were working great, however
incoming calls were a different matter altogether, I cannot get incoming calls
to work. So I have gone back to a very basic FWD config, with one phone which
as far as I am aware should work, but doesn’t. I cannot find info on how
to fix this.
Below is my sip.conf
[general]
port = 5060
bindaddr = xxx.xxx.xxx.xxx
context = sip
register => 2:[EMAIL PROTECTED]/1001
[fwd]
type=friend
secret=xx
username=xx
host=fwd.pulver.com
;
;
[1001]
type=friend
username=xx
host=dynamic
secret=xxx
callerid=Home
<1001>
dtmfmode=RFC2833
mailbox=1001
context=sip
and here is my
extensions.conf:
[general]
static=yes
writeprotect=no
;
[globals]
HOME=SIP/1001
;
[sip]
exten =>
1001,2,Dial(SIP/1001,20,t)
include =>
fwdnet
;
[fwdnet]
exten =>
_8.,1,Dial,SIP/${EXTEN:[EMAIL PROTECTED],t
Now as I said I can call out no probs by dialing 8 then the FWD
number, but incoming calls don’t work, and as far as I can see that should
ring ext 1001 for 20 secs.
Could someone please help a complete Linux/Asterisk Newb, as
apart from this I have learnt a hell of a lot. But it’s
the last thing I need to solve.
The linux box for this testing has a
unfirewalled public IP address, so there is no problems with NAT
Please please can someone help. If I have missed something important then I aplogise, as I have been scouring the wiki
and the archives to no avail
Regards
Stuart Buchanan
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RE: [Asterisk-Users] Play CD!
MP3s have to use constant bitrate not variable bit rate. Look in the documentation for mpg123. -Original Message- From: Jozeph Brasil [mailto:[EMAIL PROTECTED] Sent: Saturday, July 24, 2004 5:30 AM To: [EMAIL PROTECTED] Subject: RES: [Asterisk-Users] Play CD! I do that. But when I play MusicOnHold the music is played slowly! I don´t know why... but is how the bitrate is playing with a different number. -Mensagem original- De: Chris Foster [mailto:[EMAIL PROTECTED] Enviada em: sábado, 24 de julho de 2004 01:37 Para: [EMAIL PROTECTED] Assunto: Re: [Asterisk-Users] Play CD! On Fri, 23 Jul 2004 23:29:19 -0300, Jozeph Brasil <[EMAIL PROTECTED]> wrote: > Hi all, > > Is it possible to play a CD has MusicOnHold? > > Thanks, > Jozeph > Why don't you just rip the CD to MP3? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hack to make * -> (H323) -> CCM -> IOS GW work
Jeremy McNamara wrote (on Jul 24): > Chris Luke wrote: > >The chan_h323 in CVS and chan_oh323 both exhibited the same behaviour. > > > I have setup chan_h323 to talk to CCM without any trouble, after someone > informed me we had to override the External RTP object, which is part of > cvs -head now. I highly doubt the obsolete -stable has it. I'm running HEAD, dilligently updating at least once a day. It still requires your overrided RTP object since the GW RTP endpoint is on a different address. It works with CCM without my hack, provided I don't try to place a call that routes via the GW behind CCM. It could well be a tweakable on CCM too, but I only have so much access to it and didn't find such a thing. Basically, without this hack, or if the call is not answered ms after it begins to ring, the CCM never ever sends me an openLogicalChannelAck, which means I never get told to send my RTP to the GW. We send the openLogicalChannel message - it doesn't get answered. And in any case, there's no point the CCM/GW sending me ringing audio, since rtp.c will ignore it until there's a far end RTP address to reciprocate to - and which h323 doesn't ask for until after it's answered. Chris. -- == [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need to block incoming collect calls
All right Steve. I'll ask them.. But if anybody knows that, please post an answer to the list. This is a very important Asterisk security configuration to avoid people call you without having to pay the call.. thank you Oz From: "Steve Totaro" <[EMAIL PROTECTED]> Reply-To: [EMAIL PROTECTED] To: <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] Need to block incoming collect calls Date: Sat, 24 Jul 2004 11:57:05 -0400 I dont know about blocking in * but you should be able give the telco a call and tell them no collect calls. - Original Message - From: "Osvaldo Mundim Junior" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Saturday, July 24, 2004 10:06 AM Subject: [Asterisk-Users] Need to block incoming collect calls > Hi everybody,, > > I need to block incoming collect calls to my Asterisk box but I could not > find out where to do that. > > Went to zaptel.h but I did not see any timing which can be applied to > collect calls. Does anybody knows if I can set this up in Asterisk? > > I'm using an E100P connected to the PSTN and a T100P connected to a Zhone > 100. Version: > Asterisk CVS-05/30/04-16:28:04 > > thank you > Oz > > _ > MSN Messenger: instale grátis e converse com seus amigos. > http://messenger.msn.com.br > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ MSN Messenger: instale grátis e converse com seus amigos. http://messenger.msn.com.br ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Attendant configured AutoAttendant
The recording part is easy. ; exten for recording greetings/menus exten => 12,1,Wait(2) exten => 12,2,Record(/var/lib/asterisk/sounds/maingreeting:gsm) exten => 12,3,Wait(2) exten => 12,4,Playback(/var/lib/asterisk/sounds/swelcome) exten => 12,5,Wait(2) exten => 12,6,Hangup add authenticate to prevent accidental recording - Original Message - From: "Frank" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Saturday, July 24, 2004 10:25 AM Subject: [Asterisk-Users] Attendant configured AutoAttendant > Anyone have a user configured auto attendant setup? Something that can > be used without the * admin helping to make changes. > > Something where the operator can record the message like 'press 1 for > john, 2 for bill, 3 for jean' and then the operator can enter the > extension that gets dialed when the caller presses 1 or 2 or 3? > > This would be useful if Bill leaves the company, the operator can change > the message and put in a different person at 'position 2' and then > change the extension that gets dialed when caller presses 2. all this > without involving the * admin. > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need to block incoming collect calls
I dont know about blocking in * but you should be able give the telco a call and tell them no collect calls. - Original Message - From: "Osvaldo Mundim Junior" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Saturday, July 24, 2004 10:06 AM Subject: [Asterisk-Users] Need to block incoming collect calls > Hi everybody,, > > I need to block incoming collect calls to my Asterisk box but I could not > find out where to do that. > > Went to zaptel.h but I did not see any timing which can be applied to > collect calls. Does anybody knows if I can set this up in Asterisk? > > I'm using an E100P connected to the PSTN and a T100P connected to a Zhone > 100. Version: > Asterisk CVS-05/30/04-16:28:04 > > thank you > Oz > > _ > MSN Messenger: instale grátis e converse com seus amigos. > http://messenger.msn.com.br > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 to SIP Server Load
> Steve Totaro wrote: > > > Does anyone do any large scale SIP to H323 conversion? How many > > simultaneous calls can your server handle and on what hardware? I think > > I read on the wiki that twenty five would max out most servers. > > > The wiki is very wrong then. > > > Jeremy McNamara > That is what I figured. Care to share some actual numbers? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 to SIP Server Load
Jeremy McNamara wrote: The wiki is very wrong then. At least regarding chan_h323. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hack to make * -> (H323) -> CCM -> IOS GW work
Chris Luke wrote: The chan_h323 in CVS and chan_oh323 both exhibited the same behaviour. I have setup chan_h323 to talk to CCM without any trouble, after someone informed me we had to override the External RTP object, which is part of cvs -head now. I highly doubt the obsolete -stable has it. Try again. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 to SIP Server Load
Steve Totaro wrote: Does anyone do any large scale SIP to H323 conversion? How many simultaneous calls can your server handle and on what hardware? I think I read on the wiki that twenty five would max out most servers. The wiki is very wrong then. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hack to make * -> (H323) -> CCM -> IOS GW work
The hack below is for OpenH323, not Asterisk. This is not an Asterisk problem AFAICT. I am posting it here so that any other Asterisk user with a similar problem might benefit from it. I may or may not post it to an OpenH323 list, but since both variants of the H.323 channel in Asterisk use non-current OpenH323 versions, it may not be of any benefit to anyone anytime soon if I went that route! I've not checked newer OpenH323 source to see if the hack below can be shifted into the application either. I've been messing with the companys CCM PBX, which has an IOS based gateway box with a PRI behind it, and talking to it from my * box with H.323. For the most part it works. It's in Lonodn UK, I'm (currently) in Boston, MA, USA. It's all VPNed. People being able to call me has made working remotely that much easier. However, placing outgoing calls that went via the GW had issues - if they weren't answered within a few milliseconds of ringing, then the call would fail - sometimes you'd get one way audio, sometimes none at all, and the GW or the CCM always sent a RELEASE within a few seconds, even if not answered. If you answered the call quckly, within ms of ringing, the call worked fine. Any other call (ie, that didn't use the PSTN gateway for outgoing calls) would work fine too, which was the most baffling. Even incoming via the PSTN gateway were fine. The chan_h323 in CVS and chan_oh323 both exhibited the same behaviour. I'd come accross a number of message in Mantis, this lists archives and by googling in general suggesting that "H.323 via CCM" "Cisco GWs" and other combinations of Cisco and H.323 don't work, have less functionality, need faststart, needs faststart disabled, needs other things doing to it and generally won't be fixed without money, etc etc. So I spent some time pouring over traces, and noticed one difference between calls that were answered unreasonably quickly and those not. After the call has been setup and the H.323 neighbors have exchanged their capabilities, a few tens of ms pass, and then the CCM sends an openlogicalchannel, so it can pass the audio of the ringing. However, when answered, the CCM/GW doesn't send back any open message indicating its RTP address/port. If you answer it early - the channel messages work as expected in both directions, and thus it works. So I hunted down an option to make the CCM not do this. And (eventually) came across mediaWaitForConnect in the OpenH323 source. It's not something you can change easily from client applications that I can see, but doing this: --- h323.cxx.old2004-07-23 16:04:45.109780688 -0400 +++ h323.cxx2004-07-23 16:04:49.577950415 -0400 @@ -2797,6 +2797,8 @@ if (hasVideoOrData) setupPDU.GetQ931().SetBearerCapabilities(Q931::TransferUnrestrictedDigital, 6); + setup.m_mediaWaitForConnect = TRUE; + if (!OnSendSignalSetup(setupPDU)) return EndedByNoAccept; and rebuilding the OpenH323 libraries made my problem "go away". The CCM doesn't insist on trying to send me any audio until the call is answered, and when answered the audio streams in both directions get setup as expected. Hopefully someone here will point out an easy way to set this option from ast_h323.cpp or something - while my C voodoo is strong, C++ is just foo to me. YMMV. Cheers, Chris. -- == [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Attendant configured AutoAttendant
Anyone have a user configured auto attendant setup? Something that can be used without the * admin helping to make changes. Something where the operator can record the message like 'press 1 for john, 2 for bill, 3 for jean' and then the operator can enter the extension that gets dialed when the caller presses 1 or 2 or 3? This would be useful if Bill leaves the company, the operator can change the message and put in a different person at 'position 2' and then change the extension that gets dialed when caller presses 2. all this without involving the * admin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PBX functions and different channels grouping
Hi All,
I need to replace old analog PBX with Asteriskl and X-Lise SIP
SoftPhones as client phones.
First: I have problems with implementation of PBX functions. I need and
unsuccesfully tried theese functions (took info at
http://voip-info.org/wiki-Asterisk+PBX+functions)
Call Pickup: Supported in the standard installation (*8 - defined in
res_parking.c +54)
- Just don't understand how to define pickup groups
Unattended Transfer (or "blind transfer"): Implemented in Asterisk (#),
optionally also in the phone
- when I press # on X-Lite it hangs up
Attended transfer: Implemented in Asterisk (FLASH)
- How nj make FLASH on X-Lite?
Call Pickup: Supported in the standard installation
- How to use it?
Automatic Call Distribution: ACD
- is it possible at all?
Also I tried Transfer command like this:
exten => 9,1,Transfer(SIP/234)
And when I press 9 got thiss error:
-- Executing Transfer("SIP/233-b6ad", "SIP/234") in new stack
Jul 24 16:34:09 WARNING[213006]: chan_sip.c:673 retrans_pkt: Maximum
retries exceeded on call
[EMAIL PROTECTED] for seqno 102
(Non-critical Request)
Where 192.168.1.1 IP of SIP client calling from (233 phone)
Second: Is it possible to group different channels? I need thing like
this:
Try to call ZAP channel, if it's busy (or another problemm with it) then
try to call H323 channel, if busy again try to call IAX2 channel
I found info only for ZAP channel grouping and dialing channels
simultaneously
Please help me with theese problemms
Thanks!
--
Sincerely,
Elman Efendiyev
[EMAIL PROTECTED]
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[Asterisk-Users] Should configured devices show up with "show channels"
At the CLI, should the command "show channels" list the configured devices that may be used for calls? Or, is that command simply a way of viewing the channels on a T1/E1 if you have that type of interface? If the latter, is there an equiv. command to list the single FXO interfaces? If the answer is yes, all configured devices should have something in channels, how do you map a linejack from /dev/phone0 to channel 1? I can see in the zapata.conf file where channels may be set, but not in phone.conf. I'm trying to get * to answer inbound calls. * reports seeing the caller ID for the incoming call, but never seems to try and actually answer the call. My "show channels" is also empty, leading me to believe that the phone module is loaded and generating events, but the component or configuration that should map from the phone0 device to a channel for * is missing, so nothing is happening with the events coming from the linejack. Does that sound plausible? Thanks, Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 'Asterisk for Small Office Setup'
I'm willing to take it on faith that everything in the book works. However, that is not the only (or even the most important) criteria for a technical book. A short list of deficiencies includes: 1. Poor spelling, punctuation, grammer. A flat, hard-to-read table of contents. These characteristics and a lack of good use of white space and different font types make the content less accessible. For people that already know the material, the content is understandable but then they don't need the book, do they? 2. Covers use of a single X100P card. That is not my definition of a small office (which, BTW, is not defined in the book). I have "small" customers, e.g. 6 employees, using a fractional T1. And what about a company with two incoming phones lines? 2. Content/$: A fair price for this book would be $14.95 3. White space: there are so many pages that are blank, or nearly so. What's the point? It looks like padding for page count (200). The material could be covered in 100 pages. 4. Misrepresenting the market: p. 182, "Recommended Consultants" lists exactly one, www.saww.net, the authors of the book. Etc., etc. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, July 23, 2004 12:36 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] 'Asterisk for Small Office Setup' please do tell what the problem is with the book, all the content with in the book works and has been tested, by myself and other no the book is not designed for the more advance users of asterisk but the beginners, which one could be figure out by the title "Asterisk For Small Office Setup" > > Any more information than that? I have a copy here as well but haven't > had time to read through it. > > P.S. Yes I know my name is mentioned in the book. No need to flame me > on that fact. I am a regular consumer like anyone. Author felt > inclined to put it in there. > > > - Original Message - > From: John Vogel <[EMAIL PROTECTED]> > Date: Fri, 23 Jul 2004 22:44:35 -0700 > Subject: [Asterisk-Users] "Asterisk for Small Office Setup" > To: [EMAIL PROTECTED] > > Don't buy this book for its content. It's a waste of $40. However, it > is useful to wave in front of my customer's faces to show them that > Asterisk is real. > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Need to block incoming collect calls
Hi everybody,, I need to block incoming collect calls to my Asterisk box but I could not find out where to do that. Went to zaptel.h but I did not see any timing which can be applied to collect calls. Does anybody knows if I can set this up in Asterisk? I'm using an E100P connected to the PSTN and a T100P connected to a Zhone 100. Version: Asterisk CVS-05/30/04-16:28:04 thank you Oz _ MSN Messenger: instale grátis e converse com seus amigos. http://messenger.msn.com.br ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Large Enterprises using asterisk
I want to clarify how this works now on *. First, on a legacy PBX, like a merlin legend, I have phones that have shared call appearances so that my assistant can answer my calls or see that I am on the phone, or so that I can have one phone on my desk and one at my conference table. This means that if (312) 221-1212 appears on 3 phones and is in use, all will indicate that. Can any hard phones (like Cisco 7960's) be configured that way? /carmi On Jul 23, 2004, at 7:43 AM, Robinson Tim-W10277 wrote: It is the hard phones that need this before Asterisk is a salable solution to small/medium businesses. What sells the system is the phones and the flashing lights. As most users already have a legacy system with a real BLF etc, until Asterisk has hard phones that have all those features it will be a tricky sale. Rgds Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christian Hoffmeyer Sent: 23 July 2004 15:00 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Large Enterprises using asterisk PS: If already existing soft (and/or hard) phones have more of this functionality - please let me know. WAMi and other gui interfaces already support this. http://www.voip-info.org/wiki-Asterisk+WAMI We are starting work on WAMi 2.0, and I am trying to make the source available for everyone as quickly as possible. It's not a matter of the source for WAMi being open. Rather, it's just a matter of having the time to make the WAMi source code available and having the structure setup to support bugs, maintenance, and contributions. J.Christian Hoffmeyer Asterisk Solutions Group, Inc. Huntsville, AL (o)256.705.0265 (c)256.655.0321 (fax) 256.705.0280 (tf)877.ASGI.4.ME (iax) 700.ASGI.4.ME Ask me about Asterisk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Question when using a Cisco as a PSTN GW
HI all I have a little question, and since there is a alot of Cisco Gurus somebody might be able to help me. I think It is an easy problem. My PSTN proviver strips the first digit in the callerid on all incoming calls. So when the call reaches my Asterisk I am missing a "0" in the CLID I guess it should be easy to prepend a digit on all incoming calls on a Cisco 5350 ? But I am unsure how a translationrule for that would look like. Anybody ? /Regards Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] yes shady dial running now but not dialling
hi there was wondering if anybody knows this.. have successfully installed shady dial and the agent is now logging in successfully i've enabled postgres debugging and i found out that no request had been made by the shady dialer to query the database for the numbers... also i am able to login to the queue simply by entering the agent id... it doesnt ask for the password...it simply plays the agent login ok message and then does nothing...any ideas why this might be happening?? also am not sure what the technology and group is?? they are both in the global variables... we are using sip soft phones to login as agent and then the calls are routed using iax2 protocol to another server. any help regarding this will be appreciated Do you Yahoo!? Yahoo! Mail Address AutoComplete - You start. We finish.
[Asterisk-Users] yes shady dial running now but not dialling
hi there was wondering if anybody knows this.. have successfully installed shady dial and the agent is now logging in successfully i've enabled postgres debugging and i found out that no request had been made by the shady dialer to query the database for the numbers... also i am able to login to the queue simply by entering the agent id... it doesnt ask for the password...it simply plays the agent login ok message and then does nothing...any ideas why this might be happening?? also am not sure what the technology and group is?? they are both in the global variables... we are using sip soft phones to login as agent and then the calls are routed using iax2 protocol to another server. Do you Yahoo!? Yahoo! Mail Address AutoComplete - You start. We finish.
[Asterisk-Users] [br] ---> indications.conf
Hello all, Me again! How to use [Br] on indications.conf file? When I set loadzone = br; defaultzone = br; on /etc/zaptel.conf I receive an error... Maybe I need to setup it from other file... Anyone can help? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM04B Dead?
On Jul 23, 2004, at 1:22 AM, Andres Junge wrote: What is a RMA? Return Merchandise/Materials(something like that) Authorization. It's a number from the mfr, that when the product arrives with it on the box, tells them to expect some dead hardware. rt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RES: [Asterisk-Users] Play CD!
I do that. But when I play MusicOnHold the music is played slowly! I don´t know why... but is how the bitrate is playing with a different number. -Mensagem original- De: Chris Foster [mailto:[EMAIL PROTECTED] Enviada em: sábado, 24 de julho de 2004 01:37 Para: [EMAIL PROTECTED] Assunto: Re: [Asterisk-Users] Play CD! On Fri, 23 Jul 2004 23:29:19 -0300, Jozeph Brasil <[EMAIL PROTECTED]> wrote: > Hi all, > > Is it possible to play a CD has MusicOnHold? > > Thanks, > Jozeph > Why don't you just rip the CD to MP3? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Documentation
Hello all, Anyone know where can I get a complete source that describe "all" options available in the configuration files? I like to know all available options in configuration files with a description and a correct syntax. Another think I would like to understand is what´s the real function of all files in the modules directory... I found someone in voip-info.org, but don´t have "all" files described. Thank you for any help! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Faild Echotest
> >
> > I tried that but I still get:
> > -- Executing Answer("SIP/2000-00b8", "") in new stack
> > -- Executing Echo("SIP/2000-00b8", "") in new stack
> > == Spawn extension (from-sip, 700, 2) exited non-zero on 'SIP/2000-00b8'
> > dev*CLI>
> >
Just tried from a Cisco 7960 with sip and it works fine:
-- Executing Playback("SIP/3000-43eb", "demo-echotest") in new stack
-- Playing 'demo-echotest' (language 'en')
-- Executing Echo("SIP/3000-43eb", "") in new stack
== Spawn extension (from-sip, 3910, 2) exited non-zero on 'SIP/3000-43eb'
running CVS-HEAD-07/12/04 with the following in extensions.conf:
; Create an extension for evaulating echo latency.
exten => 3910,1,Playback(demo-echotest) ; Let them know what's going on
exten => 3910,2,Echo ; Do the echo test
exten => 3910,3,Playback(demo-echodone) ; Let them know it's over
Rich
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RE: [Asterisk-Users] hang up when going to voicemail
Are you sure you have a mailbox for this number ?
Umar
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Matthew
Simpson
Sent: 23 July 2004 16:34
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] hang up when going to voicemail
I have a little menu set up where hitting 1, 2, or 3 places the call through
to a cellular phone over IAX. That works. However, if caller hits 4 to go
into voicemail, the system hangs up. Am I doing something wrong in the dial
plan, or is this a CVS change? I had no trouble with this until I upgraded
to about 07/21 CVS, and I'm on 07/23 [latest] now with same results.
My dial plan:
[txlink]
exten => s,1,Answer
exten => s,2,Background(/txlink/txlink-main)
exten => 1,1,Dial(IAX2/:[EMAIL PROTECTED]/12149490280)
exten => 1,2,Hangup
exten => 2,1,Dial(IAX2/:[EMAIL PROTECTED]/14693373687)
exten => 2,2,Hangup
exten => 3,1,Dial(IAX2/:[EMAIL PROTECTED]/18174017579)
exten => 3,2,Hangup
exten => 4,1,VoiceMail(s2147649296)
exten => 4,2,Hangup
exten => t,1,Goto(txlink,s,2)
exten => i,1,Playback(invalid)
[didin]
exten => 2147649296,1,Dial(SIP/2147649296,15)
exten => 2147649296,2,Goto(txlink,s,1)
exten => 2147649296,3,Hangup
Here is console output:
-- Executing Goto("SIP/2147649296-fb41", "txlink|s|1") in new stack
-- Goto (txlink,s,1)
-- Executing Answer("SIP/2147649296-fb41", "") in new stack
-- Executing BackGround("SIP/2147649296-fb41", "/txlink/txlink-main") in
new stack
-- Playing '/txlink/txlink-main' (language 'en')
== CDR updated on SIP/2147649296-fb41
-- Executing VoiceMail("SIP/2147649296-fb41", "s2147649296") in new
stack
-- Executing Hangup("SIP/2147649296-fb41", "") in new stack
== Spawn extension (txlink, 4, 2) exited non-zero on 'SIP/2147649296-fb41'
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Re: [Asterisk-Users] Priorizing of packets
I am using ISDN 64k + VOIP (iLBC, G729, GSM codecs only!) + bulk traffic (FTP, P2P, E-Mail, etc.). I am using tc, HTB 3.6 + finer tuned wshaper script. It works pretty well for me. The callee never misses any VOIP packet from my side. So I guess HTB + QOS works pretty well, even for VOIP. I use a VOIP only queue and all queues with fixed rates + correct ceil values. I never have tested CBQ yet so I can't tell you, if CBQ outperformes HTB on VOIP queuing. Blackvel - Original Message - From: "Dr. Rich Murphey" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Saturday, July 24, 2004 1:56 AM Subject: RE: [Asterisk-Users] Priorizing of packets > > [mailto:[EMAIL PROTECTED] On Behalf Of Chris Shaw > > > > - Original Message - > > From: "Dr. Rich Murphey" <[EMAIL PROTECTED]> > > To: <[EMAIL PROTECTED]> > > Sent: Friday, July 23, 2004 1:28 PM > > Subject: RE: [Asterisk-Users] Priorizing of packets > > > > > > > Can HTB minimize latency better than CBQ? > > > > > > Just curious, > > > Rich > > > > Hmm I've had problems using HTB, it seems to make voice > > quality a lot worse than with a well tuned CBQ Maybe I'm > > just using it wrong? > > > > -Chris > > Yea, lag is the only issue I've observed with CBQ, and most papers point to > solutions that involve management (lowering) of MTU of the non-voip packets. > There doesn't seem to be any open source for that yet. > > Cheers, > Rich > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 to SIP Server Load
- Original Message - From: Steve Totaro To: [EMAIL PROTECTED] Sent: Saturday, July 24, 2004 6:31 AM Subject: [Asterisk-Users] h323 to SIP Server Load Does anyone do any large scale SIP to H323 conversion? How many simultaneous calls can your server handle and on what hardware? I think I read on the wiki that twenty five would max out most servers.
Re: [Asterisk-Users] 'Asterisk for Small Office Setup'
i usually demo a system to show that it is real - Original Message - From: <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, July 23, 2004 3:35 PM Subject: [Asterisk-Users] 'Asterisk for Small Office Setup' > please do tell what the problem is with the book, all the content with in > the book works and has been tested, by myself and other no the book is not > designed for the more advance users of asterisk but the beginners, which > one could be figure out by the title "Asterisk For Small Office Setup" > > > > > Any more information than that? I have a copy here as well but haven't > > had time to read through it. > > > > P.S. Yes I know my name is mentioned in the book. No need to flame me > > on that fact. I am a regular consumer like anyone. Author felt > > inclined to put it in there. > > > > > > - Original Message - > > From: John Vogel <[EMAIL PROTECTED]> > > Date: Fri, 23 Jul 2004 22:44:35 -0700 > > Subject: [Asterisk-Users] "Asterisk for Small Office Setup" > > To: [EMAIL PROTECTED] > > > > Don't buy this book for its content. It's a waste of $40. However, it > > is useful to wave in front of my customer's faces to show them that > > Asterisk is real. > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] h323 to SIP Server Load
Does anyone do any large scale SIP to H323 conversion? How many simultaneous calls can your server handle and on what hardware? I think I read on the wiki that twenty five would max out most servers.
Re: [Asterisk-Users] Doublehash transfers
I think this will be coming in kapejod's bri-stuff in the next few days. Rgds Tim Dave Cotton wrote: On Fri, 2004-07-23 at 22:17 +0200, wrote: Yeah! Like having your dialplan "listening in" on the bridged call. Add some helper apps and we can program consultative transfers and much much more in a channel/device independent way! Channel/device independent consultative transfers. For me that would be the icing and cherry on the cake all at the same time! Is this a bounty candidate? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 'Asterisk for Small Office Setup'
please do tell what the problem is with the book, all the content with in the book works and has been tested, by myself and other no the book is not designed for the more advance users of asterisk but the beginners, which one could be figure out by the title "Asterisk For Small Office Setup" > > Any more information than that? I have a copy here as well but haven't > had time to read through it. > > P.S. Yes I know my name is mentioned in the book. No need to flame me > on that fact. I am a regular consumer like anyone. Author felt > inclined to put it in there. > > > - Original Message - > From: John Vogel <[EMAIL PROTECTED]> > Date: Fri, 23 Jul 2004 22:44:35 -0700 > Subject: [Asterisk-Users] "Asterisk for Small Office Setup" > To: [EMAIL PROTECTED] > > Don't buy this book for its content. It's a waste of $40. However, it > is useful to wave in front of my customer's faces to show them that > Asterisk is real. > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
