Re: [asterisk-users] Caller ID Names

[Todd R .]

To be sure you could setup a soft phone and see if the caller ID name comes in 
correctly.




> On Mar 10, 2015, at 8:41 AM, Jordan Cook - Gyron Networks 
>  wrote:
> 
> Hi,
>  
> In my dialplan I have the following line.
>  
> same => n,Set(CALLERID(name)=Support)
>  
> I am expecting this to always set the caller id name to ‘Support’  - however, 
> we are getting calls come in as “Anonymous” with the number as something like 
> “unknown@unknown”
>  
> We’re using Cisco 7945 phones – I possibly wonder if they are displaying this 
> rather than asterisk not changing it?
>  
> Anyone had similar experiences before?
> 
> 
> This message may be private and confidential. If you have received this 
> message in error, please notify us and remove it from your system.
> 
> Gyron may monitor email traffic data and the content of email for the 
> purposes of security and staff training.
> 
> Gyron Internet Ltd is a limited company registered in England and Wales. 
> Registered number: 4239332. Registered office: 3 Centro, Boundary Way, Hemel 
> Hempstead, HP2 7SU. VAT reg no 804 2532 63. Gyron is a registered trademark.
> 
> Gyron is a Deloitte Technology Fast 50 ranked company.
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Re: [asterisk-users] sip show channelstats reliable?

[Todd R .]

Thanks but no Adtran here.
I do think these stats are indicating an issue, I just don't know how to prove 
it outside Asterisk.

From: ewiel...@nyigc.com
To: tjrl...@live.com; asterisk-users@lists.digium.com
Date: Mon, 19 Jan 2015 13:55:33 -0500
Subject: RE: [asterisk-users] sip show channelstats reliable?

I’ve seen something similar with Adtran SIP gateways.When a re-invite 
happens the Adtran gets all confused about call stats and marks the 
pre-reinvite leg of the call as losing large numbers of packets.BTW, IIRC 
reinvites happen when a codec changes or the channel switches to T.38. Also 
Adtran SIP gateways appear not to support OPTIONS packets when running in SIP 
proxy mode, which is very annoying. At some point I’ll try and arrange a 
slugfest between Digium and Adtran and they can figure out why it doesn’t work. 
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Todd R.
Sent: Monday, January 19, 2015 1:45 PM
To: Asterisk-Users List
Subject: Re: [asterisk-users] sip show channelstats reliable? Additional info: 
At the moment I am running 1.8.x but the other day I was getting the same 
results on 11.x Here is a sample from show channelstats. I do think this 
command is showing that there is trouble between specific IP's and my Asterisk 
box but I don't know if the numbers are accurate and reliable. PeerCall 
IDDurationRecv: PackLost( %)JitterSend: 
PackLost(%)Jitterx.x.x.x5531341d06b00:07:42023123063836(73.41%)0.02310200(0.00%)0.0007
 Peer IP changed to protect the innocent :-) From: tjrl...@live.com
To: asterisk-users@lists.digium.com
Date: Mon, 19 Jan 2015 12:17:25 -0600
Subject: [asterisk-users] sip show channelstats reliable?I am seeing lots of 
lost packets when running the command sip show channelstats at the CLI. There 
are issues across multiple Asterisk servers I am trying to diagnose but 
everything I read seems to point to this command being pretty unreliable. Can I 
trust the info this command shows? I am showing lots of lost packets in sip 
show channelstats but I can't see any packet loss when pinging the same IP's 
to/from. Since I don't 100% control the network my gear is on, I need something 
outside of Asterisk to show the network engineer to convince here and myself 
that there are network issues. All I have is the loss that's shown from this 
command with no real network stats to back it up. Is there a magic command in 
CentOS anyone can recommend to diagnose and match up the issues shown in 
Asterisk using this command? Moving gear around on the network changes the info 
Asterisk shows a LOT. For example, if I point traffic to the main physical 
gateway I get loss to a particular customer's IP (their PBX), if I move it to 
another place on the network (as a VM) their IP is good and other customers 
IP's start showing loss using the channelstats info. Driving me freakin' crazy. 
It does appear there are network issues causing my troubles but I can't get 
help if I can't point to some hard and fast issues outside of Asterisk. The 
only thing I have right now is collissions showing on one of a few of our 
pfSense devices but they are virtual running on XenServer, still this would 
indicate a problem in my opinion. Thanks in advance for any assistance on this 
issue. Stepping back from the ledge now LOL  
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Re: [asterisk-users] sip show channelstats reliable?

[Todd R .]

Additional info:
At the moment I am running 1.8.x but the other day I was getting the same 
results on 11.x
Here is a sample from show channelstats. I do think this command is showing 
that there is trouble between specific IP's and my Asterisk box but I don't 
know if the numbers are accurate and reliable.

 
 
 
 
 
 
 
 
 
 
 
 
 
  Peer
  Call ID
  Duration
  Recv: Pack
  Lost
  ( %)
  Jitter
  Send: Pack
  Lost
  (
  %)
  Jitter
 
 
  x.x.x.x
  5531341d06b
  00:07:42
  023123
  063836
  (73.41%)
  0.
  023102
  00
  (
  0.00%)
  0.0007
 
Peer IP changed to protect the innocent :-)

From: tjrl...@live.com
To: asterisk-users@lists.digium.com
Date: Mon, 19 Jan 2015 12:17:25 -0600
Subject: [asterisk-users] sip show channelstats reliable?




I am seeing lots of lost packets when running the command sip show channelstats 
at the CLI.
There are issues across multiple Asterisk servers I am trying to diagnose but 
everything I read seems to point to this command being pretty unreliable.
Can I trust the info this command shows?
I am showing lots of lost packets in sip show channelstats but I can't see any 
packet loss when pinging the same IP's to/from.
Since I don't 100% control the network my gear is on, I need something outside 
of Asterisk to show the network engineer to convince here and myself that there 
are network issues.
All I have is the loss that's shown from this command with no real network 
stats to back it up.
Is there a magic command in CentOS anyone can recommend to diagnose and match 
up the issues shown in Asterisk using this command?
Moving gear around on the network changes the info Asterisk shows a LOT. For 
example, if I point traffic to the main physical gateway I get loss to a 
particular customer's IP (their PBX), if I move it to another place on the 
network (as a VM) their IP is good and other customers IP's start showing loss 
using the channelstats info.
Driving me freakin' crazy. It does appear there are network issues causing my 
troubles but I can't get help if I can't point to some hard and fast issues 
outside of Asterisk.
The only thing I have right now is collissions showing on one of a few of our 
pfSense devices but they are virtual running on XenServer, still this would 
indicate a problem in my opinion.
Thanks in advance for any assistance on this issue. Stepping back from the 
ledge now LOL

  

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[asterisk-users] sip show channelstats reliable?

[Todd R .]

I am seeing lots of lost packets when running the command sip show channelstats 
at the CLI.
There are issues across multiple Asterisk servers I am trying to diagnose but 
everything I read seems to point to this command being pretty unreliable.
Can I trust the info this command shows?
I am showing lots of lost packets in sip show channelstats but I can't see any 
packet loss when pinging the same IP's to/from.
Since I don't 100% control the network my gear is on, I need something outside 
of Asterisk to show the network engineer to convince here and myself that there 
are network issues.
All I have is the loss that's shown from this command with no real network 
stats to back it up.
Is there a magic command in CentOS anyone can recommend to diagnose and match 
up the issues shown in Asterisk using this command?
Moving gear around on the network changes the info Asterisk shows a LOT. For 
example, if I point traffic to the main physical gateway I get loss to a 
particular customer's IP (their PBX), if I move it to another place on the 
network (as a VM) their IP is good and other customers IP's start showing loss 
using the channelstats info.
Driving me freakin' crazy. It does appear there are network issues causing my 
troubles but I can't get help if I can't point to some hard and fast issues 
outside of Asterisk.
The only thing I have right now is collissions showing on one of a few of our 
pfSense devices but they are virtual running on XenServer, still this would 
indicate a problem in my opinion.
Thanks in advance for any assistance on this issue. Stepping back from the 
ledge now LOL

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[asterisk-users] ITSP Gateway Solution?

[Todd R .]

Right now we I am using Asterisk boxes as a gateway between our Level 3 SIP 
trunks and our customer PBXs.
I love and understand Asterisk but the company I am working for is looking for 
a more "Commercial" type solution where we can go to a vendor for support etc. 
I know, we can get Asterisk support etc.. It's not my decision and I sort of 
get why they are leaning away from Asterisk, I just don't agree.
I need to at least explore other options for more appliance products that will 
do the job the Asterisk boxes are doing now but, with a simple interface to 
add/remove trunks, DIDs etc. Integrated security and billing options/add-ons 
would be great.
I know Digium offers appliance solutions but they don't seem to be anywhere 
near the power of what we are currently using.
One big advantage I could see is going diskless but, I am really not sure whats 
out there, I am just kicking tires at the moment.
The best of all worlds would be something with commercial support, a good GUI, 
billing and security built in but all based on the Asterisk core which I can 
understand :-)
Again, just kicking tires as I can't just scream Asterisk and not be willing to 
look around to see what's out there.
Everything I see out there seems to want to Transcode and such.. All we need is 
something to do SIP to SIP, no TDM here at all. Some codec support beyond G711 
of course but that's it.
I know there is every reason to do all this with Asterisk and that is my 
preference but in this case, I have lots of folks that lean more towards 
commercial products and I have not been able to completely sell them on the joy 
and flexibility of Asterisk.
I don't want a Virtual PBX GUI solution, I want something that is built to be a 
work-horse, as a gateway only. No extensions, voicemail, ring groups or any of 
that. Just calls in/out to/from trunks, security and billing.
Thanks!   -- 
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Re: [asterisk-users] Setting Music on Hold with the Manager Interface

[Todd R .]

Thanks Matt.
I tried that already, no luck.
Still, I get blank nothingness instead of MOH. I will try again just to be sure 
I didn't miss something.

>>>>I have also tried surrounding musicclass with CHANNEL() but that didn't 
>>>>work and didn't seem right anyhow since it already knows it's a channel 
>>>>variable.>>>>

Date: Mon, 27 Oct 2014 08:51:42 -0500
Subject: Re: [asterisk-users] Setting Music on Hold with the Manager Interface
From: mjor...@digium.com
To: tjrl...@live.com; asterisk-users@lists.digium.com



On Sun, Oct 26, 2014 at 10:42 PM, Todd R.  wrote:



Does anyone know how to set the music on hold class with the Manager Interface 
in 1.8?
Here is what I am using but I end up just getting no music when I put this in 
place, when I remove it the default is back.
The classes I am setting work elsewhere just fine.
I did not include the opening of the socket, logging in etc because that's all 
working fine along with other things I am doing within the same login, socket 
session. Just trying to add this additional task.
This is from PHP as you may have recognized. I have also tried surrounding 
musicclass with CHANNEL() but that didn't work and didn't seem right anyhow 
since it already knows it's a channel variable.
Thanks in advance for any help on this.# Set the Music on Hold
fputs($socket2, "Action: Setvar\r\n");
fputs($socket2, "Channel: ".$channel."\r\n");
fputs($socket2, "Variable: musicclass\r\n");
fputs($socket2, "Value: ".$mohclass."\r\n");
  

Use the CHANNEL function:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CHANNEL

Action: SetvarChannel: (your channel name here)Variable: 
CHANNEL(musicclass)Value: (your MoH class here)
-- 
Matthew Jordan
Digium, Inc. | Engineering Manager445 Jan Davis Drive NW - Huntsville, AL 35806 
- USACheck us out at: http://digium.com & http://asterisk.org



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[asterisk-users] Setting Music on Hold with the Manager Interface

[Todd R .]

Does anyone know how to set the music on hold class with the Manager Interface 
in 1.8?
Here is what I am using but I end up just getting no music when I put this in 
place, when I remove it the default is back.
The classes I am setting work elsewhere just fine.
I did not include the opening of the socket, logging in etc because that's all 
working fine along with other things I am doing within the same login, socket 
session. Just trying to add this additional task.
This is from PHP as you may have recognized. I have also tried surrounding 
musicclass with CHANNEL() but that didn't work and didn't seem right anyhow 
since it already knows it's a channel variable.
Thanks in advance for any help on this.# Set the Music on Hold
fputs($socket2, "Action: Setvar\r\n");
fputs($socket2, "Channel: ".$channel."\r\n");
fputs($socket2, "Variable: musicclass\r\n");
fputs($socket2, "Value: ".$mohclass."\r\n");
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[asterisk-users] Lost audio on forwarded calls

[Todd R .]

OK, been messing with Asterisk for a long time and I have my opinion on where 
the issues lies but sometimes it's just nice to see what others think that can 
relate :-)
Here goes.. 
Inbound calls flow like this:Tier 1 Provider (SIP) > Asterisk 1.8 > Name Brand 
PBX - Calls work fine
Outbound calls flow like this:Name Brand PBX > Asterisk 1.8 > Tier 1 provider 
(SIP) - Calls work fine

Problem is being reported on that many (not all) calls have no audio when they 
are forwarded.
Example of forwarded call:Inbound call comes in from Tier 1 Provider > Asterisk 
1.8 > Name Brand PBX
Name Brand PBX then forwards the call back out to users cell phone:Name Brand 
PBX > Asterisk 1.8 > Tier 1 provider
No audio a large percentage of the time.

It's my opinion that the Asterisk box only sees the forwarded call as a regular 
outbound call and forwards it on to the Tier 1 provider then to the users cell 
phone.
I don't see how Asterisk even knows or cares if it was forwarded within the 
Name Brand PBX. The Name Brand PBX is the one making the connection of the 
inbound and outbound call. All other inbound and outbound calls are fine, audio 
is only lost when the Name Brand PBX connects the two calls and creates the 
forward.
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[asterisk-users] Asterisk and alternate RTP ports

[Todd R .]

Been working with Asterisk for a long time but this is the first time I have 
dealt with this issue.
I am setting up an Asterisk box (FreePBX not my choice) to interface with an 
e911 provider.
They say their switches only listen for RTP on ports 2-21001 which is 
outside the normal range Asterisk listens on 1-2.
I wish I knew more about this topic but since I have never had an issue 
interfacing with providers, ITSP etc., I just haven't had a need to know.
I get audio on some calls and others not so much.
How do I deal with this?
I don't really want to change the RTP ports that Asterisk listens on because 
this is a production system with trunks pointing to several other providers etc.
The 911 provider says I don't need to listen on 2-21001, that's just what 
they listen on.
In fact, they say this exactly "You can listen on what you want, as long as the 
RTP port is sent to us in the INVITE SDP info.".
Any assistance with solving this issue would be greatly appreciated, I have 
done my digging in Google etc before asking here as always.
Thanks in advance.







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Re: [asterisk-users] Text to Speech Engine

[Todd R .]

Actually, scratch that.. Luminvox is not text to speech it's speech recognition 
software. Got this mixed up and turned around :-) Anyhow, see the link I posted 
earlier, it's got some good info to get you started.

From: tjrl...@live.com
To: asterisk-users@lists.digium.com
Date: Fri, 10 Jan 2014 14:42:27 -0600
Subject: Re: [asterisk-users] Text to Speech Engine




Luminvox is one.. There are others out there.. 
Here's an article by Ward Mundy that might help:http://nerdvittles.com/?p=7448

From: jpra...@gmail.com
Date: Fri, 10 Jan 2014 12:16:43 -0800
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Text to Speech Engine

Hello, 

Anyone know good quality text to speach engine for building IVRs for asterisk. 
Open-source will be nice, but I wont mind paying for thing really good. 

Regards,


-Jai 


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Re: [asterisk-users] Text to Speech Engine

[Todd R .]

Luminvox is one.. There are others out there.. 
Here's an article by Ward Mundy that might help:http://nerdvittles.com/?p=7448

From: jpra...@gmail.com
Date: Fri, 10 Jan 2014 12:16:43 -0800
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Text to Speech Engine

Hello, 

Anyone know good quality text to speach engine for building IVRs for asterisk. 
Open-source will be nice, but I wont mind paying for thing really good. 

Regards,


-Jai 


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Re: [asterisk-users] Asterisk API

[Todd R .]

Search google for "Asterisk Manager Interface".
For the most part, if you have raw Asterisk installed then that's what you get 
and have to build what you want on top of it or hire a developer to do it.
Date: Fri, 10 Jan 2014 12:12:47 -0500
From: szilvertho...@gmail.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk API

Hello Folks;
I have an Asterisk server Asterisk 11.7.0 built by root @xxx on a 
x86_64 running Linux on 2013-12-27 18:47:44 UTC

No FreePBX, no AsteriskNOW, no Elastix. Just Asterisk.

Is there an API out there that anyone knows of that I can pass commands, etc to 
Asterisk? Creating Extensions, adding voicemail users, setting up voicemail, 
etc?
I'm kind of clueless. Is there something available?

Thanks - Glen

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Re: [asterisk-users] Convert Asterisk Appliance (AA50) to "Open" Asterisk?

[Todd R .]

May not be what you are looking for exactly but search Google for "Nerd Vittles 
BeagleBone". I am not suggesting you use that exact solution but, reading the 
article with give you all sorts if ideas about what you could use in your 
situation.

The BeagleBone is a small form factor computer like the Raspberry Pi but more 
powerful for not much more $$. They have a few builds you can out in it with 
full instructions. For your analog lines you could just use a couple if cheap 
SIP gateway devices which I think run $50-80/ea. USD.

In a small environment, this is what I would be using these days and I am 
itching to build something on the BeagleBone boards.

Total solution with a few SIP gateway devices should not cost you much at all. 
The boards are under $50 and a nice little case can be had for under $18 or 
maybe way less.

You could likely use your existing device as a pass through for your analog 
lines if you don't want to purchase the SIP gateways right away.

Let me know if you go down this road and what your results are. Good luck.




> On Dec 28, 2013, at 12:38 PM, "Lincoln King-Cliby"  
> wrote:
> 
> Hi All, 
> 
> Thanks for all of the help I've been given in the past and info I've picked 
> up from this list over the years. 
> 
> I have an "official" Asterisk appliance (the AA50) running my PBX at home (we 
> previously also had an AA50 in a satellite office-that one was recently 
> retired and replaced with Asterisk running on commodity server hardware). 
> 
> Anyway - the AA50 software/Asterisk version is beyond outdated at this point, 
> and the GUI has done nothing but infuriate me. Has anyone - or does anyone 
> know if it's possible to - replace the "commercial" Linux/Asterisk running on 
> the AA50 with another Linux flavor (say Ubuntu) and current open source 
> Asterisk (ideally 11.something with Gareth's Cisco patch).
> 
> I don't need - or want - a pretty GUI... just something I can SSH into and 
> perhaps manhandle config files with Nano or something similar - worst case, 
> something I can FTP/TFTP configuration files to. 
> 
> If that isn't feasible, anything low power/low profile/low cost that's 
> particularly popular these days [bonus points if it's wall mountable/about 
> the same size as the AA50]? My demands really aren't that severe -- one FXO, 
> two FXS, a SIP "trunk" to the office (via hardware VPN), and maybe a half 
> dozen Cisco 79xx phones. 
> 
> If it's not already apparent, I'm a relative Linux newb, but I'm farly well 
> versed in patching and building Asterisk from source and generally getting 
> things done once I have the pointers.
> 
> Thanks in advance -- and happy new year!
> 
> --
> Lincoln King-Cliby, CTS, DMC-D
> Commercial Market Director
> Sr. Systems Architect | Crestron Certified Master Programmer (Silver) 
> ControlWorks Consulting, LLC Crestron Services Provider 
> http://www.controlworks.com
> 
> 
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Re: [asterisk-users] Answering agent

[Todd R .]

I do this by writing custom CDR. I write the agents extension write into the 
CDR records. This makes is easy to just parse through the CDR and get all the 
info you need about the call.

Google something like "asterisk custom CDR"




> On Nov 29, 2013, at 11:36 AM, "Leandro Dardini"  wrote:
> 
> Hello friends,
> when a call arrives in the queue, a CDR record is created, but there is no 
> info about which agent has picked up the call. I can find that info only in 
> queue_log.
> 
> Is there a way to have that info in the CDR or maybe in a variable in the "h" 
> context, when the call is ended?
> 
> Leandro
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Re: [asterisk-users] Amazon, Asterisk and reliability beyond a hobby system?

[Todd R .]

Did you have the externalip setting in sip.conf set to the Elastic IP?


> Date: Sat, 23 Nov 2013 23:42:36 -0500
> From: ja...@fivecats.org
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Amazon, Asterisk and reliability beyond a hobby 
> system?
> 
> On 11/22/2013 12:52 PM, Todd R. wrote:
> > Just checking one more time to see if anyone has an opinion on this. I
> > am primarily interested in using a cloud type setup such as Amazon AWS
> > for the redundancy, easy backup and recovery options. It's not about
> > price but the idea that it will be very hard for a single piece of
> > hardware to ruin my day.
> 
> I have only one small datapoint.  I ran an EC2 microinstance with 
> Asterisk and a dozen offboard users.  The only problem I had was SIP 
> wasn't dealing well with the Elastic IP one-to-one NAT that Amazon uses. 
>   I had the usual Asterisk/NAT issues of one-way audio.  I eventually 
> moved from Amazon to Linode to get away from the NAT issues.  Once I did 
> that, everything worked fine, but again it was only a dozen users.
> 
> 
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Re: [asterisk-users] Amazon, Asterisk and reliability beyond a hobby system?

[Todd R .]

Oh and, I could be wrong but.. I suspect Twilio is one of the companies doing 
big things with Asterisk on AWS specifically.
I am 90% sure at this point that Twilio uses Asterisk as the base for their 
product. When I emailed them and asked them where their voice gateways were 
they mentioned something about Amazon's servers which I assumed to mean they 
were using Amazon's cloud services. The possibility of Twilio pushing tons of 
calls through virtualized Asterisk boxes is part of what has made me so curious 
about going down this road again.

From: tjrl...@live.com
To: asterisk-users@lists.digium.com
Date: Fri, 22 Nov 2013 12:18:35 -0600
Subject: Re: [asterisk-users] Amazon, Asterisk and reliability beyond a hobby 
system?




I would have said the same thing a while back but, I can't ignore the fact that 
there have been what seems to be many "Virtualization" success stories.
The idea that Asterisk just likes to be on it's own dedicated hardware has 
always caused me to prefer dedicated hardware.
But, is the possibility of a single piece of hardware failing "better" than 
something that will likely never just flat out die?
I know there are high availability solutions out there and it's not that I 
don't have backups and disaster recovery plans in place.
I just want to make things far better regarding redundancy, recovery and 
scalability and virtualization is hard to beat when you start talking about 
these things.
There are definitely people/companies using virtualized Asterisk solutions 
successfully, so I feel like it can be done.
Asterisk has come a long way since I first starting messing with Asterisk and 
so has Asterisk itself.
So, I am trying to determine what is bad, what to look out for in terms of 
virtualizing. If it's still as bad of an idea as it was say 5 years ago, then I 
need to understand why and if there is a work around.
At this point, the benefits of virtualizing my Asterisk boxes are too many to 
count. So, if I can't find any concrete reasons to NOT do this beyond "That's a 
bad idea" then I am going to give it a go. If I do, I am looking for any advice 
good or bad from those that have gone down this road successfully or with 
miserable failure.
My opinion all along has been Asterisk + Virtualization + Real Live Production 
Use = BAD IDEA!
Now, I am trying to figure out if that's just the opinion of an old man (sort 
of old) who just doesn't want to accept that virtualization if a better way (in 
terms of Asterisk).
So, I am hoping for people to tell me why Amazon AWS specifically is a good or 
bad idea with as much detail as possible.
Thanks!

> To: tjrl...@live.com; asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Amazon, Asterisk and reliability beyond a hobby 
> system?
> Date: Fri, 22 Nov 2013 13:04:44 -0500
> From: cov...@ccs.covici.com
> 
> I would thinktwice about Amazon -- and virtual in general is not a good
> idea for this sort of thing.  I have seen messages about bad results
> with amazon specifically.
> 
> Todd R.  wrote:
> 
> > Just checking one more time to see if anyone has an opinion on this. I am 
> > primarily interested in using a cloud type setup such as Amazon AWS for the 
> > redundancy, easy backup and recovery options. It's not about price but the 
> > idea that it will be very hard for a single piece of hardware to ruin my 
> > day.
> > 
> > From: tjrl...@live.com
> > To: asterisk-users@lists.digium.com
> > Date: Mon, 18 Nov 2013 18:33:38 -0600
> > Subject: [asterisk-users] Amazon,   Asterisk and reliability beyond a hobby 
> > system?
> > 
> > 
> > 
> > 
> > Took me a while but I have finally embraced cloud computing and all the 
> > benefits.
> > The only thing I have yet to feel comfortable about putting in the cloud is 
> > real live Asterisk boxes to be used in production. I know it's being done 
> > because as far as I know Twilio is using Amazon for their Asterisk boxes.
> > I have read all the fun articles on building hobby type systems and that's 
> > all great.
> > What I really need to hear is from those that have deployed Asterisk in 
> > Amazon or Digital Ocean and how many simultaneous calls they are pushing 
> > through it and what the call quality and reliability has been.
> > Right now I am still using dedicated hardware but I could become much more 
> > redundant and scale much faster using Amazon or Digital Ocean.
> > Thanks in advance for any information from those that have already been 
> > down this road... 
> > 
> > -- 
> > _
> > --

Re: [asterisk-users] Amazon, Asterisk and reliability beyond a hobby system?

[Todd R .]

I would have said the same thing a while back but, I can't ignore the fact that 
there have been what seems to be many "Virtualization" success stories.
The idea that Asterisk just likes to be on it's own dedicated hardware has 
always caused me to prefer dedicated hardware.
But, is the possibility of a single piece of hardware failing "better" than 
something that will likely never just flat out die?
I know there are high availability solutions out there and it's not that I 
don't have backups and disaster recovery plans in place.
I just want to make things far better regarding redundancy, recovery and 
scalability and virtualization is hard to beat when you start talking about 
these things.
There are definitely people/companies using virtualized Asterisk solutions 
successfully, so I feel like it can be done.
Asterisk has come a long way since I first starting messing with Asterisk and 
so has Asterisk itself.
So, I am trying to determine what is bad, what to look out for in terms of 
virtualizing. If it's still as bad of an idea as it was say 5 years ago, then I 
need to understand why and if there is a work around.
At this point, the benefits of virtualizing my Asterisk boxes are too many to 
count. So, if I can't find any concrete reasons to NOT do this beyond "That's a 
bad idea" then I am going to give it a go. If I do, I am looking for any advice 
good or bad from those that have gone down this road successfully or with 
miserable failure.
My opinion all along has been Asterisk + Virtualization + Real Live Production 
Use = BAD IDEA!
Now, I am trying to figure out if that's just the opinion of an old man (sort 
of old) who just doesn't want to accept that virtualization if a better way (in 
terms of Asterisk).
So, I am hoping for people to tell me why Amazon AWS specifically is a good or 
bad idea with as much detail as possible.
Thanks!

> To: tjrl...@live.com; asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Amazon, Asterisk and reliability beyond a hobby 
> system?
> Date: Fri, 22 Nov 2013 13:04:44 -0500
> From: cov...@ccs.covici.com
> 
> I would thinktwice about Amazon -- and virtual in general is not a good
> idea for this sort of thing.  I have seen messages about bad results
> with amazon specifically.
> 
> Todd R.  wrote:
> 
> > Just checking one more time to see if anyone has an opinion on this. I am 
> > primarily interested in using a cloud type setup such as Amazon AWS for the 
> > redundancy, easy backup and recovery options. It's not about price but the 
> > idea that it will be very hard for a single piece of hardware to ruin my 
> > day.
> > 
> > From: tjrl...@live.com
> > To: asterisk-users@lists.digium.com
> > Date: Mon, 18 Nov 2013 18:33:38 -0600
> > Subject: [asterisk-users] Amazon,   Asterisk and reliability beyond a hobby 
> > system?
> > 
> > 
> > 
> > 
> > Took me a while but I have finally embraced cloud computing and all the 
> > benefits.
> > The only thing I have yet to feel comfortable about putting in the cloud is 
> > real live Asterisk boxes to be used in production. I know it's being done 
> > because as far as I know Twilio is using Amazon for their Asterisk boxes.
> > I have read all the fun articles on building hobby type systems and that's 
> > all great.
> > What I really need to hear is from those that have deployed Asterisk in 
> > Amazon or Digital Ocean and how many simultaneous calls they are pushing 
> > through it and what the call quality and reliability has been.
> > Right now I am still using dedicated hardware but I could become much more 
> > redundant and scale much faster using Amazon or Digital Ocean.
> > Thanks in advance for any information from those that have already been 
> > down this road... 
> > 
> > -- 
> > _
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> > New to Asterisk? Join us for a live introductory webinar every Thurs:
> >http://www.asterisk.org/hello
> > 
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> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users  
> >   
> > 
> > Alternatives:
> > 
> > 
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> > New to Aste

Re: [asterisk-users] Amazon, Asterisk and reliability beyond a hobby system?

[Todd R .]

Just checking one more time to see if anyone has an opinion on this. I am 
primarily interested in using a cloud type setup such as Amazon AWS for the 
redundancy, easy backup and recovery options. It's not about price but the idea 
that it will be very hard for a single piece of hardware to ruin my day.

From: tjrl...@live.com
To: asterisk-users@lists.digium.com
Date: Mon, 18 Nov 2013 18:33:38 -0600
Subject: [asterisk-users] Amazon,   Asterisk and reliability beyond a hobby 
system?




Took me a while but I have finally embraced cloud computing and all the 
benefits.
The only thing I have yet to feel comfortable about putting in the cloud is 
real live Asterisk boxes to be used in production. I know it's being done 
because as far as I know Twilio is using Amazon for their Asterisk boxes.
I have read all the fun articles on building hobby type systems and that's all 
great.
What I really need to hear is from those that have deployed Asterisk in Amazon 
or Digital Ocean and how many simultaneous calls they are pushing through it 
and what the call quality and reliability has been.
Right now I am still using dedicated hardware but I could become much more 
redundant and scale much faster using Amazon or Digital Ocean.
Thanks in advance for any information from those that have already been down 
this road... 

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[asterisk-users] Amazon, Asterisk and reliability beyond a hobby system?

[Todd R .]

Took me a while but I have finally embraced cloud computing and all the 
benefits.
The only thing I have yet to feel comfortable about putting in the cloud is 
real live Asterisk boxes to be used in production. I know it's being done 
because as far as I know Twilio is using Amazon for their Asterisk boxes.
I have read all the fun articles on building hobby type systems and that's all 
great.
What I really need to hear is from those that have deployed Asterisk in Amazon 
or Digital Ocean and how many simultaneous calls they are pushing through it 
and what the call quality and reliability has been.
Right now I am still using dedicated hardware but I could become much more 
redundant and scale much faster using Amazon or Digital Ocean.
Thanks in advance for any information from those that have already been down 
this road... -- 
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Re: [asterisk-users] Make phone ring through webserver using Asterisk

[Todd R .]

What do you want to happen once the call is made?
You can choose to fire the call off using the originate command with the 
Asterisk Manager Interface from a PHP page or some other similar language. No 
need for Perl on the Asterisk box at all really unless you need it for 
something else.
http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate


Date: Sat, 16 Nov 2013 16:53:59 +0530
From: omakhileshch...@gmail.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Make phone ring through webserver using Asterisk

What is the easiest way? And how can it be implemented?
I thought to something like:
I request a page to the webserverPerl sends to asterisk a number to dial (Perl 
and asterisk are running in the same machine)
Asterisk calls the phoneor
A Perl sip client registers to remote asterisk serverPerl sip client sends to 
asterisk the number to dialPhone ringsi don't care if i can hear something, 
it's enough that it rings



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[asterisk-users] chan_sip.c:9602 copy_header: No field 'CSeq' present to copy

[Todd R .]

Just put a new phone in place with the latest firmware from Cisco. This is the 
first SPA501G we have with this firmware.
In the Asterisk CLI we are now seeing the error message below about once every 
second. When we unplug the phone, the messages quit.










NOTICE[15539]: chan_sip.c:9602 copy_header: No field 'CSeq' present to copy


Thanks in advance for any assistance on this.   
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[asterisk-users] Pull call out of queue

[Todd R .]

Trying to figure out the best way to pull an active call out of a queue by 
unique id and put it on hold. I don't want to put it on hold on the agent's 
phone but I want it to be pulled away from the agent's phone and into Asterisk 
limbo somewhere.
Shortly after I want to pull the same call out of limbo and redirect it back to 
either the same agent or another.
I was thinking about call parking but, I think parking is more than I need and 
it potentially introduces more complications.
I will be doing this through the manager interface on Asterisk 1.8.x.
Any ideas, thoughts or help would be greatly appreciated.
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Re: [asterisk-users] Local agent/member in-use after transfer

[Todd R .]

From: tjrl...@live.com
To: asterisk-users@lists.digium.com
Date: Thu, 1 Aug 2013 12:50:32 -0500
Subject: [asterisk-users] Local agent/member in-use after transfer




I currently have all agents/members logged in with local channels. When a call 
is sent to one of the agents, then the agent transfers the call out the line 
frees up on their phone but still shows in-use until the call that was 
transferred is hung up.
How can I free up the agent/local channel when the call is transferred?
This is a huge problem because the agent can no longer receive calls on their 
extension. If they are the only agent logged in, then no other calls can be 
answered. If the transferred calls last an hour then no calls can be answered 
by this agent for an hour.
I know I can set ringinuse=yes but this causes the agent to be interrupted 
while on calls which is not the desired result. 
   

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I found a solution and wanted to post it for those that may run into this 
trouble in the future.
I use the manager interface to login my agents using a web page.
After much digging I finally found the StateInterface: option available in 1.6 
and above. I added it to my PHP login screen like this..
fputs($socket2, "StateInterface: SIP/".$agentid."\r\n");
The problem is that the queue was monitoring the local channel in terms of when 
a call was hungup or not, allowing other calls to come through.
When a transfer happened the Local channel was not released.
Adding the StateInterface option apparently allows the queue to monitor the 
actual channel, not the local channel. I couldn't find much documentation on 
this option, just stumbled upon it.
Fixed my issue though! Thought I would add to the little info that seems to be 
out there about this option.   --
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[asterisk-users] Local agent/member in-use after transfer

[Todd R .]

I currently have all agents/members logged in with local channels. When a call 
is sent to one of the agents, then the agent transfers the call out the line 
frees up on their phone but still shows in-use until the call that was 
transferred is hung up.
How can I free up the agent/local channel when the call is transferred?
This is a huge problem because the agent can no longer receive calls on their 
extension. If they are the only agent logged in, then no other calls can be 
answered. If the transferred calls last an hour then no calls can be answered 
by this agent for an hour.
I know I can set ringinuse=yes but this causes the agent to be interrupted 
while on calls which is not the desired result. 
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Re: [asterisk-users] Asterisk 1.4 CDR vs VoIP Innovations CDR

[Todd Routhier]

May be as simple as this:

When you terminate a call you start the call before they even get it.

When they originate a call, they start the call before you get it.

Just a guess without really thinking about this too much.


On Thu, Aug 1, 2013 at 10:28 AM, Adam Moffett wrote:

> When I compare my total minutes on the bill from VoIP Innovations, to the
> number from our CDRs, I'm finding a smalish (3-4%) discrepancy in the count
> of minutes.  I'm wondering why it's there.
>
> Are there different methods of counting the billable start or end point of
> a phone call?
>
> If it matters, I'm counting more termination minutes than they are and
> they're counting more origination minutes than I am.
>
>
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Re: [asterisk-users] Transfer only, no outbound calling

[Todd Routhier]

Nathan,

 Yes, SIP.. :-)

I ended up deciding to just not allow attended transfer at all since it
seemed so hard to deal with. If someone really wants attended transfer they
can put the call on hold, dial using the other line then transfer the call
on the other line if they want the call on the other end. Same thing, just
one more step.

I am just going to set a var in sip.conf so when people try to dial out
direct, it will catch it in the dial plan and kill the call. With blind
transfer I can set a var on the way in and it's held onto nicely and I can
allow the transfer based on that.

Again, thanks for your detailed response.



On Tue, Apr 16, 2013 at 9:59 PM, Nathan Anderson  wrote:

> On Tuesday, April 16, 2013 6:25 PM, Todd Routhier wrote:
>
> > New Problem, now operators can pick up the previous inbound only line and
> > dial out to anything that matches the patterns I have defined in the
> > context for their extension in sip.conf.
> >
> > What I really need to make work here is Attended-Transfer since that is
> > what is desired by those using the system.
>
> I'll assume we are talking about SIP extensions here.
>
> What is doing the actual transfer?  Is it Asterisk (res_features /
> features.conf), or the phones themselves?
>
> If it is the phones themselves, you're probably out of luck because in an
> attended transfer scenario, the transferor has to send a regular ol' INVITE
> to the transfer target before sending a REFER to the transferee, and so
> there's really no way that Asterisk can know whether that INVITE to the
> transfer target is someone in the middle of attempting an attended
> transfer, or someone trying to place a regular outbound call.  Your only
> hope would be to sniff the SIP traffic between your handsets and Asterisk,
> and see if there is a SIP header difference that is detectable between what
> your phones generate for an attended transfer vs. an outbound call.  If
> there is, you can use the ${SIP_HEADER()} function in your dialplan to
> check for the presence of that difference in order to determine whether a
> call is an attended transfer or not.
>
> If you have the option of using Asterisk's built-in attended transfer
> feature (features.conf + passing option 't' to the Dial() command that
> calls a given extension for an inbound call) instead of a button on your
> phones, you can override which context a transfer target's number is
> executed in by overriding the global variable TRANSFER_CONTEXT.  So you can
> create a new stub context that sets your variable to let you know that this
> is a transfer and then jumps to the SIP client's normal context, and set
> TRANSFER_CONTEXT=your_new_context under the [globals] section of
> extensions.conf.  Check for the presence of your variable in the SIP
> client's context, and act accordingly.
>
> Note that in either scenario, as long as you allow attended transfers, the
> system can be gamed by people.  For example, assuming that extensions can
> call other extensions, someone who wants to make an unsanctioned outbound
> call simply walks over to a vacant phone in another cubicle, calls their
> own phone/extension, rushes back to answer it, and then initiates an
> attended transfer that they never end up completing (they just talk to the
> person they initiated the transfer to the whole time).
>
> Hope this helps,
>
> --
> Nathan Anderson
> First Step Internet, LLC
> nath...@fsr.com
>
> --
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[asterisk-users] Transfer only, no outbound calling

[Todd Routhier]

OK, it's been a while since I drank from the pool of wisdom hear on the
list.

After cracking my head against the wall for a few days trying to figure
this out, I have decided to swallow my pride and take the drink.

So, on to my question:

I have some agents/operators setup in sip.conf which point to a context
where I have just about disabled outbound calls (only specific numbers can
be dialed).

The purpose of this is to allow the inbound calls to come in, then if the
operator has a need, they transfer the call to a pre-defined extension
which lives in the limited context defined in sip.conf.

This has worked for some time to restrict outbound calling and where calls
can be transferred to.

Now I would like to open up the numbers the inbound calls can be
transferred to. So, easy enough I thought and I went on my merry way adding
the regular patterns to the context such as NXXNXX and so on.

Hooray, now the operators can transfer anywhere.

New Problem, now operators can pick up the previous inbound only line and
dial out to anything that matches the patterns I have defined in the
context for their extension in sip.conf.

What I really need to make work here is Attended-Transfer since that is
what is desired by those using the system.

It seems that any variables I try to set on the way in don't carry through
too well during an attended transfer.

Basically, I need the ability to know for sure at the point the call ends
up in the outbound context (defined in sip.conf) if the call is actually a
transfer from an inbound call or if it's a direct dial outbound call with
no incoming call attached. If I can figure out how to know this for sure, I
can just do a GoToIf type of thing in the outbound context that just kills
the call if there is no proof that it's a transfer.

I hope this makes sense, please let me know if more info is needed.

Running Asterisk 1.8.8.0.

A huge thanks in advance to the list for any help with this, it's driving
me batty.

Regards,
 Todd R.
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Re: [asterisk-users] Run AGI while agent ringing instead of only when connected

[Todd Routhier]

Thanks again Danny, Perl was the first thing I tinkered with back in the
90's but haven't messed with it for years.

Looking over what you sent, I get about 90% of what's going on there. With
a little searching and brushing up on my Perl, I think I will be able to
make this work.

This is a good solution and, if I can get this to work, I won't even need
the AGI. I can basically just hit what I need using CURL within the Perl
script (I think).

All the AGI was going to do for me is hit a URL with some parameters out on
the Internet. So, pretty sure I can do all that within the Perl Script and
leave AGI out of it completely.

--Todd

On Tue, Apr 10, 2012 at 4:02 PM, Danny Nicholas  wrote:

> Were this my task, I would do a PERL/C daemon to run the AGI.  This is how
> I do it in PERL
>
>my $astman = new Asterisk::Manager;
>
>$astman->user('user');
>
>$astman->secret('secret');
>
>my $man_addr='127.0.0.1';
>
> 
>
>my $man_ok=1;
>
>open (my $man_in, "/etc/asterisk/manager.conf") or
> $man_ok=undef;
>
>if ($man_ok) {
>
>   while (<$man_in>) {
>
>  if ($_ =~ /^bindaddr/) {
>
> (undef,$man_addr) = split /\=/, $_;
>
> }
>
>  }
>
>   close $man_in;
>
>   }
>
>$man_addr =~ s/\s//g;
>
> 
>
>( $man_addr )=( $man_addr =~ /(.*)/ );
>
> 
>
>$astman->host($man_addr);
>
>$astman->connect || die "Could not connect to " .
> $astman->host . "!\n";
>
> ** **
>
>my %resp = $astman->sendcommand(  Action => 'Originate',***
> *
>
>Channel =>
> $extval,
>
>Variable =>
> "ARG1=$fileval",
>
>Exten =>
> $extval,
>
>Context =>
> 'playit',
>
>priority => 1,*
> ***
>
>    Number =>
> 5551212
>
>);
>
> 
>
>sleep 2;
>
>%resp = $astman->sendcommand(  Action => 'Logoff');
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Todd Routhier
> *Sent:* Tuesday, April 10, 2012 3:55 PM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Run AGI while agent ringing instead of
> only when connected
>
> ** **
>
> Yes Sir.. Studied it pretty hard, did I miss a solution? Trust me, been at
> this for a number of years off and on, I never post unless I have dug hard,
> searching all the Asterisk resources I know of.
>
> ** **
>
> This is where I got most of my info but the solutions mentioned on that
> page require the call to be "Connected" to the agent before the AGI fires.
> Once the agent is connected, I can get all sorts of info from Channel Vars.
> Still, once the agent is connected, it's sort of too late, I need the AGI
> to fire will the agent is ringing.
>
> ** **
>
> Thanks for your help so far.
>
> On Tue, Apr 10, 2012 at 3:42 PM, Danny Nicholas  wrote:
> 
>
> You have read this thread?
> http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue
>
>  
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Todd Routhier
> *Sent:* Tuesday, April 10, 2012 3:15 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Run AGI while agent ringing instead of only
> when connected
>
>  
>
> What I am trying to accomplish is to run an AGI script each time an
> agent's line starts ringing. I currently have the AGI firing when the agent
> answers the call using the Queue command, something like
> queue(MyQueue,MyAgi.php). Works great but I need the AGI to run when
> the agent's phone starts ringing.
>
>  
>
> Strangely, I can&

Re: [asterisk-users] Run AGI while agent ringing instead of only when connected

[Todd Routhier]

Yes Sir.. Studied it pretty hard, did I miss a solution? Trust me, been at
this for a number of years off and on, I never post unless I have dug hard,
searching all the Asterisk resources I know of.

This is where I got most of my info but the solutions mentioned on that
page require the call to be "Connected" to the agent before the AGI fires.
Once the agent is connected, I can get all sorts of info from Channel Vars.
Still, once the agent is connected, it's sort of too late, I need the AGI
to fire will the agent is ringing.

Thanks for your help so far.

On Tue, Apr 10, 2012 at 3:42 PM, Danny Nicholas  wrote:

> You have read this thread?
> http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Todd Routhier
> *Sent:* Tuesday, April 10, 2012 3:15 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Run AGI while agent ringing instead of only
> when connected
>
> ** **
>
> What I am trying to accomplish is to run an AGI script each time an
> agent's line starts ringing. I currently have the AGI firing when the agent
> answers the call using the Queue command, something like
> queue(MyQueue,MyAgi.php). Works great but I need the AGI to run when
> the agent's phone starts ringing.
>
> ** **
>
> Strangely, I can't find anything real useful on this after searching
> Google, this list, various Asterisk forums etc.
>
> ** **
>
> Is this supported? If not, is there some other maybe not so supported way
> to accomplish this?
>
> ** **
>
> I get how I can just fire an AGI from the dial plan but once I leave
> control to the queue, I can't really do that, I don't think.
>
> ** **
>
> Thanks in advance for any help!
>
> ** **
>
> --Todd
>
> ** **
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Run AGI while agent ringing instead of only when connected

[Todd Routhier]

I was trying to leave the AMI out of this because I am unsure if I can
monitor it in real time without building an external listener which Flash
Operator Panel does for me right now.

When I went down this road, I thought it would be a piece of cake to just
fire an AGI, well it is until you get queues involved.

--Todd

On Tue, Apr 10, 2012 at 3:34 PM, Carlos Chavez wrote:

> On Tue, 2012-04-10 at 15:15 -0500, Todd Routhier wrote:
> > What I am trying to accomplish is to run an AGI script each time an
> > agent's line starts ringing. I currently have the AGI firing when the
> > agent answers the call using the Queue command, something like
> > queue(MyQueue,MyAgi.php). Works great but I need the AGI to run
> > when the agent's phone starts ringing.
> >
> >
> > Strangely, I can't find anything real useful on this after searching
> > Google, this list, various Asterisk forums etc.
> >
> >
> > Is this supported? If not, is there some other maybe not so supported
> > way to accomplish this?
> >
> >
> > I get how I can just fire an AGI from the dial plan but once I leave
> > control to the queue, I can't really do that, I don't think.
> >
> >
> The only way I really see to do that is to monitor events via AMI
> so
> you can trigger the AGI when the phone starts to ring.
>
>
> --
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez Prats
> Director de Tecnología
> +52-55-91169161 ext 2001
>
> --
> _
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Re: [asterisk-users] Run AGI while agent ringing instead of only when connected

[Todd Routhier]

Thanks Danny.

What I am trying to do is send a popup screen to the agent. I am doing this
now using Flash Operator Panel but I am trying to get away from that. I
need to know the agent the call is being sent to when calling the AGI. So,
right now I am getting all that but only when the call is connected.
Running the AGI first will just allow me to popup a window for all agents
instead of the one that is currently ringing. This is why I need it to be
fired as part of the queue, it seems only the Queue (and the AMI) know
which agent is ringing.

--Todd


On Tue, Apr 10, 2012 at 3:21 PM, Danny Nicholas  wrote:

> Put your Queue command In a macro like this
>
> [agi-and-queue]
>
> Exten => s,1,Verbose(start AGI then do queue)
>
> Exten => s,n,AGI(queproc.sh)
>
> Exten => s,n,queue(myqueue)
>
> ** **
>
> You will need to put nohup into the AGI so it can run whether the line
> gets picked up or not.
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Todd Routhier
> *Sent:* Tuesday, April 10, 2012 3:15 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Run AGI while agent ringing instead of only
> when connected
>
> ** **
>
> What I am trying to accomplish is to run an AGI script each time an
> agent's line starts ringing. I currently have the AGI firing when the agent
> answers the call using the Queue command, something like
> queue(MyQueue,MyAgi.php). Works great but I need the AGI to run when
> the agent's phone starts ringing.
>
> ** **
>
> Strangely, I can't find anything real useful on this after searching
> Google, this list, various Asterisk forums etc.
>
> ** **
>
> Is this supported? If not, is there some other maybe not so supported way
> to accomplish this?
>
> ** **
>
> I get how I can just fire an AGI from the dial plan but once I leave
> control to the queue, I can't really do that, I don't think.
>
> ** **
>
> Thanks in advance for any help!
>
> ** **
>
> --Todd
>
> ** **
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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[asterisk-users] Run AGI while agent ringing instead of only when connected

[Todd Routhier]

What I am trying to accomplish is to run an AGI script each time an agent's
line starts ringing. I currently have the AGI firing when the agent answers
the call using the Queue command, something like
queue(MyQueue,MyAgi.php). Works great but I need the AGI to run when
the agent's phone starts ringing.

Strangely, I can't find anything real useful on this after searching
Google, this list, various Asterisk forums etc.

Is this supported? If not, is there some other maybe not so supported way
to accomplish this?

I get how I can just fire an AGI from the dial plan but once I leave
control to the queue, I can't really do that, I don't think.

Thanks in advance for any help!

--Todd
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Re: [asterisk-users] Mute DTMF

[Todd Routhier]

Thanks John and SamyGo,

 I tried your solution SamyGo and it does the trick fine. John, just liked
doing it in the dial plan better but I may use the firewall trick in the
future.

Thanks again!

--Todd


On Fri, Mar 30, 2012 at 12:45 AM, SamyGo  wrote:

> Hey,
> I not sure why your dtmfmode isn't working. The way I turned off the dtmf
> within an IVR was:
>
> 1- fix the dtmfmode of any sip user to rfc2833, so he is able to send dtmf
> to navigate within the IVR.
> 2- For places where I wanted to ignore any user DTMF key presses, I
> changed the dtmfmode of channel in the dialplan.
>
> That way I knew that the call will be negotiated on rfc2833 but changing
> that during the call ignores any key presses and reverting back again makes
> it functional again !!
>
> I hope this helps.
>
> Regards,
> Sammy.
>
> On Fri, Mar 30, 2012 at 2:54 AM, John Kiniston wrote:
>
>>
>> On Thu, Mar 29, 2012 at 12:09 PM, Todd Routhier wrote:
>>
>>> I have been breaking my head on this, can't find a solution.
>>>
>>> Anyone know a way to mute DTMF on SIP? I have already tried changing the
>>> dtmfmode option and messing with different codec/dtmfmode settings but so
>>> far, not having any luck.
>>>
>>>
>> It's not an asterisk based solution but you could use RFC2833 signalling
>> and then drop the RTP DTMF packets at your firewall.
>>
>>
>> --
>> A human being should be able to change a diaper, plan an invasion,
>> butcher a hog, conn a ship, design a building, write a sonnet, balance
>> accounts, build a wall, set a bone, comfort the dying, take orders, give
>> orders, cooperate, act alone, solve equations, analyze a new problem, pitch
>> manure, program a computer, cook a tasty meal, fight efficiently, die
>> gallantly. Specialization is for insects.
>> ---Heinlein
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>   http://www.asterisk.org/hello
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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[asterisk-users] Mute DTMF

[Todd Routhier]

I have been breaking my head on this, can't find a solution.

Anyone know a way to mute DTMF on SIP? I have already tried changing the
dtmfmode option and messing with different codec/dtmfmode settings but so
far, not having any luck.

Not even sure changing the codec is an option but pulling at straws at the
moment.

Thanks in advance for any assistance.
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[asterisk-users] Disconnect after 12 seconds w/Cisco 303g Phones

[Todd Routhier]

So, I have this customer with a completely bizarr issue.

She reports that on either of her shiny new Cisco 303g phone, calls are
disconnected at exactly 12 seconds after taking caller off hold.

To be clearer:
-Answers incoming call, can talk forever no problem.
-Places caller on hold and can leave them there forever.
-Picks up call and 12 seconds later, call disconnects.

I am going to turn on some debug logs on our Asterisk box a bit later and
watch the CLI as this happens to see if I can capture anything interesting
but wanted to check with the list in the meantime.

Other older Linksys/Cisco phone with slightly different model numbers don't
have this issue and have worked phone for many moons :-)

Never ran into this before, really puzzled. Google, searching the list
archives etc, weren't any help.

I am thinking this has to be the phones or some settings within. Two
phones, same model on same network are doing the same thing. No way to
easily get customer a different model phone to test with so I will have to
test with softphones to see if removing the phones from the equation
resolves the issue.

Any help would be appreciated, thanks.

--Todd
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Re: [asterisk-users] Define custom vm-login sound file per VM context?

[Todd Routhier]

Wow, that looks like good stuff.

On Tue, Feb 21, 2012 at 12:24 PM, Johan Wilfer  wrote:

> 2012-02-21 19:20, Todd Routhier skrev:
> > OK, this will work and is probably a better solution than the language
> > idea. Although, the language idea just sounds easier and a little more
> > fun :-)
> >
> > Hmm, I think I will try the language solution and see if it works with
> > a fake country/language code like Cust327 or whatever.
> >
> > Just wonder if that will break anything else now or with future upgrades.
> >
> > Thanks for all the help!
>
> You can also use en_baselevel_customer234 as a language,
> asterisk will first try to find a soundfile in the
> en_baselevel_customer234-directory, and if not found in the
> en_baselevel-directory. After that it will look in the en-dir.
>
> Can't find the docs for this right now but this way you don't need to
> copy all the recordings, and you can stack as many layers as you like.. :-)
>
> /Johan
>
> --
> Med vänlig hälsning
>
> Johan Wilfer email: jo...@jttech.se
> JT Tech | Utvecklare webb: http://jttech.se
> direkt: +46 31 380 91 01  support: +46 31 380 91 00
>
>
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Re: [asterisk-users] Define custom vm-login sound file per VM context?

[Todd Routhier]

OK, this will work and is probably a better solution than the language
idea. Although, the language idea just sounds easier and a little more fun
:-)

Hmm, I think I will try the language solution and see if it works with a
fake country/language code like Cust327 or whatever.

Just wonder if that will break anything else now or with future upgrades.

Thanks for all the help!

--Todd


On Tue, Feb 21, 2012 at 11:47 AM, Matthew Jordan  wrote:

>
> > From: "Todd Routhier" 
> > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> > 
> > Sent: Tuesday, February 21, 2012 11:30:34 AM
> > Subject: Re: [asterisk-users] Define custom vm-login sound file per
> > VM context?
>
> > Danny,
>
> > This seems to be a solution for sending people to leave a voicemail,
> > I need a solution for VoiceMailMain() when people call in to get
> > their messages, change greeting etc.
>
> > If I use the s option with VoiceMailMain it just skips checking the
> > passcode according to the docs.
>
> > Thanks for your help though, any similar ideas for VoiceMailMain?
>
> > I am playing the sound file I need before sending them to
> > VoiceMailMain but then Comedian Mail! plays right after of course.
>
> > --Todd
>
> The sound files referenced by voicemail.conf are global for all
> mailboxes defined in the configuration file, regardless of whether or
> not those mailboxes are defined in separate contexts.  Hence, whatever
> is defined for the 'vm-login' sound will be played for all users.
>
> For this one sound file (and this one sound file only), there is a
> mechanism you can use to bypass playing this sound file back.  You
> can tell VoiceMailMain to skip authentication of the user using the
> 's' flag, and use VMAuthenticate to authenticate the user yourself.
> Note that internally, VoiceMailMain uses VMAuthenticate, so you're
> using the exact same mechanism, just from the dialplan. If you pass the
> 's' flag to VMAuthenticate, it will not play the vm-login sound,
> allowing you, if you want, to play a different soundfile.
>
> In general, it would look something like this (please don't expect this
> to work verbatim, but it gives you an idea):
>
> exten => 1,1,NoOp()
> same => n,Background("Your-sound-file")
> same => n,VMAuthenticate(1@default,s)
> same => n,GotoIf($[${AUTH_MAILBOX}=1] &
> $[${AUTH_CONTEXT}=default]?auth:failed)
> same => n(auth),VoiceMailMain(s)
> same => n,Hangup()
> same => n(fail),Hangup()
>
> Matthew Jordan
> Digium, Inc. | Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: http://digium.com & http://asterisk.org
>
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Re: [asterisk-users] Define custom vm-login sound file per VM context?

[Todd Routhier]

Wow, that makes me wonder if I could do something like:

Set(CHANNEL(language)=Cust327)

Then create a "Language" folder named Cust327 and have it just work.
Weee... :-)

Of course that leads me to think that I could have whole sets of custom
sounds for all of Asterisk based on setting this Language bit on the way in.

Guess this would all work as long as there is not some requirement in
Asterisk that a language setting must be a real country/language code and
not something made up.

--Todd


On Tue, Feb 21, 2012 at 11:37 AM, Danny Nicholas  wrote:

> There was a “kludgy” solution posted a while back that might work for
> you.  Since Asterisk is “multi-lingual” you could do this
>
> Exten => _X.,123,Set(CHANNEL(language)=fr)
>
> Exten => _X.,124,Voicemailmain()
>
> ** **
>
> This assumes you aren’t using fr(French).  Just copy
> /var/lib/asterisk/sounds/en to /var/lib/asterisk/sounds/fr and record your
> alternate instructions in /var/lib/asterisk/sounds/fr/vm-login.gsm (or
> whatever codec you are using).  Using this work-around you could have as
> many greetings as you can specify “languages” for.
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Todd Routhier
> *Sent:* Tuesday, February 21, 2012 11:31 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Define custom vm-login sound file per VM
> context?
>
> ** **
>
> Danny,
>
> ** **
>
>  This seems to be a solution for sending people to leave a voicemail, I
> need a solution for VoiceMailMain() when people call in to get their
> messages, change greeting etc.
>
> ** **
>
> If I use the s option with VoiceMailMain it just skips checking the
> passcode according to the docs.
>
> ** **
>
> Thanks for your help though, any similar ideas for VoiceMailMain?
>
> ** **
>
> I am playing the sound file I need before sending them to VoiceMailMain
> but then Comedian Mail! plays right after of course.
>
> ** **
>
> --Todd
>
> ** **
>
> ** **
>
> On Tue, Feb 21, 2012 at 10:59 AM, Danny Nicholas 
> wrote:
>
> I believe this is what you want.  Instead of this
>
> Exten => _X.,123,Voicemail(100)
>
>  
>
> Do 
>
> Exten => _X.,123,playback(your-message)
>
> Exten => _X.,123,voicemail(100,s)
>
>  
>
> Per the instructions, (100) plays the standard message, (100,b) plays busy
> (100,u) plays unavailable and (100,s) plays nothing (skip instructions).**
> **
>
>  
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Todd Routhier
> *Sent:* Tuesday, February 21, 2012 10:53 AM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] Define custom vm-login sound file per VM
> context?
>
>  
>
> Is it possible to define a customize the which sound file is played when I
> send a caller to VoiceMailMain()?
>
>  
>
> By default the sound file is vm-login..
>
>  
>
> Is there a way to specify which sound file is played per context or some
> other way to play a different sound file in place of vm-login?
>
>  
>
> I have already replaced the default file and named it the same vm-login.x
> but still I am only able to play one file, not a different file depending
> on the VM context I send the caller to.
>
>  
>
> I am sure someone has figured this out so, any shortcut to keep me from
> frying my brain on this would be appreciated.
>
>  
>
> Thanks!
>
>  
>
> --Todd
>
>  
>
>
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> ** **
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Re: [asterisk-users] Define custom vm-login sound file per VM context?

[Todd Routhier]

Danny,

 This seems to be a solution for sending people to leave a voicemail, I
need a solution for VoiceMailMain() when people call in to get their
messages, change greeting etc.

If I use the s option with VoiceMailMain it just skips checking the
passcode according to the docs.

Thanks for your help though, any similar ideas for VoiceMailMain?

I am playing the sound file I need before sending them to VoiceMailMain but
then Comedian Mail! plays right after of course.

--Todd


On Tue, Feb 21, 2012 at 10:59 AM, Danny Nicholas  wrote:

> I believe this is what you want.  Instead of this
>
> Exten => _X.,123,Voicemail(100)
>
> ** **
>
> Do 
>
> Exten => _X.,123,playback(your-message)
>
> Exten => _X.,123,voicemail(100,s)
>
> ** **
>
> Per the instructions, (100) plays the standard message, (100,b) plays busy
> (100,u) plays unavailable and (100,s) plays nothing (skip instructions).**
> **
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Todd Routhier
> *Sent:* Tuesday, February 21, 2012 10:53 AM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] Define custom vm-login sound file per VM
> context?
>
> ** **
>
> Is it possible to define a customize the which sound file is played when I
> send a caller to VoiceMailMain()?
>
> ** **
>
> By default the sound file is vm-login..
>
> ** **
>
> Is there a way to specify which sound file is played per context or some
> other way to play a different sound file in place of vm-login?
>
> ** **
>
> I have already replaced the default file and named it the same vm-login.x
> but still I am only able to play one file, not a different file depending
> on the VM context I send the caller to.
>
> ** **
>
> I am sure someone has figured this out so, any shortcut to keep me from
> frying my brain on this would be appreciated.
>
> ** **
>
> Thanks!
>
> ** **
>
> --Todd
>
> ** **
>
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[asterisk-users] Define custom vm-login sound file per VM context?

[Todd Routhier]

Is it possible to define a customize the which sound file is played when I
send a caller to VoiceMailMain()?

By default the sound file is vm-login..

Is there a way to specify which sound file is played per context or some
other way to play a different sound file in place of vm-login?

I have already replaced the default file and named it the same vm-login.x
but still I am only able to play one file, not a different file depending
on the VM context I send the caller to.

I am sure someone has figured this out so, any shortcut to keep me from
frying my brain on this would be appreciated.

Thanks!

--Todd
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Re: [asterisk-users] Answering call from queue, then put back in queue?

[Todd Routhier]

Thanks Jim, I like that idea.


On Sun, Jan 8, 2012 at 12:42 PM, Jim Dickenson  wrote:

> One way to deal with this is to have two queues. Give priority to the
> original queue callers land in. Once answered put the call in to the second
> queue. They will then be in the second queue in the order the agents
> answered the first queue.
> --
> Jim Dickenson
> mailto:dicken...@cfmc.com 
>
> CfMC
> http://www.cfmc.com/
>
>
>
> On Jan 8, 2012, at 10:26 AM, Todd Routhier wrote:
>
> Version: Asterisk 1.8.x
>
> Question: Is it possible for an agent to answer a call from a queue, then
> place the call back in the queue in the same position they were in?
>
>
> Seems that the answer would be yes to the remove from queue, then place
> back in by having the agent just transfer the call back to the queue but is
> there any way to put them back in line where they were?
>
> The idea is that the owner of the queue doesn't want callers waiting on
> hold without first having an agent at least answer the call and ask them to
> please hold. What's the best way to handle this?
>
> Thanks in advance!
>
> --Todd
> --
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[asterisk-users] Answering call from queue, then put back in queue?

[Todd Routhier]

Version: Asterisk 1.8.x

Question: Is it possible for an agent to answer a call from a queue, then
place the call back in the queue in the same position they were in?


Seems that the answer would be yes to the remove from queue, then place
back in by having the agent just transfer the call back to the queue but is
there any way to put them back in line where they were?

The idea is that the owner of the queue doesn't want callers waiting on
hold without first having an agent at least answer the call and ask them to
please hold. What's the best way to handle this?

Thanks in advance!

--Todd
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Re: [asterisk-users] Ringing agents cell as an alert?

[Todd Routhier]

Sounds perfect, I will need to look into how to blend them together like
that.

I wonder though, will channel state still work using that method? I think
it's needed by something in the queue but I can't remember at the moment.



On Tue, Jan 3, 2012 at 12:15 PM, James Sharp  wrote:

> On 01/03/2012 01:06 PM, Todd Routhier wrote:
>
>> Happy New Year to all!
>>
>> Asterisk 1.8.x
>>
>> I have a queue to which I add agent channels like SIP/300 dynamically
>> using the manager interface. Once logged in, there SIP/300 of course
>> rings when a call is distributed to them.
>>
>> How can I also get the agents cell phone to ring without actually adding
>> it to the queue? I mean id I add something goofy like
>> SIP/MyProvider/1555444 to the queue, I don't know what will happen
>> at this point, haven't tested it. Even if it works (asterisk channel
>> state etc) it will mess with the queue and treat the cell phone like a
>> separate agent, messing up call distribution etc.
>>
>> I am trying to be as clear as possible, sorry if my questions are cloudy.
>>
>> Basically, I have the queue doing what I want right now, I just want to
>> add the ability to have an agent's cell phone ring as a means of
>> alerting them if they are away from their desk. If they can answer the
>> call and the queue will handle it just as they answered it from their
>> SIP device, that would be a bonus. I know this can all be done, just not
>> sure how to tackle it at the moment.
>>
>> Any guidance would be appreciated, thanks in advance.
>>
>
> Perhaps blend the agent's SIP phone + cell phone together as a local
> channel and then add that local channel to the queue instead of SIP phone +
> cell phone.  Asterisk will see the local channel as one agent rather than
> two.
>
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[asterisk-users] Ringing agents cell as an alert?

[Todd Routhier]

Happy New Year to all!

Asterisk 1.8.x

I have a queue to which I add agent channels like SIP/300 dynamically using
the manager interface. Once logged in, there SIP/300 of course rings when a
call is distributed to them.

How can I also get the agents cell phone to ring without actually adding it
to the queue? I mean id I add something goofy like
SIP/MyProvider/1555444 to the queue, I don't know what will happen at
this point, haven't tested it. Even if it works (asterisk channel state
etc) it will mess with the queue and treat the cell phone like a separate
agent, messing up call distribution etc.

I am trying to be as clear as possible, sorry if my questions are cloudy.

Basically, I have the queue doing what I want right now, I just want to add
the ability to have an agent's cell phone ring as a means of alerting them
if they are away from their desk. If they can answer the call and the queue
will handle it just as they answered it from their SIP device, that would
be a bonus. I know this can all be done, just not sure how to tackle it at
the moment.

Any guidance would be appreciated, thanks in advance.
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Re: [asterisk-users] sendvoicemail=yes not quite working

[Todd Routhier]

On Tue, Dec 20, 2011 at 7:03 PM, M Maki  wrote:

> I have a system working great with the exception of the sendvoicemail=yes
> voicemail.conf option. I can not figure out what I am missing or have
> configured wrong...
>
>
> While in voicemail after selecting 3 for advanced options, then 5 to leave
> a message I am directed to the correct mailbox. But after hearing the
> mailbox number/name announcement I am immediately taken back to my mailbox.
> No option is given to leave a message. I can forward messages, but can't
> leave a message. All other aspects of the voicemail system I have tested
> work great.This is on Debian Squeeze (Asterisk 1.8.8.0 on a x86_64) with
> about 120 phones.
>
> Here is the vebose/debug output of that part of the call.
>
>
> == Using SIP RTP CoS mark 5
>  -- Executing [8000@LocalSets:1] VoiceMailMain("SIP/3323-0499", "")
> in new stack
>  -- Playing 'vm-login.ulaw' (language 'en')
>  -- Playing 'vm-password.ulaw' (language 'en')
>  -- Playing 'vm-youhave.ulaw' (language 'en')
>  -- Playing 'vm-no.ulaw' (language 'en')
>  -- Playing 'vm-messages.ulaw' (language 'en')
>  -- Playing 'vm-leavemsg.ulaw' (language 'en')
>  -- Playing 'vm-starmain.ulaw' (language 'en')
>  -- Playing 'vm-extension.ulaw' (language 'en')
>  -- Playing '/var/spool/asterisk/voicemail/default/3318/greet.slin'
> (language 'en')
> <-THIS IS WHERE IT GOES BACK TO MY VOICEMAIL BOX >
>  == Using SIP RTP CoS mark 5
>  -- Playing 'vm-opts.ulaw' (language 'en')
>  == Spawn extension (LocalSets, 8000, 1) exited non-zero on
> 'SIP/3323-0499'
>
>
> Thanks!
>
> Mike
>
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Pretty sure you need this in your voicemail.conf file:

sendvoicemail=yes
sendvoicemailThis setting takes a *yes* or *no* value. It enables the
"Leave a message" menu option from the Advanced Options menu which allows
the voicemail user to send a message to another voicemail user.
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Re: [asterisk-users] sendvoicemail=yes not quite working

[Todd Routhier]

On Tue, Dec 20, 2011 at 8:53 PM, Todd Routhier  wrote:

>
>
> On Tue, Dec 20, 2011 at 7:03 PM, M Maki  wrote:
>
>> I have a system working great with the exception of the sendvoicemail=yes
>> voicemail.conf option. I can not figure out what I am missing or have
>> configured wrong...
>>
>>
>> While in voicemail after selecting 3 for advanced options, then 5 to
>> leave a message I am directed to the correct mailbox. But after hearing the
>> mailbox number/name announcement I am immediately taken back to my mailbox.
>> No option is given to leave a message. I can forward messages, but can't
>> leave a message. All other aspects of the voicemail system I have tested
>> work great.This is on Debian Squeeze (Asterisk 1.8.8.0 on a x86_64) with
>> about 120 phones.
>>
>> Here is the vebose/debug output of that part of the call.
>>
>>
>> == Using SIP RTP CoS mark 5
>>  -- Executing [8000@LocalSets:1] VoiceMailMain("SIP/3323-0499", "")
>> in new stack
>>  -- Playing 'vm-login.ulaw' (language 'en')
>>  -- Playing 'vm-password.ulaw' (language 'en')
>>  -- Playing 'vm-youhave.ulaw' (language 'en')
>>  -- Playing 'vm-no.ulaw' (language 'en')
>>  -- Playing 'vm-messages.ulaw' (language 'en')
>>  -- Playing 'vm-leavemsg.ulaw' (language 'en')
>>  -- Playing 'vm-starmain.ulaw' (language 'en')
>>  -- Playing 'vm-extension.ulaw' (language 'en')
>>  -- Playing '/var/spool/asterisk/voicemail/default/3318/greet.slin'
>> (language 'en')
>> <-THIS IS WHERE IT GOES BACK TO MY VOICEMAIL BOX >
>>  == Using SIP RTP CoS mark 5
>>  -- Playing 'vm-opts.ulaw' (language 'en')
>>  == Spawn extension (LocalSets, 8000, 1) exited non-zero on
>> 'SIP/3323-0499'
>>
>>
>> Thanks!
>>
>> Mike
>>
>> --
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>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
> Pretty sure you need this in your voicemail.conf file:
>
> sendvoicemail=yes
> sendvoicemailThis setting takes a *yes* or *no* value. It enables the
> "Leave a message" menu option from the Advanced Options menu which allows
> the voicemail user to send a message to another voicemail user.
>
>
Oh, wow.. Nevermind, you started your original post saying you have that
option set.
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Re: [asterisk-users] fromstring in voicemail.conf

[Todd Routhier]

Thanks Matt!

It's a multi-tenant system and basically I just wanted to customize the
look of the From info in the emails.

For everyone's reference, I think I have at least found a direction to go
on this.

I am thinking I will redirect the job of mailing the file to a custom
script at some point which should then allow me to do pretty much whatever
I want.

After searching for a way to have Asterisk send the attachments as mp3
files, I came across lots of interesting stuff that will allow me to do
just about anything with the emails that are sent out.

--Todd


On Fri, Dec 16, 2011 at 1:02 PM, Matthew Jordan  wrote:

>
>
> - Original Message -
> > From: "Matthew Jordan" 
> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" <
> asterisk-users@lists.digium.com>
> > Sent: Friday, December 16, 2011 12:57:43 PM
> > Subject: Re: [asterisk-users] fromstring in voicemail.conf
> >
> >
> > - Original Message -
> >
> > > From: "Todd Routhier" 
> > > To: asterisk-users@lists.digium.com
> > > Sent: Friday, December 16, 2011 11:32:31 AM
> > > Subject: [asterisk-users] fromstring in voicemail.conf
> >
> > > I have attempted to set the fromstring option on a per context
> > > basis
> > > in voicemail.conf but it doesn't seem to work. I would like to
> > > somehow either set this based on context, number dialed into or
> > > some
> > > other way.
> >
> > When you say per context basis, are you talking about multiple
> > contexts for voicemail users?  Or multiple voicemail settings
> > contexts?  app_voicemail only reads global settings from a single
> > context - [general].  Hence, global settings are only defined in
> > that context.  I'm assuming by "fromstring" you mean the setting
> > "email".
>
> Scratch that.  You did mean "fromstring" :-)
>
> That setting is not currently overrideable on a mailbox by mailbox basis.
>  If you'd like that functionality, a patch would have to be written against
> Asterisk trunk.
>
> Sorry for the confusion.
>
> > > Would it be possible to set this option in the general section to a
> > > channel variable, then set the variable as the call is on the way
> > > in?
> >
> > What would that buy you?  app_voicemail does not expect to read its
> > application specific settings from a channel variable.
> >
> > > Any other way?
> >
> > Yes.
> >
> > > OR am I just stuck with one single fromstring for all voice mails
> > > forever?
> >
> > No.  Each mailbox you define in voicemail.conf (or alternatively,
> > users.conf) can override settings that were set in the [global]
> > settings context.  This includes email, emailbody, and emailsubject
> > (at least in versions 1.8 and greater).
> >
> > > Also, does anyone know where I can find a list of options that are
> > > definable at the context level in voicemail.conf? I have searched
> > > for this quite a bit but no luck so far. I now the obvious ones
> > > like
> > > time zone etc..
> >
> > http://ofps.oreilly.com/titles/9780596517342/asterisk-Voicemail.html
> >
> > > Hmm, I am wondering now if I can set it on email mailbox. Instead
> > > of
> > > one time in the context. Trying that now.
> >
> > > Thanks in advance!
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
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Re: [asterisk-users] followme forking/parallel dialing breaks when 1 sip device unreachable

[Todd Routhier]

Perfect, patched my install and it fixed the issue!

Thanks a ton Richard..

--Todd


On Fri, Dec 16, 2011 at 10:17 AM, Richard Mudgett wrote:

> > OK, read all about the patch, thanks for the fix Richard.
> >
> >
> > I would like to apply this patch to my current 1.8.7.1 but I am
> > afraid I don't have a clue how.
>
> https://issues.asterisk.org/jira/browse/ASTERISK-17557
>
> Get the patch by following the reviewboard link in the issue and
> download it from reviewboard.
>
> Apply the patch with the following patch command:
> ~/projects/sa/asterisk/tags/v1.8.7.1$ patch -p3 -i bugASTERISK-17557.patch
>
> Richard
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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[asterisk-users] fromstring in voicemail.conf

[Todd Routhier]

I have attempted to set the fromstring option on a per context basis in
voicemail.conf but it doesn't seem to work. I would like to somehow either
set this based on context, number dialed into or some other way.

Would it be possible to set this option in the general section to a channel
variable, then set the variable as the call is on the way in?

Any other way?

OR am I just stuck with one single fromstring for all voice mails forever?

Also, does anyone know where I can find a list of options that are
definable at the context level in voicemail.conf? I have searched for this
quite a bit but no luck so far. I now the obvious ones like time zone etc..

Hmm, I am wondering now if I can set it on email mailbox. Instead of one
time in the context. Trying that now.

Thanks in advance!
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Re: [asterisk-users] followme forking/parallel dialing breaks when 1 sip device unreachable

[Todd Routhier]

Richard,

 Thanks, I'll give it a whirl.

I upgraded Asterisk to 1.8.8.0 last night and this fixed the original issue
I was having. Now calls to the other extensions continue as normal, even
when one of the SIP extensions is unreachable.

Caller-id is still lost on anything that hits follow me. I will try
applying the patch as you instructed to my 1.8.8.0 install and see what
happens.

Thanks..


On Fri, Dec 16, 2011 at 10:17 AM, Richard Mudgett wrote:

> > OK, read all about the patch, thanks for the fix Richard.
> >
> >
> > I would like to apply this patch to my current 1.8.7.1 but I am
> > afraid I don't have a clue how.
>
> https://issues.asterisk.org/jira/browse/ASTERISK-17557
>
> Get the patch by following the reviewboard link in the issue and
> download it from reviewboard.
>
> Apply the patch with the following patch command:
> ~/projects/sa/asterisk/tags/v1.8.7.1$ patch -p3 -i bugASTERISK-17557.patch
>
> Richard
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
>
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] followme forking/parallel dialing breaks when 1 sip device unreachable

[Todd Routhier]

OK, read all about the patch, thanks for the fix Richard.

I would like to apply this patch to my current 1.8.7.1 but I am afraid I
don't have a clue how.

Is this just a case of getting a copy of app_followme.c and replacing it on
my current Asterisk install? If not, do I have to grab a new Asterisk
version and recompile everything and start over? I know I can save my
configs but I was hoping for a simple fix without having to recompile
Asterisk from source etc.

Thanks for any help.

--Todd


On Thu, Dec 15, 2011 at 9:07 PM, Todd Routhier  wrote:

> No, I get no error in the CLI at all, just shows that the followme is
> being executed then dumps straight to Vmail which is defined in my dialplan
> on the next line after calling the followme.
>
> I checked out the link and it also shows problems with callerid not
> passing, this is also a problem for me and that was what I was going to
> tackle next.
>
> I will checkout the patch, I have never applied a patch though, only done
> fresh installs. So, I will need to figure that out.
>
> I am running Asterisk 1.8.7.1 to be more specific.
>
> Thanks Richard.
>
>
> On Thu, Dec 15, 2011 at 8:56 PM, Richard Mudgett wrote:
>
>> > *
>> > Summary:
>> >
>> >
>> > I need to be able to ring multiple numbers in followme.conf at the
>> > same time, even if one of the SIP extensions is unreachable.
>> > This works in 1.4.8 but not in 1.8, just barfs and sends to voice
>> > mail instead of ringing the other 2 extensions on the same line in
>> > the followme.conf
>> >
>> >
>> > See more details below.
>> > *
>> >
>> > I decided to mess around with followme and it actually suits my needs
>> > quit well. I want to know what number the caller called into when my
>> > cell phone rings, then decide if I want to answer it by pressing 1
>> > or not. Also helps with making sure voicemail is only left on my
>> > Asterisk voicemail instead of my cell phone voice mail.
>> >
>> >
>> > So, I set up followme on Asterisk 1.4.8 something like this and it
>> > worked great:
>> >
>> >
>> > from my followme.conf:
>> > number=>207&206&5554441212,28
>> >
>> >
>> > Problem is after moving this same config to my new 1.8 box the call
>> > fails and goes to voice mail if either of the two sip extensions are
>> > unreachable.
>> >
>> >
>> > So, let me explain further...
>> >
>> >
>> > If both SIP/207 and SIP/206 are up and running and accessible to
>> > receive the call then all goes well, if one of them is down for some
>> > reason then none of the 3 extensions ring and it just goes to voice
>> > mail. This stinks because I lose all calls to voice mail if for
>> > example my Internet connection goes down at home (207). Wouldn't
>> > this be the time you really want your other phones to ring?
>> >
>> >
>> > I thought about doing something like this:
>> > number=>207&206&5554441212,28
>> > number=>207&5554441212,28
>> >
>> >
>> > In case say 206 fails but when 206 is up, they will be on hold for
>> > almost a minute before going to voice mail if I don't answer.
>> >
>> >
>> > I know there are other solutions outside followme but this worked in
>> > 1.4.8 and I have to think it should work in 1.8. Just not sure what
>> > I am missing.
>> >
>> >
>> > Thanks in advance for any help.
>>
>> This may work now after I fixed this issue last week on SVN v1.8:
>> https://issues.asterisk.org/jira/browse/ASTERISK-17557
>>
>> Do you get "Extension '%s@%s' doesn't exist\n" error messages?
>>
>> Richard
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>   http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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Re: [asterisk-users] followme forking/parallel dialing breaks when 1 sip device unreachable

[Todd Routhier]

No, I get no error in the CLI at all, just shows that the followme is being
executed then dumps straight to Vmail which is defined in my dialplan on
the next line after calling the followme.

I checked out the link and it also shows problems with callerid not
passing, this is also a problem for me and that was what I was going to
tackle next.

I will checkout the patch, I have never applied a patch though, only done
fresh installs. So, I will need to figure that out.

I am running Asterisk 1.8.7.1 to be more specific.

Thanks Richard.

On Thu, Dec 15, 2011 at 8:56 PM, Richard Mudgett wrote:

> > *
> > Summary:
> >
> >
> > I need to be able to ring multiple numbers in followme.conf at the
> > same time, even if one of the SIP extensions is unreachable.
> > This works in 1.4.8 but not in 1.8, just barfs and sends to voice
> > mail instead of ringing the other 2 extensions on the same line in
> > the followme.conf
> >
> >
> > See more details below.
> > *
> >
> > I decided to mess around with followme and it actually suits my needs
> > quit well. I want to know what number the caller called into when my
> > cell phone rings, then decide if I want to answer it by pressing 1
> > or not. Also helps with making sure voicemail is only left on my
> > Asterisk voicemail instead of my cell phone voice mail.
> >
> >
> > So, I set up followme on Asterisk 1.4.8 something like this and it
> > worked great:
> >
> >
> > from my followme.conf:
> > number=>207&206&5554441212,28
> >
> >
> > Problem is after moving this same config to my new 1.8 box the call
> > fails and goes to voice mail if either of the two sip extensions are
> > unreachable.
> >
> >
> > So, let me explain further...
> >
> >
> > If both SIP/207 and SIP/206 are up and running and accessible to
> > receive the call then all goes well, if one of them is down for some
> > reason then none of the 3 extensions ring and it just goes to voice
> > mail. This stinks because I lose all calls to voice mail if for
> > example my Internet connection goes down at home (207). Wouldn't
> > this be the time you really want your other phones to ring?
> >
> >
> > I thought about doing something like this:
> > number=>207&206&5554441212,28
> > number=>207&5554441212,28
> >
> >
> > In case say 206 fails but when 206 is up, they will be on hold for
> > almost a minute before going to voice mail if I don't answer.
> >
> >
> > I know there are other solutions outside followme but this worked in
> > 1.4.8 and I have to think it should work in 1.8. Just not sure what
> > I am missing.
> >
> >
> > Thanks in advance for any help.
>
> This may work now after I fixed this issue last week on SVN v1.8:
> https://issues.asterisk.org/jira/browse/ASTERISK-17557
>
> Do you get "Extension '%s@%s' doesn't exist\n" error messages?
>
> Richard
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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[asterisk-users] followme forking/parallel dialing breaks when 1 sip device unreachable

[Todd Routhier]

*
Summary:

I need to be able to ring multiple numbers in followme.conf at the same
time, even if one of the SIP extensions is unreachable.
This works in 1.4.8 but not in 1.8, just barfs and sends to voice mail
instead of ringing the other 2 extensions on the same line in the
followme.conf

See more details below.
*

I decided to mess around with followme and it actually suits my needs quit
well. I want to know what number the caller called into when my cell phone
rings, then decide if I want to answer it by pressing 1 or not. Also helps
with making sure voicemail is only left on my Asterisk voicemail instead of
my cell phone voice mail.

So, I set up followme on Asterisk 1.4.8 something like this and it worked
great:

from my followme.conf:
number=>207&206&5554441212,28

Problem is after moving this same config to my new 1.8 box the call fails
and goes to voice mail if either of the two sip extensions are unreachable.

So, let me explain further...

If both SIP/207 and SIP/206 are up and running and accessible to receive
the call then all goes well, if one of them is down for some reason then
none of the 3 extensions ring and it just goes to voice mail. This stinks
because I lose all calls to voice mail if for example my Internet
connection goes down at home (207). Wouldn't this be the time you really
want your other phones to ring?

I thought about doing something like this:
number=>207&206&5554441212,28
number=>207&5554441212,28

In case say 206 fails but when 206 is up, they will be on hold for almost a
minute before going to voice mail if I don't answer.

I know there are other solutions outside followme but this worked in 1.4.8
and I have to think it should work in 1.8. Just not sure what I am missing.

Thanks in advance for any help.

--Todd
--
_
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Re: [asterisk-users] Recommendations

[Todd Routhier]

OK, so yeah.. Lack of sleep.

I think my brain just kept seeing 4.2 instead of 42.

Thanks for the clarification.

Man do I feel like a bozo now.

--Todd


On Mon, Nov 28, 2011 at 11:15 PM, Douglas Mortensen  wrote:

> 8 comes before 42.
>
> ** **
>
> -
>
> Doug Mortensen
>
> Network Consultant
>
> Impala Networks
>
> P: 505.327.7300****
>
> .
>
> ** **
>
> *From:* Todd Routhier [mailto:fonema...@gmail.com]
> *Sent:* Monday, November 28, 2011 3:55 PM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Recommendations
>
> ** **
>
> Danny,
>
> ** **
>
>  Thanks for your help.
>
> ** **
>
> OK, guess I got confused by the fact that 8 comes after 4.
>
> ** **
>
> Maybe I missed something in the version number or maybe I am just, well,
> lacking sleep.
>
> ** **
>
> When did the version numbers start going backwards?
>
> ** **
>
> By the way, I hope I am not top posting but Gmail is a little whacky,
> trying to get use to it.
>
> ** **
>
> --Todd
>
> ** **
>
> On Mon, Nov 28, 2011 at 4:50 PM, Danny Nicholas  wrote:
> 
>
> 1.4.42 is newer than 1.4.8.
>
>  
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Todd Routhier
> *Sent:* Monday, November 28, 2011 4:49 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Recommendations
>
>  
>
> Hmmm, so then nothing really major in regards to Queues has changed since
> the version I am using 1.4.8?
>
> You are actually recommending an older version than I am using?
>
>  
>
> In regards to CentOS 5 then say 5.3 should be fine?
>
>  
>
> I usually build from source but I am thinking about just installing CentOS
> 5.3 and installing it from YUM but I will have to see what version that
> installs.
>
>  
>
> Any good or bad things to say about using AsteriskNow?
>
>  
>
>  
>
>  
>
> On Mon, Nov 28, 2011 at 3:16 PM, Danny Nicholas  wrote:
> 
>
> App_read hangs up the call if an error is encountered in 1.4 (really
> annoying as I don’t want my user to have to enter 2 or more times).  This
> behavior was resolved in 10.0.
>
>  
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *john Millican
> *Sent:* Monday, November 28, 2011 3:11 PM
> *To:* asterisk-users@lists.digium.com
> *Subject:* Re: [asterisk-users] Recommendations
>
>  
>
>
>
> On 11/28/2011 3:35 PM, Danny Nicholas wrote: 
>
> If you put a gun to my head I would say to stay with Centos 5 and either
> 1.4.42 or 10.0.0-rc2.  10.0.0-rc2 removes a “feature” that was killing me
> in 1.4, but if you aren’t doing IVR stuff, you can stay with what you
> know.  Another thing to consider though; 1.4.8 is prior to the
> Zaptel-to-Dahdi conversion so that might cause you some “joy”.
>
>  
>
> *From:* asterisk-users-boun...@lists.digium.com [
> mailto:asterisk-users-boun...@lists.digium.com]
> *On Behalf Of *Todd Routhier
> *Sent:* Monday, November 28, 2011 2:31 PM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] Recommendations
>
>  
>
> I am currently running Asterisk 1.4.8 and have been for quite a while, it
> has served me well.
>
>  
>
> Getting ready to build a new box to replace the existing installation of
> Asterisk.
>
>  
>
> My primary use of the Asterisk box is run queues. I am sure the queue
> features and functionality have been updated, expanded since 1.4.8 and I am
> wondering what version of Ast you guys would recommend. Looking for the
> best version in terms of queue features, functionality.
>
>  
>
> Also, an OS recommendation would be great. Been running on CentOS forever
> and no reason to want to change. Just looking for the best version of
> CentOS to run the best/stable version of Asterisk on.
>
>  
>
> To be clear:
>
>  
>
> Recommended:
>
>  
>
> Asterisk Version:
>
> OS & Version:
>
>  
>
> I am even thinking about using AsteriskNow, don't need the FreePBX but I
> have worked with it before and it used to be possible to still do custom
> stuff and co-exist with FreePBX. I like having FreePBX available for the
> simple stuff so it's not a bad thing if it's there. Does it have an
> intergrated web serv

Re: [asterisk-users] Recommendations

[Todd Routhier]

Danny,

 Thanks for your help.

OK, guess I got confused by the fact that 8 comes after 4.

Maybe I missed something in the version number or maybe I am just, well,
lacking sleep.

When did the version numbers start going backwards?

By the way, I hope I am not top posting but Gmail is a little whacky,
trying to get use to it.

--Todd


On Mon, Nov 28, 2011 at 4:50 PM, Danny Nicholas  wrote:

> 1.4.42 is newer than 1.4.8.
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Todd Routhier
> *Sent:* Monday, November 28, 2011 4:49 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Recommendations
>
> ** **
>
> Hmmm, so then nothing really major in regards to Queues has changed since
> the version I am using 1.4.8?
>
> You are actually recommending an older version than I am using?
>
> ** **
>
> In regards to CentOS 5 then say 5.3 should be fine?
>
> ** **
>
> I usually build from source but I am thinking about just installing CentOS
> 5.3 and installing it from YUM but I will have to see what version that
> installs.
>
> ** **
>
> Any good or bad things to say about using AsteriskNow?
>
> ** **
>
> ** **
>
> ** **
>
> On Mon, Nov 28, 2011 at 3:16 PM, Danny Nicholas  wrote:
> 
>
> App_read hangs up the call if an error is encountered in 1.4 (really
> annoying as I don’t want my user to have to enter 2 or more times).  This
> behavior was resolved in 10.0.
>
>  
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *john Millican
> *Sent:* Monday, November 28, 2011 3:11 PM
> *To:* asterisk-users@lists.digium.com
> *Subject:* Re: [asterisk-users] Recommendations
>
>  
>
>
>
> On 11/28/2011 3:35 PM, Danny Nicholas wrote: 
>
> If you put a gun to my head I would say to stay with Centos 5 and either
> 1.4.42 or 10.0.0-rc2.  10.0.0-rc2 removes a “feature” that was killing me
> in 1.4, but if you aren’t doing IVR stuff, you can stay with what you
> know.  Another thing to consider though; 1.4.8 is prior to the
> Zaptel-to-Dahdi conversion so that might cause you some “joy”.
>
>  
>
> *From:* asterisk-users-boun...@lists.digium.com [
> mailto:asterisk-users-boun...@lists.digium.com]
> *On Behalf Of *Todd Routhier
> *Sent:* Monday, November 28, 2011 2:31 PM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] Recommendations
>
>  
>
> I am currently running Asterisk 1.4.8 and have been for quite a while, it
> has served me well.
>
>  
>
> Getting ready to build a new box to replace the existing installation of
> Asterisk.
>
>  
>
> My primary use of the Asterisk box is run queues. I am sure the queue
> features and functionality have been updated, expanded since 1.4.8 and I am
> wondering what version of Ast you guys would recommend. Looking for the
> best version in terms of queue features, functionality.
>
>  
>
> Also, an OS recommendation would be great. Been running on CentOS forever
> and no reason to want to change. Just looking for the best version of
> CentOS to run the best/stable version of Asterisk on.
>
>  
>
> To be clear:
>
>  
>
> Recommended:
>
>  
>
> Asterisk Version:
>
> OS & Version:
>
>  
>
> I am even thinking about using AsteriskNow, don't need the FreePBX but I
> have worked with it before and it used to be possible to still do custom
> stuff and co-exist with FreePBX. I like having FreePBX available for the
> simple stuff so it's not a bad thing if it's there. Does it have an
> intergrated web server that I could run a lightweight control panel on?
> That would be another plus.
>
>  
>
> Thanks in advance for any help, been out of touch for a while. I will be
> doing my research and lots of reading over the next few days but thought it
> couldn't hurt to see what the general consensus is on these topics.
>
>  
>
>  
>
>  
>
> Danny,
> Can you expand on what "feature" 10.0.0-rc2 removed that was causing you
> problems with IVR?  I am starting to undertake some major IVR scripting so
> am rather curious.
> Thanks,
> JohnM
>
>
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> _
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>   http://www.asteri

Re: [asterisk-users] Recommendations

[Todd Routhier]

Hmmm, so then nothing really major in regards to Queues has changed since
the version I am using 1.4.8?

You are actually recommending an older version than I am using?

In regards to CentOS 5 then say 5.3 should be fine?

I usually build from source but I am thinking about just installing CentOS
5.3 and installing it from YUM but I will have to see what version that
installs.

Any good or bad things to say about using AsteriskNow?



On Mon, Nov 28, 2011 at 3:16 PM, Danny Nicholas  wrote:

> App_read hangs up the call if an error is encountered in 1.4 (really
> annoying as I don’t want my user to have to enter 2 or more times).  This
> behavior was resolved in 10.0.
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *john Millican
> *Sent:* Monday, November 28, 2011 3:11 PM
> *To:* asterisk-users@lists.digium.com
> *Subject:* Re: [asterisk-users] Recommendations
>
> ** **
>
>
>
> On 11/28/2011 3:35 PM, Danny Nicholas wrote: 
>
> If you put a gun to my head I would say to stay with Centos 5 and either
> 1.4.42 or 10.0.0-rc2.  10.0.0-rc2 removes a “feature” that was killing me
> in 1.4, but if you aren’t doing IVR stuff, you can stay with what you
> know.  Another thing to consider though; 1.4.8 is prior to the
> Zaptel-to-Dahdi conversion so that might cause you some “joy”.
>
>  
>
> *From:* asterisk-users-boun...@lists.digium.com [
> mailto:asterisk-users-boun...@lists.digium.com]
> *On Behalf Of *Todd Routhier
> *Sent:* Monday, November 28, 2011 2:31 PM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] Recommendations
>
>  
>
> I am currently running Asterisk 1.4.8 and have been for quite a while, it
> has served me well.
>
>  
>
> Getting ready to build a new box to replace the existing installation of
> Asterisk.
>
>  
>
> My primary use of the Asterisk box is run queues. I am sure the queue
> features and functionality have been updated, expanded since 1.4.8 and I am
> wondering what version of Ast you guys would recommend. Looking for the
> best version in terms of queue features, functionality.
>
>  
>
> Also, an OS recommendation would be great. Been running on CentOS forever
> and no reason to want to change. Just looking for the best version of
> CentOS to run the best/stable version of Asterisk on.
>
>  
>
> To be clear:
>
>  
>
> Recommended:
>
>  
>
> Asterisk Version:
>
> OS & Version:
>
>  
>
> I am even thinking about using AsteriskNow, don't need the FreePBX but I
> have worked with it before and it used to be possible to still do custom
> stuff and co-exist with FreePBX. I like having FreePBX available for the
> simple stuff so it's not a bad thing if it's there. Does it have an
> intergrated web server that I could run a lightweight control panel on?
> That would be another plus.
>
>  
>
> Thanks in advance for any help, been out of touch for a while. I will be
> doing my research and lots of reading over the next few days but thought it
> couldn't hurt to see what the general consensus is on these topics.
>
>  
>
>  
>
> ** **
>
> Danny,
> Can you expand on what "feature" 10.0.0-rc2 removed that was causing you
> problems with IVR?  I am starting to undertake some major IVR scripting so
> am rather curious.
> Thanks,
> JohnM
>
> --
> _
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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[asterisk-users] Recommendations

[Todd Routhier]

I am currently running Asterisk 1.4.8 and have been for quite a while, it
has served me well.

Getting ready to build a new box to replace the existing installation of
Asterisk.

My primary use of the Asterisk box is run queues. I am sure the queue
features and functionality have been updated, expanded since 1.4.8 and I am
wondering what version of Ast you guys would recommend. Looking for the
best version in terms of queue features, functionality.

Also, an OS recommendation would be great. Been running on CentOS forever
and no reason to want to change. Just looking for the best version of
CentOS to run the best/stable version of Asterisk on.

To be clear:

Recommended:

Asterisk Version:
OS & Version:

I am even thinking about using AsteriskNow, don't need the FreePBX but I
have worked with it before and it used to be possible to still do custom
stuff and co-exist with FreePBX. I like having FreePBX available for the
simple stuff so it's not a bad thing if it's there. Does it have an
intergrated web server that I could run a lightweight control panel on?
That would be another plus.

Thanks in advance for any help, been out of touch for a while. I will be
doing my research and lots of reading over the next few days but thought it
couldn't hurt to see what the general consensus is on these topics.
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[asterisk-users] Follow me unreachable message default

[Todd Routhier]

Hey folks, been out of the loop for a while and need to make a few changes
to my Ast box.

I have been digging around trying to find an answer on the list archives,
the wiki, google etc. but not joy.

I have using Asterisk 1.4.8 and have recently added the FollowMe feature.

Basically, I have what I need working but when nobody answers on any of the
follow me numbers the caller hears the "The party you are calling is
unreachable" message, then goes to my vmail.

This is OK but I don't like the unreachable message and I want that
skipped/removed from my config.

I read in the docs that you are suppose to use the "n" option to enable
this which I have NOT done anywhere in followme.conf or when I call the
followme app from my dial plan. For some reason it's defaulting to this
option. Is there a way to turn the playing of the unreachable message off,
short of changing the message it plays to something more desirable?

n - Playback the unreachable status message if we've run out of steps to
reach the

Thanks,
 Todd
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Re: [asterisk-users] Asterisk 1.8 -- queue not recognizing that agent is busy

[Todd Fulton]

Hi,


Setting "callcounter = yes" in sip.conf definitely made a difference!  Now,
the agent state goes to Ringing.  However, after the call is ended, the
state does not change out of Ringing, and new calls do not get routed to the
agent any longer.  Below is after the 1st call has ended, and while a new
call is in the queue waiting:


QUEUE_3 has 1 calls (max unlimited) in 'wrandom' strategy (2s holdtime, 2s
talktime), W:0, C:1, A:0, SL:0.0% within 0s
   Members:
  TODD (SIP/14155331...@jnctn) (dynamic) (Ringing) has taken 1 calls
(last was 38 secs ago)
   Callers:
  1. SIP/vitel-inbound-0002 (wait: 0:16, prio: 0)


I tried the state_interface setting, no go.


Any ideas?



Todd

  Original Message 
Subject: Re: [asterisk-users] Asterisk 1.8 -- queue not recognizing
that agent is busy
From: Carlos Chavez 
Date: Wed, November 10, 2010 11:36 am
To: Asterisk Users Mailing List - Non-Commercial Discussion


On Wed, 2010-11-10 at 11:09 -0800, Todd Fulton wrote:
> Hi All,
>
> I've got a realtime queue in place (strategy is "wrandom"), and have
> added a member dynamically via "queue add member ". My agent shows in
> the queue, but when he gets the call is not recognized as "In Use".
> Here is the output from "queue show" prior to the call:
>
> *CLI> queue show
> QUEUE_3 has 0 calls (max unlimited) in 'wrandom' strategy (0s
> holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s
> Members:
> Todd1 (SIP/14153436...@jnctn) (dynamic) (Not in use) has taken
> no calls yet
> No Callers
>
> Here is the output when actually connected to the inbound caller:
>
> *CLI> queue show
> QUEUE_3 has 0 calls (max unlimited) in 'wrandom' strategy (1s
> holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s
> Members:
> Todd1 (SIP/14153436...@jnctn) (dynamic) (Not in use) has taken
> no calls yet
> No Callers
>
> If I make a second inbound call while the agent is still connected to
> the first, the second call is also routed to the agent. Queue doesn't
> appear know the agent is already busy.
>
> My question: doesn't the Queue application keep track of the agent's
> interface status

Do you have "callcounter = yes" in your sip.conf?

-- 
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Director de Tecnología
+52-55-91169161 ext 2001
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[asterisk-users] Asterisk 1.8 -- queue not recognizing that agent is busy

[Todd Fulton]

Hi All,

I've got a realtime queue in place (strategy is "wrandom"), and have
added a member dynamically via "queue add member ".  My agent shows in
the queue, but when he gets the call is not recognized as "In Use".
Here is the output from "queue show" prior to the call:

*CLI> queue show
QUEUE_3 has 0 calls (max unlimited) in 'wrandom' strategy (0s
holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s
   Members:
  Todd1 (SIP/14153436...@jnctn) (dynamic) (Not in use) has taken
no calls yet
   No Callers

Here is the output when actually connected to the inbound caller:

*CLI> queue show
QUEUE_3 has 0 calls (max unlimited) in 'wrandom' strategy (1s
holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s
   Members:
  Todd1 (SIP/14153436...@jnctn) (dynamic) (Not in use) has taken
no calls yet
   No Callers

If I make a second inbound call while the agent is still connected to
the first, the second call is also routed to the agent.  Queue doesn't
appear know the agent is already busy.

My question:  doesn't the Queue application keep track of the agent's
interface status?



Todd

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[asterisk-users] AstriCon update - less than two weeks!

[John Todd]

[ text with links can be found on 
http://blogs.digium.com/2010/10/14/astricon-update/ 
  ]

AstriCon is less than two weeks away! If you haven’t booked your  
flight to Washington DC, now’s your chance! The main hotel (the  
Gaylord) is pretty booked, but that’s OK – there are still rooms a few  
hundred feet away at some of the hotels around the complex (Aloft,  
Wyndham, Hampton Inn, Residence Inn) and there are more hotels within  
a short drive/cab of the venue.
Speakers at AstriCon

We’ve got some great last-minute speakers to announce – I’m pleased to  
say that Ruben Sousa will be giving a talk on one of the largest open- 
source Asterisk installations in the world (100,000 users, 184  
servers) which is arrayed across the University system in Portugal. We  
have a really solid line-up this year of talks focused on security and  
scalability from Kevin Lynn, Sandro Gauci, the Great Olle Johansson,  
and more! Many of the most active community developers, integrators,  
and speakers will be on hand, along with some very interesting  
announcements from Digium including the yearly roadmap and status  
update from the Digium engineering group – don’t miss out on hearing  
what’s new and what’s coming up! With the huge number of features that  
have been added to 1.8, it’s possible that you’ll learn from someone  
at the show how the newest release of Asterisk can benefit your  
organization in a way you never expected.

Win an Apple iPAD at the AstriCon Ringer Rodeo!

“Saddle up pardners!” and prepare to reap the benefit of being the  
fastest Asterisk geek in the West… make that “East”. Well, the fastest  
Asterisk geek at AstriCon, anyways. We’ve devised a stunningly simple  
contest (it’s almost too simple) to give away an Apple iPad – and with  
only a low number of opportunities to compete (during the show party),  
this could be the easiest iPad you win this millennium. Second and  
third prizes are a Polycom IP-650 deskphone and aPanasonic KX-TGP500  
DECT wireless phone.

The contest will demand the ability to hook up two SIP phones and an  
IAX ‘trunk’, in addition to a small amount of dialplan programming.  
You’ll be given all the details you need and you will not have to know  
how to set up the actual phones – we’ve done that part for you.

Just like last year (when we gave away an unlocked HTC Hero Android  
phone) there will be a number of timed rounds with the winning time in  
each round going on to the leader board – the dude (or dudette) with  
the fastest time on that board at the end of the party will be  
presented with said iPad at the end of conference session in addition  
to being admired and envied in equal measure by the gathered crowd!  
David Duffett will be the chief rodeo wrangler – at the all-conference  
party on Wednesday night, look for the man in pinstriped jacket and  
cane.

Etc. etc.

I’ve been told there is a Water Taxi from the hotel dock to old Town  
Alexandria. This is a great place to have dinner, wander around, and  
see some of the sights of the DC area. And a boat taxi is always fun!  
A mile or two up the river from Alexandria, there is also lots to do  
in Washington, DC – it’s a great time of year to be there when it’s  
not too hot, and not too cold.
Developers: there is the AstriDevCon on Friday from around 8:00 to  
5:00, which will focus on hard-core code development and discussion of  
particular issues in the codebase. If you speak C and find yourself  
typing “make config” in your head when you meet someone new, this is  
probably where you’ll find some interesting discussion. Sign up here  
if you haven’t already. We do ask this to be a developer-only session,  
so please be familiar with the code and the issue tracker if you plan  
to attend. Also, don’t forget there is the chance to take the dCAP  
test at AstriCon – kill two birds with one stone! See the AstriCon  
website for more details.

Just two more weeks! See everyone there.

JT


---
John Todd   email:jt...@digium.com
Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW -  Huntsville AL 35806  -   USA
direct: +1-256-428-6083 http://www.digium.com/





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Re: [asterisk-users] help with dialplan

[Todd Reese]

 Interesting things going on herel.

After your suggestions, Steve.  I reran the dialplan show 
16789542...@remote command with the below results.



Phone calls are geting the 404 error and the NOTICE on the console.
[Aug 31 11:06:46] NOTICE[11884]: chan_sip.c:20161 handle_request_invite: 
Call from '150' to extension '16789542133' rejected because extension 
not found in context 'remote'.



asterisk*CLI> dialplan show 16789542...@remote
[ Included context 'dialout1' created by 'pbx_config' ]
  '_1NXXNXX' => -2. Dial(SIP/v6787500600/${EXTEN},40,Ttr) 
[pbx_config]


[ Included context 'dialout1' created by 'pbx_config' ]
  '_1NXXNXX' => -2. Dial(SIP/v6787500600/${EXTEN},40,Ttr) 
[pbx_config]


[ Included context 'dialout2' created by 'pbx_config' ]
  '_1NXXNXX' => -2. Dial(SIP/voip.comACA/${EXTEN},40,Ttr) 
[pbx_config]


[ Included context 'dialout3' created by 'pbx_config' ]
  '_1NXXNXX' => -2. Dial(SIP/v6783747049/${EXTEN},40,Ttr) 
[pbx_config]


[ Included context 'dialout1' created by 'pbx_config' ]
  '_1NXXNXX' => -2. Dial(SIP/v6787500600/${EXTEN},40,Ttr) 
[pbx_config]


[ Included context 'dialout2' created by 'pbx_config' ]
  '_1NXXNXX' => -2. Dial(SIP/voip.comACA/${EXTEN},40,Ttr) 
[pbx_config]


[ Included context 'dialout3' created by 'pbx_config' ]
  '_1NXXNXX' => -2. Dial(SIP/v6783747049/${EXTEN},40,Ttr) 
[pbx_config]


-= 7 extensions (7 priorities) in 7 contexts. =-
[Aug 31 11:03:38] WARNING[11903]: pbx.c:5741 show_dialplan_helper: 
Avoiding circular include of from-internal within remote



On 8/31/2010 10:49 AM, Steve Murphy wrote:

Todd--

There is probably some nifty anti-infinite-recursion code in the 
extensions.conf parser,
to keep asterisk from going into infinite loops trying to descend into 
the right context.


In your dialplan, [remote] includes dialout1, dialout2, dialout3, and 
each of those

include remote.

Straighten out that mess and maybe things might work. Just a guess, 
but worth a try!


murf


On Tue, Aug 31, 2010 at 8:25 AM, Todd Reese <mailto:trees...@gmail.com>> wrote:


 From extensions.conf

[remote]
include => from-internal
include => dialout1
include => dialout2
include => dialout3
include => intercom
exten => 150,1,Macro(oneline,${EXTERNPHONE0})

[dialout1]
include => from-internal
include => 411
include => remote
exten => 911,1,Goto(nineoneone,s,1)
exten => _1NXXNXX,n,Dial(SIP/v6787500600/${EXTEN},40,Ttr)
[dialout2]
include => from-internal
include => 411
include => remote
exten => 911,1,Goto(nineoneone,s,1)
exten => _1NXXNXX,n,Dial(SIP/voip.comACA/${EXTEN},40,Ttr)
[dialout3]
include => from-internal
include => 411
include => remote
exten => 911,1,Goto(nineoneone,s,1)
exten => _1NXXNXX,n,Dial(SIP/v6783747049/${EXTEN},40,Ttr)





On 8/31/2010 9:58 AM, Paul Belanger wrote:
> On Tue, Aug 31, 2010 at 9:16 AM, Todd Reesemailto:trees...@gmail.com>>  wrote:
>>   Here's the updated debug log.
>>
> [Aug 30 15:27:41] NOTICE[11568] chan_sip.c: Call from '150' to
> extension '6789542133' rejected because extension not found in
context
> 'remote'.
>
> So, again, a context problem.  You can confirm by entering:
>
> *CLI>  dialplan show 6789542...@remote
>
>


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--
Steve Murphy
ParseTree Corp



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Re: [asterisk-users] help with dialplan

[Todd Reese]

  I had already check on this.   Thanks for the info, though.


On 8/31/2010 10:36 AM, Danny Nicholas wrote:
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Todd Reese
> Subject: Re: [asterisk-users] help with dialplan
>
>>   asterisk*CLI>  dialplan show 6789542...@remote
>> There is no existence of 'remote' context
>> Command 'dialplan show 6789542...@remote' failed.
>> asterisk*CLI>
>
>>> On 8/31/2010 9:58 AM, Paul Belanger wrote:
>>>> dialplan show 6789542...@remote
> Ok. I'm a late joiner to this thread.  Reading the "original post" I see
> that you are trying to do an external SIP dial to 678-954-2133.  These
> questions:
> 1. Does the dial string need to be 1xxx instead of xxx (presumably local 10
> digit dialing)? If yes, change
> exten =>  _NXXNXX,n,Dial(SIP/v6781234567/${EXTEN},40,Ttr)
> to
> exten =>  _NXXNXX,n,Dial(SIP/v6781234567/1${EXTEN},40,Ttr)
> 2. voipdialACA and v6781234567 are registered trunks with credentials?
>
> Hope this helps.
>
>
>
>


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Re: [asterisk-users] help with dialplan

[Todd Reese]

  From extensions.conf

[remote]
include => from-internal
include => dialout1
include => dialout2
include => dialout3
include => intercom
exten => 150,1,Macro(oneline,${EXTERNPHONE0})

[dialout1]
include => from-internal
include => 411
include => remote
exten => 911,1,Goto(nineoneone,s,1)
exten => _1NXXNXX,n,Dial(SIP/v6787500600/${EXTEN},40,Ttr)
[dialout2]
include => from-internal
include => 411
include => remote
exten => 911,1,Goto(nineoneone,s,1)
exten => _1NXXNXX,n,Dial(SIP/voip.comACA/${EXTEN},40,Ttr)
[dialout3]
include => from-internal
include => 411
include => remote
exten => 911,1,Goto(nineoneone,s,1)
exten => _1NXXNXX,n,Dial(SIP/v6783747049/${EXTEN},40,Ttr)





On 8/31/2010 9:58 AM, Paul Belanger wrote:
> On Tue, Aug 31, 2010 at 9:16 AM, Todd Reese  wrote:
>>   Here's the updated debug log.
>>
> [Aug 30 15:27:41] NOTICE[11568] chan_sip.c: Call from '150' to
> extension '6789542133' rejected because extension not found in context
> 'remote'.
>
> So, again, a context problem.  You can confirm by entering:
>
> *CLI>  dialplan show 6789542...@remote
>
>


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Re: [asterisk-users] help with dialplan

[Todd Reese]

  asterisk*CLI> dialplan show 6789542...@remote
There is no existence of 'remote' context
Command 'dialplan show 6789542...@remote' failed.
asterisk*CLI>


On 8/31/2010 9:58 AM, Paul Belanger wrote:
> dialplan show 6789542...@remote


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Re: [asterisk-users] help with dialplan

[Todd Reese]

  Here's the updated debug log.

http:/www.computerworkx.net/client/Document.txt



On 8/30/2010 2:55 PM, Paul Belanger wrote:
> On Mon, Aug 30, 2010 at 2:48 PM, Todd Reese  wrote:
>> Thanks for pointing out the misspelling.  I've corrected that and still no
>> luck.
>>
> Create a new debug log with your recent changes, re-attach it the list.
>


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Re: [asterisk-users] help with dialplan

[Todd Reese]

 Thanks for pointing out the misspelling.  I've corrected that and 
still no luck.


On 8/30/2010 2:33 PM, Alex Bell wrote:

possibly check you spelling:  [from-interal] -> [dialout1]
include => from-internal

??

On Mon, Aug 30, 2010 at 10:14 AM, Todd Reese <mailto:trees...@gmail.com>> wrote:


 Hi all,

I've been have problems with getting this system on line and would
like
to acquire some help with the extensions.conf.

My current problem is that the phones won't dialout.on the VOIP lines
listed as dialout1, dialout2, dialout3. This version of asterisk is
1.6.2.11.  Below is the extensions.conf file.


[globals]



QPHONE0=SIP/10
QPHONE1=SIP/11
QPHONE2=SIP/12
QPHONE3=SIP/13
QPHONE4=SIP/14
QPHONE5=SIP/15
QPHONE6=SIP/16
QPHONE7=SIP/17

ACAPHONE0=SIP/20
ACAPHONE1=SIP/21
ACAPHONE2=SIP/22
ACAPHONE3=SIP/23
ACAPHONE4=SIP/24
ACAPHONE5=SIP/25
ACAPHONE6=SIP/26
ACAPHONE7=SIP/27

GMNETPHONE0=SIP/30
GMNETPHONE1=SIP/31
GMNETPHONE2=SIP/32
GMNETPHONE3=SIP/33
GMNETPHONE4=SIP/34
GMNETPHONE5=SIP/35
GMNETPHONE6=SIP/36
GMNETPHONE7=SIP/37

EXTERNPHONE0=SIP/150

CPHONE1=SIP/1678000
CPHONE2=SIP/177

EMERGENCY=0
EMERGENCY_TRUNK=DAHDI/G1
; Change this for production use:
EMERGENCY_NUM=6789542133


[from-pstn]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)




[from-pstn1]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn2]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn3]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn4]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming3,s,1)

[from-pstn5]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming2,s,1)

[from-pstn6]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn7]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn8]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)


[incoming1]
include => from-internal
include => parkedcalls
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Set(CHANNEL(musicclass)=QCI)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Background(thank-you-for-calling)
exten =>

s,n,Dial(${QPHONE0}&${QPHONE1}&${QPHONE2}&${QPHONE3}&${QPHONE4}&${QPHONE5}&${QPHONE6}&${QPHONE7},40,Ttr)
exten => s,n,Hangup


[incoming2]
include => from-internal
include => parkedcalls
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Set(CHANNEL(musicclass)=QCI)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Background(thank-you-for-calling)
exten =>

s,n,Dial(${ACAPHONE0}&${ACAPHONE1}&${ACAPHONE2}&${ACAPHONE3}&${ACAPHONE4}&${ACAPHONE5}&${ACAPHONE6}&${ACAPHONE7},40,TTr)
exten => s,n,Hangup

[incoming3]
include => from-internal
include => parkedcalls
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Set(CHANNEL(musicclass)=QCI)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Background(thank-you-for-calling)
exten => s,n,GotoIfTime(17:01-09:00|*|*|*?closed,s,1)
exten =>

s,n,Dial(${GMNETPHONE0}&${GMNETPHONE1}&${GMNETPHONE2}&${GMNETPHONE3}&${GMNETPHONE4}&${GMNETPHONE5}&${GMNETPHONE6}&${GMNETPHONE7},40,Ttr)
exten => s,n,Hangup

[from-interal]
include => dialout1
include => dialout2
include => dialout3
include => parkedcalls
include => intercom

exten => 10,1,Macro(oneline,${QPHONE0})
exten => 11,1,Macro(oneline,${QPHONE1})
exten => 12,1,Macro(oneline,${QPHONE2})
exten => 13,1,Macro(oneline,${QPHONE3})
exten => 14,1,Macro(oneline,${QPHONE4})
exten => 15,1,Macro(oneline,${QPHONE5})
exten => 16,1,Macro(oneline,${QPHONE6})
exten => 17,1,Macro(oneline,${QPHONE7})

exten => 20,1,Macro(oneline,${ACAPHONE0})
exten =&g

Re: [asterisk-users] help with dialplan

[Todd Reese]

  I actually found that one and corrected it.  I have replaced the 
context with the from-internal, remote, and dialout1.  Each has produced 
the same results of a 404 error.




On 8/30/2010 2:10 PM, Paul Belanger wrote:
> On Mon, Aug 30, 2010 at 11:42 AM, Todd Reese  wrote:
>>   Here's a debug for extension 150
>>
> In the future, simply attach your debug log to your email.  Here is
> your problem:
>
> [Aug 30 11:34:55] NOTICE[2079] chan_sip.c: Call from '150' to extension
> '6789542133' rejected because extension not found in context
> 'extensions.conf'.
>
>


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Re: [asterisk-users] help with dialplan

[Todd Reese]

 Unfortunately, that didn't work.  The phone is still giving me a 404 
error.


I have my own system that is 1.6.2.7 with Grandstream phones that works 
fine.  Using it as a guide, I built this server for a client which also 
has Grandstream phones.


Last week, it dialed out fine.  Since the weekend, no dialing at all.

On 8/30/2010 11:42 AM, Bryant Zimmerman wrote:

Todd

Your context must be set to where you want your extension to start 
each time it dials out. Without getting into your dialplan code too 
much try changing the context to point to dialout1


context=dialout1

If dialout1 is working you should be able to dial.

The best way to handle this is to create a context that when you dial 
from your phones it decieds if you have dialed an extension or an 
external number and then routes the call correclty. This way you can 
pickup an extension and dial either and get the desired results.


Bryant




*From*: "Todd Reese" 
*Sent*: Monday, August 30, 2010 11:20 AM
*To*: asterisk-users@lists.digium.com
*Subject*: Re: [asterisk-users] help with dialplan

Here is the sip.conf portion for extension 150

[150]
deny=0.0.0.0/0.0.0.0
type=friend
secret=1234567890
qualify=yes
port=5060
pickupgroup=
permit=0.0.0.0/0.0.0.0
nat=yes
host=dynamic
dtmfmode=rfc2833
dial=SIP/150
context=from-trunk
canreinvite=no
callgroup=
callerid=device <150>
accountcode=
call-limit=50


On 8/30/2010 10:37 AM, Bryant Zimmerman wrote:

Todd

How do you have the context in the phones sip configs set?

Bryant
*
From*: "Todd Reese" trees...@gmail.com <mailto:trees...@gmail.com>

Hi all,

I've been have problems with getting this system on line and would like
to acquire some help with the extensions.conf.

My current problem is that the phones won't dialout.on the VOIP lines
listed as dialout1, dialout2, dialout3. This version of asterisk is
1.6.2.11. Below is the extensions.conf file.


[globals]



QPHONE0=SIP/10
QPHONE1=SIP/11
QPHONE2=SIP/12
QPHONE3=SIP/13
QPHONE4=SIP/14
QPHONE5=SIP/15
QPHONE6=SIP/16
QPHONE7=SIP/17

ACAPHONE0=SIP/20
ACAPHONE1=SIP/21
ACAPHONE2=SIP/22
ACAPHONE3=SIP/23
ACAPHONE4=SIP/24
ACAPHONE5=SIP/25
ACAPHONE6=SIP/26
ACAPHONE7=SIP/27

GMNETPHONE0=SIP/30
GMNETPHONE1=SIP/31
GMNETPHONE2=SIP/32
GMNETPHONE3=SIP/33
GMNETPHONE4=SIP/34
GMNETPHONE5=SIP/35
GMNETPHONE6=SIP/36
GMNETPHONE7=SIP/37

EXTERNPHONE0=SIP/150

CPHONE1=SIP/1678000
CPHONE2=SIP/177

EMERGENCY=0
EMERGENCY_TRUNK=DAHDI/G1
; Change this for production use:
EMERGENCY_NUM=6789542133


[from-pstn]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)




[from-pstn1]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn2]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn3]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn4]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming3,s,1)

[from-pstn5]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming2,s,1)

[from-pstn6]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn7]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn8]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)


[incoming1]
include => from-internal
include => parkedcalls
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Set(CHANNEL(musicclass)=QCI)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Background(thank-you-for-calling)
exten =>
s,n,Dial(${QPHONE0}&${QPHONE1}&${QPHONE2}&${QPHONE3}&${QPHONE4}&${QPHONE5}&${QPHONE6}&${QPHONE7},40,Ttr)
exten => s,n,Hangup


[incoming2]
include => from-internal
include => parkedcalls
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Set(CHANNEL(musicclass)=QCI)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Background(thank-you-for-calling)
exten =>
s,n,Dial(${ACAPHONE0}&${ACAPHONE1}&${ACAPHONE2}&${ACAPHONE3}&${ACAPHONE4}&${ACAPHONE5}&${ACAPHONE6}&${ACAPHONE7},40,TTr)
exten => s,n,Hangup

[incoming3]
include => from-internal
include => parkedcalls
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Set(CHANNEL(musicclass)=QCI)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TI

Re: [asterisk-users] help with dialplan

[Todd Reese]

  Here's a debug for extension 150



[Aug 30 11:34:53] VERBOSE[2099] config.c:   == Parsing 
'/etc/asterisk/logger.conf': [Aug 30 11:34:53] DEBUG[2099] config.c: 
Parsing /etc/asterisk/logger.conf
[Aug 30 11:34:53] VERBOSE[2099] config.c:   == Found
[Aug 30 11:34:53] VERBOSE[2099] logger.c:  Asterisk Event Logger restarted
[Aug 30 11:34:53] VERBOSE[2099] logger.c:  Asterisk Queue Logger restarted
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c:  Header  0 [ 38]: OPTIONS 
sip:76.122.117.31:5060 SIP/2.0
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c:  Header  1 [ 44]: Via: 
SIP/2.0/UDP 64.34.245.174:5060;branch=0
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c:  Header  2 [ 38]: From: 
sip:pin...@voip.com;tag=7c9c6206
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c:  Header  3 [ 26]: To: 
sip:76.122.117.31:5060
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c:  Header  4 [ 47]: Call-ID: 
89c833e4-9c8f2516-5bb...@64.34.245.174
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c:  Header  5 [ 15]: CSeq: 1 OPTIONS
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c:  Header  6 [ 17]: 
Content-Length: 0
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c:  Header  7 [  0]:
[Aug 30 11:34:53] DEBUG[2079] acl.c: Found IP address for this socket
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Setting SIP_TRANSPORT_UDP with 
address 10.0.1.102:5060
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Allocating new SIP dialog for 
89c833e4-9c8f2516-5bb...@64.34.245.174 - OPTIONS (No RTP)
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c:  Received OPTIONS (3) - 
Command in SIP OPTIONS
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Trying to put 'SIP/2.0 404' 
onto UDP socket destined for 64.34.245.174:5060
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c: SIP message could not be 
handled, bad request: 89c833e4-9c8f2516-5bb...@64.34.245.174
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c:  Header  0 [ 38]: OPTIONS 
sip:76.122.117.31:5060 SIP/2.0
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c:  Header  1 [ 44]: Via: 
SIP/2.0/UDP 64.34.245.174:5060;branch=0
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c:  Header  2 [ 38]: From: 
sip:pin...@voip.com;tag=1f9c6206
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c:  Header  3 [ 26]: To: 
sip:76.122.117.31:5060
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c:  Header  4 [ 47]: Call-ID: 
89c833e4-3f8f2516-5bb...@64.34.245.174
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c:  Header  5 [ 15]: CSeq: 1 OPTIONS
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c:  Header  6 [ 17]: 
Content-Length: 0
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c:  Header  7 [  0]:
[Aug 30 11:34:53] DEBUG[2079] acl.c: Found IP address for this socket
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Setting SIP_TRANSPORT_UDP with 
address 10.0.1.102:5060
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Allocating new SIP dialog for 
89c833e4-3f8f2516-5bb...@64.34.245.174 - OPTIONS (No RTP)
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c:  Received OPTIONS (3) - 
Command in SIP OPTIONS
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Trying to put 'SIP/2.0 404' 
onto UDP socket destined for 64.34.245.174:5060
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c: SIP message could not be 
handled, bad request: 89c833e4-3f8f2516-5bb...@64.34.245.174
[Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Auto destroying SIP dialog 
'89c833e4-806e2516-79b...@64.34.245.174'
[Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Destroying SIP dialog 
89c833e4-806e2516-79b...@64.34.245.174
[Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Auto destroying SIP dialog 
'89c833e4-236e2516-79b...@64.34.245.174'
[Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Destroying SIP dialog 
89c833e4-236e2516-79b...@64.34.245.174
[Aug 30 11:34:54] VERBOSE[2079] chan_sip.c:
<--- SIP read from UDP:97.80.176.231:5060 --->



<->
[Aug 30 11:34:54] DEBUG[2079] chan_sip.c:  Header  0 [  0]:
[Aug 30 11:34:54] DEBUG[2079] chan_sip.c:Body  0 [  0]:
[Aug 30 11:34:54] VERBOSE[2079] chan_sip.c:
<--- SIP read from UDP:97.80.176.231:5060 --->
INVITE sip:6789542...@qci.homeip.net SIP/2.0
Via: SIP/2.0/UDP 10.11.17.24:5060;branch=z9hG4bK82d40955a28df1af
From: "ATAP" ;tag=ee0cedf5f71d40f9
To: 
Contact: 
Supported: replaces, timer, path
P-Early-Media: Supported
Call-ID: 62f35b2ee0ada...@10.11.17.24
CSeq: 21395 INVITE
User-Agent: Grandstream GXP2000 1.2.3.5
Max-Forwards: 70
Allow: 
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 345

v=0
o=150 8000 8000 IN IP4 10.11.17.24
s=SIP Call
c=IN IP4 10.11.17.24
t=0 0
m=audio 5050 RTP/AVP 0 8 4 18 2 97 9 3
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=ptime:20

<->
[Aug 30 11:34:54] DEBUG[2079] chan_sip.c:  Header  0 [ 44]: INVITE 
sip:6789542...@qci.homeip.net SIP/2.0
[Aug 30 11:34:54] DEBUG[2079] chan_sip.c:  Header  1 [ 64]: Via: 
SIP/2.0/UDP 10.11.17.24:5060;branch=z9hG4bK82d40955a28df1af
[Aug 30 11:34:54] DEBUG[2079] chan_sip.c:  Header  2 [ 58]: From: "ATAP" 
;tag=ee0

Re: [asterisk-users] help with dialplan

[Todd Reese]

 Here is the sip.conf portion for extension 150

[150]
deny=0.0.0.0/0.0.0.0
type=friend
secret=1234567890
qualify=yes
port=5060
pickupgroup=
permit=0.0.0.0/0.0.0.0
nat=yes
host=dynamic
dtmfmode=rfc2833
dial=SIP/150
context=from-trunk
canreinvite=no
callgroup=
callerid=device <150>
accountcode=
call-limit=50


On 8/30/2010 10:37 AM, Bryant Zimmerman wrote:

Todd

How do you have the context in the phones sip configs set?

Bryant
*
From*: "Todd Reese" trees...@gmail.com <mailto:trees...@gmail.com>

Hi all,

I've been have problems with getting this system on line and would like
to acquire some help with the extensions.conf.

My current problem is that the phones won't dialout.on the VOIP lines
listed as dialout1, dialout2, dialout3. This version of asterisk is
1.6.2.11. Below is the extensions.conf file.


[globals]



QPHONE0=SIP/10
QPHONE1=SIP/11
QPHONE2=SIP/12
QPHONE3=SIP/13
QPHONE4=SIP/14
QPHONE5=SIP/15
QPHONE6=SIP/16
QPHONE7=SIP/17

ACAPHONE0=SIP/20
ACAPHONE1=SIP/21
ACAPHONE2=SIP/22
ACAPHONE3=SIP/23
ACAPHONE4=SIP/24
ACAPHONE5=SIP/25
ACAPHONE6=SIP/26
ACAPHONE7=SIP/27

GMNETPHONE0=SIP/30
GMNETPHONE1=SIP/31
GMNETPHONE2=SIP/32
GMNETPHONE3=SIP/33
GMNETPHONE4=SIP/34
GMNETPHONE5=SIP/35
GMNETPHONE6=SIP/36
GMNETPHONE7=SIP/37

EXTERNPHONE0=SIP/150

CPHONE1=SIP/1678000
CPHONE2=SIP/177

EMERGENCY=0
EMERGENCY_TRUNK=DAHDI/G1
; Change this for production use:
EMERGENCY_NUM=6789542133


[from-pstn]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)




[from-pstn1]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn2]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn3]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn4]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming3,s,1)

[from-pstn5]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming2,s,1)

[from-pstn6]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn7]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn8]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)


[incoming1]
include => from-internal
include => parkedcalls
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Set(CHANNEL(musicclass)=QCI)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Background(thank-you-for-calling)
exten =>
s,n,Dial(${QPHONE0}&${QPHONE1}&${QPHONE2}&${QPHONE3}&${QPHONE4}&${QPHONE5}&${QPHONE6}&${QPHONE7},40,Ttr)
exten => s,n,Hangup


[incoming2]
include => from-internal
include => parkedcalls
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Set(CHANNEL(musicclass)=QCI)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Background(thank-you-for-calling)
exten =>
s,n,Dial(${ACAPHONE0}&${ACAPHONE1}&${ACAPHONE2}&${ACAPHONE3}&${ACAPHONE4}&${ACAPHONE5}&${ACAPHONE6}&${ACAPHONE7},40,TTr)
exten => s,n,Hangup

[incoming3]
include => from-internal
include => parkedcalls
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Set(CHANNEL(musicclass)=QCI)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Background(thank-you-for-calling)
exten => s,n,GotoIfTime(17:01-09:00|*|*|*?closed,s,1)
exten =>
s,n,Dial(${GMNETPHONE0}&${GMNETPHONE1}&${GMNETPHONE2}&${GMNETPHONE3}&${GMNETPHONE4}&${GMNETPHONE5}&${GMNETPHONE6}&${GMNETPHONE7},40,Ttr)
exten => s,n,Hangup

[from-interal]
include => dialout1
include => dialout2
include => dialout3
include => parkedcalls
include => intercom

exten => 10,1,Macro(oneline,${QPHONE0})
exten => 11,1,Macro(oneline,${QPHONE1})
exten => 12,1,Macro(oneline,${QPHONE2})
exten => 13,1,Macro(oneline,${QPHONE3})
exten => 14,1,Macro(oneline,${QPHONE4})
exten => 15,1,Macro(oneline,${QPHONE5})
exten => 16,1,Macro(oneline,${QPHONE6})
exten => 17,1,Macro(oneline,${QPHONE7})

exten => 20,1,Macro(oneline,${ACAPHONE0})
exten => 21,1,Macro(oneline,${ACAPHONE1})
exten => 22,1,Macro(oneline,${ACAPHONE2})
exten => 23,1,Macro(oneline,${ACAPHONE3})
exten => 24,1,Macro(oneline,${ACAPHONE4})
exten => 25,1,Macro(oneline,${ACAPHONE5})
exten => 26,1,Macro(oneline,${ACAPHONE6})
exten => 27,1,Macro(oneline,${ACAPHONE7})

[asterisk-users] help with dialplan

[Todd Reese]

  Hi all,

I've been have problems with getting this system on line and would like 
to acquire some help with the extensions.conf.

My current problem is that the phones won't dialout.on the VOIP lines 
listed as dialout1, dialout2, dialout3. This version of asterisk is 
1.6.2.11.  Below is the extensions.conf file.


[globals]



QPHONE0=SIP/10
QPHONE1=SIP/11
QPHONE2=SIP/12
QPHONE3=SIP/13
QPHONE4=SIP/14
QPHONE5=SIP/15
QPHONE6=SIP/16
QPHONE7=SIP/17

ACAPHONE0=SIP/20
ACAPHONE1=SIP/21
ACAPHONE2=SIP/22
ACAPHONE3=SIP/23
ACAPHONE4=SIP/24
ACAPHONE5=SIP/25
ACAPHONE6=SIP/26
ACAPHONE7=SIP/27

GMNETPHONE0=SIP/30
GMNETPHONE1=SIP/31
GMNETPHONE2=SIP/32
GMNETPHONE3=SIP/33
GMNETPHONE4=SIP/34
GMNETPHONE5=SIP/35
GMNETPHONE6=SIP/36
GMNETPHONE7=SIP/37

EXTERNPHONE0=SIP/150

CPHONE1=SIP/1678000
CPHONE2=SIP/177

EMERGENCY=0
EMERGENCY_TRUNK=DAHDI/G1
; Change this for production use:
EMERGENCY_NUM=6789542133


[from-pstn]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)




[from-pstn1]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn2]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn3]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn4]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming3,s,1)

[from-pstn5]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming2,s,1)

[from-pstn6]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn7]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn8]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)


[incoming1]
include => from-internal
include => parkedcalls
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Set(CHANNEL(musicclass)=QCI)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Background(thank-you-for-calling)
exten => 
s,n,Dial(${QPHONE0}&${QPHONE1}&${QPHONE2}&${QPHONE3}&${QPHONE4}&${QPHONE5}&${QPHONE6}&${QPHONE7},40,Ttr)
exten => s,n,Hangup


[incoming2]
include => from-internal
include => parkedcalls
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Set(CHANNEL(musicclass)=QCI)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Background(thank-you-for-calling)
exten => 
s,n,Dial(${ACAPHONE0}&${ACAPHONE1}&${ACAPHONE2}&${ACAPHONE3}&${ACAPHONE4}&${ACAPHONE5}&${ACAPHONE6}&${ACAPHONE7},40,TTr)
exten => s,n,Hangup

[incoming3]
include => from-internal
include => parkedcalls
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Set(CHANNEL(musicclass)=QCI)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Background(thank-you-for-calling)
exten => s,n,GotoIfTime(17:01-09:00|*|*|*?closed,s,1)
exten => 
s,n,Dial(${GMNETPHONE0}&${GMNETPHONE1}&${GMNETPHONE2}&${GMNETPHONE3}&${GMNETPHONE4}&${GMNETPHONE5}&${GMNETPHONE6}&${GMNETPHONE7},40,Ttr)
exten => s,n,Hangup

[from-interal]
include => dialout1
include => dialout2
include => dialout3
include => parkedcalls
include => intercom

exten => 10,1,Macro(oneline,${QPHONE0})
exten => 11,1,Macro(oneline,${QPHONE1})
exten => 12,1,Macro(oneline,${QPHONE2})
exten => 13,1,Macro(oneline,${QPHONE3})
exten => 14,1,Macro(oneline,${QPHONE4})
exten => 15,1,Macro(oneline,${QPHONE5})
exten => 16,1,Macro(oneline,${QPHONE6})
exten => 17,1,Macro(oneline,${QPHONE7})

exten => 20,1,Macro(oneline,${ACAPHONE0})
exten => 21,1,Macro(oneline,${ACAPHONE1})
exten => 22,1,Macro(oneline,${ACAPHONE2})
exten => 23,1,Macro(oneline,${ACAPHONE3})
exten => 24,1,Macro(oneline,${ACAPHONE4})
exten => 25,1,Macro(oneline,${ACAPHONE5})
exten => 26,1,Macro(oneline,${ACAPHONE6})
exten => 27,1,Macro(oneline,${ACAPHONE7})

exten => 30,1,Macro(oneline,${GMNETPHONE0})
exten => 31,1,Macro(oneline,${GMNETPHONE1})
exten => 32,1,Macro(oneline,${GMNETPHONE2})
exten => 33,1,Macro(oneline,${GMNETPHONE3})
exten => 34,1,Macro(oneline,${GMNETPHONE4})
exten => 35,1,Macro(oneline,${GMNETPHONE5})
exten => 36,1,Macro(oneline,${GMNETPHONE6})
exten => 37,1,Macro(oneline,${GMNETPHONE7})

exten => 40,1,Macro(oneline,${QPHONE0})
exten => 41,1,Macro(oneline,${QPHONE1})
exten => 42,1,Macro(oneline,${QPHONE2})
exten => 43,1,Macro(oneline,${QPHONE3})
exten => 44,1,Macro(oneline,${QPHONE4})
exten => 45,1,Macro(oneline,${QPHONE5})
exten => 46,1,Macro(oneline,${QPHONE6})
exten => 47,1,Macro(oneline,${QPHONE7})

exten => 150,1,Macro(oneline,${EXTERNPHONE0})




[macro-oneline]
exten => s,1,Set(CHANNEL(musicclass)=default)
exten => s,n,Dial(${ARG1},20,Ttr)
exten => s,n,Voicemail(${MACRO_EXTEN})
exten => s,n,Hangup
exten => s,102,Vo

[asterisk-users] GXP-2000 transfer & hold problem

[Todd Reese]

  Hi all,


I'm working on a system with 4 Grandstream GP-200 Phones and the base 
Asterisk install.

I have added a 5 phone which is remote to the client and located in my 
office.

I can't get the phone to transfer a call or put a call on hold.   This 
applies to all the phones at the location.

I have been looking over configs and I'm at a loss right now.

Any help in pointing this out would be greatly appreciated.


-- 
_
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Re: [asterisk-users] Dahdi install gone wrong

[Todd Reese]

  I've made the system work by overlaying the old trixbox config in 
/etc/asterisk.  BUT this is a disaster waiting to happen with this client.

I'm having a hard time deciphering the trixbox extensions*.conf files in 
order to make a simple system where the client won't muck it up.

On 8/23/2010 11:37 AM, Cassius Smith wrote:
>* -Original Message-
>* From: Todd Reese
>* Reply-to: Asterisk Users Mailing List - Non-Commercial
>  Discussion
>* To: asterisk-users@lists.digium.com
>* Subject: [asterisk-users] Dahdi install gone wrong
>* Date: Mon, 23 Aug 2010 10:26:58 -0400
>*
>* Hi All,
>*
>*
>* I've got a project installing a Digium TDM800P card with 8 FXO's
>  in an
>* Asterisk box.
>*
>*
>* The computer is running Slackware 13.1 and I've installed the
>  current
>* Dahdi and Asterisk 1.6.2.11.
>*
>*
>* I've installed several boxes that are pure VOIP but, I haven't
>  installed
>* a Dahdi interface and I'm stumped.  I've got it to the point of
>  Dahdi
>* seeing the card and Asterisk recognizing dahdi but, I can't see
>  any
>* channels for the calls to come in on.
>*
>* I've had to borrow files from an old config of Trixbox (the
>  machine was
>* underpowered) to get to the point where I am in my setup.
>*
>* I would like to inquire some help from the group to get me up
>  and
>* receiving calls on the card.
>*
>*
>* Regards,
>*
>* Todd Reese
>*
>* Include:
>*
>*
>* chan_dahdi.conf==
>*
>*
>* ; Configuration file
>*
>* [trunkgroups]
>*
>* [channels]
>*
>* language=en
>* context=from-zaptel
>* signalling=fxs_ks
>* rxwink=300  ; Atlas seems to use long (250ms) winks
>* ;
>* ; Whether or not to do distinctive ring detection on FXO lines
>* ;
>* ;usedistinctiveringdetection=yes
>*
>* usecallerid=yes
>* hidecallerid=no
>* callwaiting=yes
>* usecallingpres=yes
>* callwaitingcallerid=yes
>* threewaycalling=yes
>* transfer=yes
>* cancallforward=yes
>* callreturn=yes
>* echocancel=yes
>* echocancelwhenbridged=no
>* ;echotraining=800
>* rxgain=0.0
>* txgain=0.0
>* group=0
>* callgroup=1
>* pickupgroup=1
>* immediate=no
>*
>* ;faxdetect=both
>* faxdetect=incoming
>* ;faxdetect=outgoing
>* ;faxdetect=no
>*
>* ;Include setup-pstn configs
>* #include dahdi-channels.conf
>*
>* group=1
>*
>* ;Include PBXconfig configs
>* #include chan_dahdi_additional.conf
>*
>*
>*
>* dahdi-channels.conf=
>*
>* ; Autogenerated by /usr/sbin/dahdi_genconf on Wed Aug 18
>  20:25:02 2010
>* ; If you edit this file and execute /usr/sbin/dahdi_genconf
>  again,
>* ; your manual changes will be LOST.
>* ; Dahdi Channels Configurations (chan_dahdi.conf)
>* ;
>* ; This is not intended to be a complete chan_dahdi.conf. Rather,
>  it is
>* intended
>* ; to be #include-d by /etc/chan_dahdi.conf that will include the
>  global
>* settings
>* ;
>*
>* ; Span 1: WCTDM/0 "Wildcard TDM800P Board 1" (MASTER)
>* ;;; line="1 WCTDM/0/0 FXSKS  (SWEC: MG2)"
>* signalling=fxs_ks
>* callerid=asreceived
>* group=0
>* context=from-pstn
>* channel =>  1
>* callerid=
>* group=
>* context=default
>*
>* ;;; line="2 WCTDM/0/1 FXSKS  (SWEC: MG2)"
>* signalling=fxs_ks
>* callerid=asreceived
>* group=0
>* context=from-pstn
>* channel =>  2
>* callerid=
>* group=
>* context=default
>*
>* ;;; line="3 WCTDM/0/2 FXSKS  (SWEC: MG2)"
>* signalling=fxs_ks
>* callerid=asreceived
>* group=0
>* context

Re: [asterisk-users] Dahdi install gone wrong

[Todd Reese]

 They are FXO modules and yes, the lines are coming in from the telco.

On 8/23/2010 12:05 PM, Doug Dawson wrote:


The card you installed has FXO or FXS modules in it ? are you 
getting your lines directly from the telco co???



Doug D


On Mon 23/08/10 8:37 AM , Cassius Smith cass...@cassius.org sent:

* -Original Message-
* From: Todd Reese mailto:trees...@gmail.com>>
* Reply-to: Asterisk Users Mailing List - Non-Commercial
Discussion mailto:asterisk-users@lists.digium.com>>
* To: asterisk-users@lists.digium.com
<mailto:asterisk-users@lists.digium.com>
* Subject: [asterisk-users] Dahdi install gone wrong
* Date: Mon, 23 Aug 2010 10:26:58 -0400
*
* Hi All,
*
*
* I've got a project installing a Digium TDM800P card with 8 FXO's
in an
* Asterisk box.
*
*
* The computer is running Slackware 13.1 and I've installed the
current
* Dahdi and Asterisk 1.6.2.11.
*
*
* I've installed several boxes that are pure VOIP but, I haven't
installed
* a Dahdi interface and I'm stumped. I've got it to the point of
Dahdi
* seeing the card and Asterisk recognizing dahdi but, I can't see
any
* channels for the calls to come in on.
*
* I've had to borrow files from an old config of Trixbox (the
machine was
* underpowered) to get to the point where I am in my setup.
*
* I would like to inquire some help from the group to get me up
and
* receiving calls on the card.
*
*
* Regards,
*
* Todd Reese
*
* Include:
*
*
* chan_dahdi.conf==
*
*
* ; Configuration file
*
* [trunkgroups]
*
* [channels]
*
* language=en
* context=from-zaptel
* signalling=fxs_ks
* rxwink=300 ; Atlas seems to use long (250ms) winks
* ;
* ; Whether or not to do distinctive ring detection on FXO lines
* ;
* ;usedistinctiveringdetection=yes
*
* usecallerid=yes
* hidecallerid=no
* callwaiting=yes
* usecallingpres=yes
* callwaitingcallerid=yes
* threewaycalling=yes
* transfer=yes
* cancallforward=yes
* callreturn=yes
* echocancel=yes
* echocancelwhenbridged=no
* ;echotraining=800
* rxgain=0.0
* txgain=0.0
* group=0
* callgroup=1
* pickupgroup=1
* immediate=no
*
* ;faxdetect=both
* faxdetect=incoming
* ;faxdetect=outgoing
* ;faxdetect=no
*
* ;Include setup-pstn configs
* #include dahdi-channels.conf
*
* group=1
*
* ;Include PBXconfig configs
* #include chan_dahdi_additional.conf
*
*
*
* dahdi-channels.conf=
*
* ; Autogenerated by /usr/sbin/dahdi_genconf on Wed Aug 18
20:25:02 2010
* ; If you edit this file and execute /usr/sbin/dahdi_genconf
again,
* ; your manual changes will be LOST.
* ; Dahdi Channels Configurations (chan_dahdi.conf)
* ;
* ; This is not intended to be a complete chan_dahdi.conf. Rather,
it is
* intended
* ; to be #include-d by /etc/chan_dahdi.conf that will include the
global
* settings
* ;
*
* ; Span 1: WCTDM/0 "Wildcard TDM800P Board 1" (MASTER)
* ;;; line="1 WCTDM/0/0 FXSKS (SWEC: MG2)"
* signalling=fxs_ks
* callerid=asreceived
* group=0
* context=from-pstn
* channel => 1
* callerid=
* group=
* context=default
*
* ;;; line="2 WCTDM/0/1 FXSKS (SWEC: MG2)"
* signalling=fxs_ks
* callerid=asreceived
* group=0
* context=from-pstn
* channel => 2
* callerid=
* group=
* context=default
*
* ;;; line="3 WCTDM/0/2 FXSKS (SWEC: MG2)"
* signalling=fxs_ks
* callerid=asreceived
* group=0
* context=from-pstn
* channel => 3
* callerid=
* group=
* context=default
*
* ;;; line="4 WCTDM/0/3 FXSKS (SWEC: MG2)"
* signalling=fxs_ks
* callerid=asreceived
* group=0
* context=from-pstn
* channel => 4
* callerid=
* group=
* context=default
*
* ;;; line="5 WCTDM/0/4 FXSKS (SWEC: MG2)"
* signalling=fxs_ks
* callerid=asreceived
* group=0
* context=from-pstn
* channel => 5
* callerid=
* group=
* context=default
*
* ;;; line="6 WCTDM/0/5 FXSKS (SWEC: MG2)"
* signalling=fxs_ks
* callerid=asreceived
* group=0
* context=from-pstn
* channel => 6
* callerid=
* group=
* context=default
*
* ;;; line="7 WCTDM/0/6 FXSKS (SWEC: MG2)"
* signalling=fxs_ks
* callerid=asreceived
* group=0
* context=from-pstn
* channel => 7
* callerid=
* group=
* context=default
*
* ;;; line="8 WCTDM/0/7 FXSKS (SWEC: MG2)"
* signalling=f

[asterisk-users] Dahdi install gone wrong

[Todd Reese]

  Hi All,


I've got a project installing a Digium TDM800P card with 8 FXO's in an 
Asterisk box.


The computer is running Slackware 13.1 and I've installed the current 
Dahdi and Asterisk 1.6.2.11.


I've installed several boxes that are pure VOIP but, I haven't installed 
a Dahdi interface and I'm stumped.  I've got it to the point of Dahdi 
seeing the card and Asterisk recognizing dahdi but, I can't see any 
channels for the calls to come in on.

I've had to borrow files from an old config of Trixbox (the machine was 
underpowered) to get to the point where I am in my setup.

I would like to inquire some help from the group to get me up and 
receiving calls on the card.


Regards,

Todd Reese

Include:


chan_dahdi.conf==


; Configuration file

[trunkgroups]

[channels]

language=en
context=from-zaptel
signalling=fxs_ks
rxwink=300  ; Atlas seems to use long (250ms) winks
;
; Whether or not to do distinctive ring detection on FXO lines
;
;usedistinctiveringdetection=yes

usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
;echotraining=800
rxgain=0.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no

;faxdetect=both
faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no

;Include setup-pstn configs
#include dahdi-channels.conf

group=1

;Include PBXconfig configs
#include chan_dahdi_additional.conf



dahdi-channels.conf=

; Autogenerated by /usr/sbin/dahdi_genconf on Wed Aug 18 20:25:02 2010
; If you edit this file and execute /usr/sbin/dahdi_genconf again,
; your manual changes will be LOST.
; Dahdi Channels Configurations (chan_dahdi.conf)
;
; This is not intended to be a complete chan_dahdi.conf. Rather, it is 
intended
; to be #include-d by /etc/chan_dahdi.conf that will include the global 
settings
;

; Span 1: WCTDM/0 "Wildcard TDM800P Board 1" (MASTER)
;;; line="1 WCTDM/0/0 FXSKS  (SWEC: MG2)"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 1
callerid=
group=
context=default

;;; line="2 WCTDM/0/1 FXSKS  (SWEC: MG2)"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 2
callerid=
group=
context=default

;;; line="3 WCTDM/0/2 FXSKS  (SWEC: MG2)"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 3
callerid=
group=
context=default

;;; line="4 WCTDM/0/3 FXSKS  (SWEC: MG2)"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 4
callerid=
group=
context=default

;;; line="5 WCTDM/0/4 FXSKS  (SWEC: MG2)"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 5
callerid=
group=
context=default

;;; line="6 WCTDM/0/5 FXSKS  (SWEC: MG2)"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 6
callerid=
group=
context=default

;;; line="7 WCTDM/0/6 FXSKS  (SWEC: MG2)"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 7
callerid=
group=
context=default

;;; line="8 WCTDM/0/7 FXSKS  (SWEC: MG2)"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 8
callerid=
group=
context=default


=system.conf=


# Autogenerated by /usr/sbin/dahdi_genconf on Sun Aug 22 19:34:02 2010
# If you edit this file and execute /usr/sbin/dahdi_genconf again,
# your manual changes will be LOST.
# Dahdi Configuration File
#
# This file is parsed by the Dahdi Configurator, dahdi_cfg
#
# Global data

loadzone= us
defaultzone = us

# Span 1: WCTDM/0 "Wildcard TDM800P Board 1" (MASTER)
fxsks=1
#echocanceller=mg2,1
fxsks=2
#echocanceller=mg2,2
fxsks=3
#echocanceller=mg2,3
fxsks=4
#echocanceller=mg2,4
fxsks=5
#echocanceller=mg2,5
fxsks=6
#echocanceller=mg2,6
fxsks=7
#echocanceller=mg2,7
fxsks=8
#echocanceller=mg2,8


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[asterisk-users] AstriCon approaches: Innovation Awards, your attendance wanted!

[John Todd]

Just a reminder: AstriCon is coming up in October in Washington, DC 
(http://www.astricon.net/ 
) and we're looking forward to seeing you there!

We're getting to the deadline for Innovation Awards for this year.   
What's an Innovation Award?  The Innovation Award is designed to  
recognize developers, customers and partners for outstanding  
achievements that are improving business processes, overcoming  
technology challenges and enhancing the company's bottom line.  Digium  
picks five different categories in which certain projects or companies  
have excelled in the last year creating amazing things with Asterisk.   
The awards are presented at AstriCon.

If you think you're doing something great with Asterisk, send it in!   
It's a great opportunity to be recognized as a leader in Asterisk  
development, implementation, and innovation.  August 1 is the deadline.

More details here -  http://www.digium.com/en/company/awards/innovation.php

Send your Innovation Award proposal to Julie Webb (jw...@digium.com)  
for inclusion.


AstriCon in general:
  I'll take this opportunity to ask everyone again to get your  
reservations in for AstriCon this year!  We're looking forward to a  
really good show, in a city slightly less oven-like than the past  
three years.  The conference has a fantastic line-up of speakers and  
as always, offers the opportunity to talk with people in an informal  
setting about their real-world experiences with Asterisk, VoIP,  
different hardware, methods of implementation, and make all sorts of  
connections that you just can't get without meeting face-to-face.   
Washington DC is convenient from Europe, with direct flights to IAD  
(Dulles), DCA (Reagan International), and BWI (Baltimore Washington)  
airports from most major European and South American cities.

JT


---
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445 Jan Davis Drive NW -  Huntsville AL 35806  -   USA
direct: +1-256-428-6083 http://www.digium.com/





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[asterisk-users] Last call for AstriCon talks

[John Todd]

AstriCon in Washington DC is only 102 days away! October 26-28 -  
slightly over three months - time is flying.  The early bird discount  
($595 for the whole conference) runs out next week - see if you can  
get in under the wire!

The final selection of AstriCon talks is under way.  If you've been  
intending to submit your talk and you missed the June 30 deadline...  
well, you're late.  :-)  But I've had so many good technical,  
informative proposals this year that I'd be foolish not to shake the  
tree one last time to see if any more perfect AstriCon session fruit  
falls into the basket.  This year is going to be heavily loaded with  
very in-depth examples, tutorials, and case studies.  We've got some  
really great talks from a fantastic array of speakers.  While I don't  
want to publish the whole list just yet (that'll be next week) I can  
tell you that there are how-to talks on IPv6 (a double-session!) by  
the developers of the code (Viagenie), VoIP encryption techniques by  
the developer of some of the code (Terry Wilson), and a practical  
session on SIP security by the author of SIPVicious (Sandro Gauci.)

If you've talked with me about giving a session, but not actually put  
it into the then it's not on the consideration list.   I know that  
there are quite a few of you who are enthusiastic about giving a  
session but haven't quite gained the momentum to fill out the form -  
now's your last chance!  Please don't deprive the attendees of your  
fantastic content!

PS: If you've submitted a talk: please be patient!  We're sifting and  
sorting and trying to find slots for everyone in the right way.

Submit a talk:http://bit.ly/speak-astricon201
Attend the show:  http://www.astricon.net/


JT

---
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445 Jan Davis Drive NW -  Huntsville AL 35806  -   USA
direct: +1-256-428-6083 http://www.digium.com/





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[asterisk-users] Using Asterisk? Get on the press list!

[John Todd]

I'll post my semi-annual request for press contacts.  If you have an  
Asterisk installation that matches one or more of these adjectives:

  - enterprise-oriented
  - government-oriented
  - education-oriented
  - unique
  - clever
  - large
  - complex

  ...we would be interested in having you talk with the press!  We at  
Digium keep a list of people who wouldn't mind talking to the press on  
their open-source Asterisk installations, because we get inquiries all  
the time about different things.  "We want to talk to people who are  
using Asterisk to do _!" is what we get, and it's great to be  
able to have a pocket-full of people who meet the criteria who don't  
mind spending a bit of time talking about how they use Asterisk to  
solve a particular task.

Remember: Asterisk and all open-source projects have almost zero press  
other than what YOU can help with.  We're competing with tens  
(hundreds?) of millions of dollars in press and marketing that is  
spent by proprietary vendors.  The press is very, very interested in  
hearing about Asterisk, but we don't have a giant list of OSS Asterisk  
users from which we can cherry-pick names, or offer discounts on  
yearly subscription fees (because there aren't any!) to help out with  
press interviews, etc.   Help the project, help yourself - put your  
name on the list!

If you've already filled out this form, no need to again - you're on  
the list, and if there is a press inquiry that seems to be a good  
match, we'll pass them to you!  If you're an integrator, please don't  
fill out the form - have your customers fill out the form - press  
folks are always looking for end users, not integrators.

   http://bit.ly/asterisk-press

JT

---
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Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW -  Huntsville AL 35806  -   USA
direct: +1-256-428-6083 http://www.digium.com/


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[asterisk-users] Digium Asterisk World at ITEXPO - Yahoo keynote update

[John Todd]

I don't know how many of you are going to be at ITEXPO/Digium Asterisk  
World in Miami next week - I hope to see as many of you as possible,  
though.

There has been an interesting change in the line-up for the show, that  
I think bears mentioning here since it possibly will help quite a few  
of you in your discussions about getting Asterisk into your company.   
We've had the good fortune to have a last-minute keynote addition at  
the show, which is going to be Jeremy Wadhams from Yahoo.   He's going  
to be talking about a Fortune 500 implementation of Asterisk across  
their entire network, and why they made the decision to move to Open- 
Source telephony for the core of their voice network.

Keynote: 3:30 - 4:00 PM on Wednesday  - Jeremy Wadhams, Yahoo

It's not yet on the ITEXPO site, since we only confirmed this  
yesterday with Jeremy, who is really being a great sport for doing  
this on short notice.

Having Yahoo giving a keynote on their multi-thousand seat  
installation is great news for the Asterisk community in general, and  
is great news specifically for other Enterprise managers who have been  
looking for that all-important set of examples of public users of a  
technology.  (Enterprise tends to move in herds.)  There are large  
numbers of Enterprise users of Asterisk, but it's always great to get  
a "name-brand" company who can be offered as a proponent of Asterisk.

Again, hope to see all of you there in Miami next week!

http://www.tmcnet.com/voip/conference/

JT

---
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Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW -  Huntsville AL 35806  -   USA
direct: +1-256-428-6083 http://www.digium.com/





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[asterisk-users] Federal, State, and Local government installations of Asterisk

[John Todd]

[sorry for cross-post here from -biz, but there are significant non- 
cross-subscribed audiences]

I'm looking for some case studies for people who are implementing Open  
Source Asterisk in Federal, State, and Local governments in the United  
States.  Please reply PRIVATELY to jt...@digium.com with contact data,  
and I'll follow up.

I have some requests from various directions for governmental uses of  
Asterisk, for people who are willing to talk to a magazine/journal  
about their use of Asterisk and how it works for them.

This kind of press contacts are vital to keeping Asterisk in the eye  
of regular business users, who often have industry journals as their  
only viewport to the outside technology world - if you know of any  
possible leads for something like this, please let me know and I'll  
hunt them down.  :-)

(PS: Any press contacts who want to add themselves to my list of  
contacts, go here:
 http://lists.digium.com/pipermail/asterisk-biz/2009-June/030694.html 
  )

JT

---
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Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW -  Huntsville AL 35806  -   USA
direct: +1-256-428-6083 http://www.digium.com/




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Re: [asterisk-users] Mixing commercial/SVN Asterisk

[John Todd]

On Dec 16, 2009, at 10:08 AM, Richard Kenner wrote:

> Am I correct that if I'm running an -rc or from an SVN release tree  
> that
> there's no way I can use any commercial add-ons from Digium, such as
> Skype, Cepstral, or G.729?


No, happily not correct.  :-)

Digium tries to make their add-ons work with all major releases of  
Asterisk.  You should be able to use all the named proprietary add-ons  
with 1.4 or 1.6.x versions of Asterisk.  SVN TRUNK releases of course  
are not guaranteed to work with the proprietary add-ons, but any  
stable version should function as you expect.  However, usually the  
add-ons work fine even in those cases where you are using -rc or TRUNK  
("usually" being the operative word here.)

JT

---
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Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW -  Huntsville AL 35806  -   USA
direct: +1-256-428-6083 http://www.digium.com/




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[asterisk-users] AstriCon Videos and Presentations: First batch is on-line!

[John Todd]

This year we recorded quite a few of the AstriCon sessions - 3 out of  
the 4 tracks were video taped.  The folks at TMC then went through a  
fairly painstaking process of synchronizing the video presentations to  
the slide decks that each presenter provided, so we have an index-able  
and fast-forward-able version of each talk.  I'm really excited about  
this video presentation method, since it's possible for the viewer to  
move back and forth in the video and see the complete slide  
presentation as well.   We have the first handful of videos ready for  
viewing, and here they are:

   http://www.astricon.net/2009/astricon/videoPresentations.aspx


We also have the vast majority of the presentations available for  
download as well.  Go to this page for the presentation grid:

http://www.astricon.net/2009/astricon/agendaAtaGlance.aspx

...and click on the name of the topic you're interested in on the  
grid.  If we have a copy of the talk, it will start to download to  
you.  If your talk isn't there, contact me with a copy of your slides  
in PPT or PDF format.

  I'd like to hear from people on what they think of the videos.  We  
have a number of additional videos of the remaining video/slide  
presentations to be post-produced and put on the site, and we'll get  
those published as soon as it's possible.  This is the first year  
we're trying this format.  I'm hoping that these videos will be used  
to further people's understanding of Asterisk, to convince others that  
Asterisk can do what they want, and to encourage people to attend  
AstriCon to meet with and talk to some of the people that make this  
project so powerful, useful, and exciting to be a part of.


  PS: I, personally, had some glitches with synchronization issues  
between the video and slides.  I suspect this is a local problem with  
my somewhat-bizarre network config, but  I'd like to hear back from  
any of you who notice incomplete video and audio experiences.  Include  
your ISP, country, and general problem description and we'll try to  
smooth the issues out with some better problem descriptions.

JT

---
John Todd   email:jt...@digium.com
Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW -  Huntsville AL 35806  -   USA
direct: +1-256-428-6083 http://www.digium.com/




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Re: [asterisk-users] Help with concurrent VoIP calls

[Todd Reese]

Hi Joe,

What is the app that generates your bandwidth table shown below?

Joe Greco wrote:

By "fast" I mean the best Business DSL Bellsouth has to offer: "Up to
6.0 Mbps downstream - Up to 512 Kbps upstream"



That almost sounds like an invitation to check out what business service
your cableco offers.

One thing to be aware of with DSL and cable modems is that there can be
various ill effects as your line gets closer to its rated capacity; do
not expect that you'll get a reliable 512Kbps upstream.  VoIP is sensitive
to loss, latency, and jitter.  You may be able, for example, to only get
384Kbps reliably out of the link (before packet loss/jitter/etc wreck its
suitability for VoIP).  That's a good time to look seriously at a gateway
package like pfSense that can prioritize certain classes of traffic while
also limiting overall bandwidth.

As an example, we noticed on the local business cable offering (2Mbps up)

Shaped  PL  min avg max stddev
2.2M3   6.4 251 557 176
2.1M1   7.8 350 584 134
2.0M3   6.4 271 535 132
1.9M1   7   254 527 131
1.8M0   6   79  339 90
1.75M   0   5.9 14  92  11
1.7M0   5.4 13  77  10
1.65M   0   4.9 11  69  7
1.6M0   5.4 13  55  9
1.5M0   5.3 11  59  7
1.4M0   5   11  57  7
1.3M0   4.9 11  54  6
1.2M0   4.9 11  52  7
1.1M0   4.8 14  53  11

The max starts trending up after 1.6M (helps to graph it) and pretty much
everything goes to hell after 1.75M.

... JG
  


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Re: [asterisk-users] [IAX] Recommended soft- and hardphones?

[John Todd]

On Oct 30, 2009, at 7:55 AM, Vincent wrote:

> Hello
>
> Since SIP/RTP is a pain to use with road warriors who need to connect
> from any location over the Internet, I'd like to get them some IAX
> phones instead.
>
> For those of you using this protocol instead of SIP, what would you
> recommend as IAX hardphones and Windows (and ideally Mac) softphones?
>
> Thank you.


I'll avoid the SIP/IAX religious war that usually breaks out over this  
topic and actually provide an answer to your question.  :-)

Citel just announced an IAX hardphone not long ago.  I haven't used  
it, but maybe you'd want to give it a shot and let the rest of us know  
how it works.  As far as softphones, I'd say Zoiper has been my choice  
for a while - runs on Mac and Windows.   (note: these are not official  
recommendations; just random personal selections made with no  
discernible criteria other than "it seemed like a good idea at the  
time")

JT


---
John Todd   email:jt...@digium.com
Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW -  Huntsville AL 35806  -   USA
direct: +1-256-428-6083 http://www.digium.com/




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[asterisk-users] AstriCon videos: a question of method

[John Todd]

I'm doing some quick research on how to get our videos from AstriCon  
available in a "reasonable" format that allows easy viewing, reduces  
our bandwidth costs, and allows good tracking for who/where/what is  
viewing the videos.

YouTube seems to have a very nice set of tools and statistics  
collection methods, and might be perfect EXCEPT  Their main  
limitation right now seems to be that they limit videos to 10 minutes,  
which clearly doesn't work for our longer presentations.  I could  
"patch" them together in multiple 10-minute sessions, but... ugh.  UGH.

There are other video sites out there - lots, actually.  I could spend  
hours digging through them all, or hopefully ask here on the list and  
have some people give me prior experiences based on their expertise  
with hosted video solutions.

Requirements (not exhaustive list):
   - free or very close to free (we'll pay, but not a lot)
   - good statistical collection (who is linking? how many views? how  
much video watched each view? where do people stop?)
   - reasonably easy interaction (good upload tools, good UI)
   - good viewing experience from North America, Europe, Asia

Before anyone suggests it, I'm not interested in Torrent-based  
distribution for various political reasons.  I've started to look at  
Flowplayer, which is appealing due to it's OSS nature and  
customization capability, but it leaves us holding the bandwidth bill  
(which may not be horrible, but it's a concern.)

What are your experiences?  I can't say we'll end up actually using  
what you might think is best, but I'm very interested to hear what  
everyone might suggest for distributing Asterisk-focused video material.

In the interests of keeping this thread from getting out of control,  
please limit yourself to factual, content-rich posts.  "I hate  
YouTube" or "Why didn't you film blah" is something we can discuss off- 
list.

JT

---
John Todd   email:jt...@digium.com
Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW -  Huntsville AL 35806  -   USA
direct: +1-256-428-6083 http://www.digium.com/




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Re: [asterisk-users] Astricon

[John Todd]

On Oct 17, 2009, at 7:47 PM, Michael Graves wrote:

> I'm told that they will show up on the event site in about three  
> weeks.
>
> On Sun, 18 Oct 2009 02:29:48 + (UTC), Jeff LaCoursiere wrote:
>
>>
>> Wish I could have made it :(  Is there a possibility of a  
>> collection of
>> the talks/slides/handouts/videos/presentations for download?  Even  
>> pay
>> for?
>>
>> Cheers,
>>
>> j


The presentations will be available "real soon now" but the videos may  
take a bit longer.

Indeed, there will be a cross-section of videos available soon.  We're  
working on the schedule for these, but it takes some time to post- 
process the videos and then we're probably not going to put them up  
all at once, nor will all of them appear.  Three of the four tracks  
were taped, and we'll pick some highlights (I'm taking suggestions -  
email me with your ideas if you liked a particular talk or want to see  
something specific.)

We have to balance a few things - if we put all the talks up, there is  
a fear (not universally held, I might add) that it will effect  
attendance next year.  Even a few percentage points would make the  
conference go from what is essentially a break-even to "in the red"  
and that's something we're trying very, VERY hard to avoid.  However,  
posting the best talks will also excite people about attending next  
year, so that's a positive for the conference.

Also, we'd like to use the talks as more than just advertising for  
AstriCon.  These videos will hopefully be used for various teaching  
purposes, or for convincing people that they can do fantastic things  
with Asterisk just like the speaker did.  I hope to see them used as  
sales tools that are used with management when skilled IT staff need  
to make a point about Asterisk; I hope to see them used in educational  
sessions where students can learn about how to use free tools; I hope  
to see them used in discussions with governments about who is already  
using OSS Asterisk to solve telecommunications problems.

JT

---
John Todd   email:jt...@digium.com
Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW -  Huntsville AL 35806  -   USA
direct: +1-256-428-6083 http://www.digium.com/




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Re: [asterisk-users] Astricon talk on wideband codecs

[John Todd]

Michael Graves & Tim Yankey (Polycom) gave that talk.

The site has already been (perhaps a bit prematurely) turned into the  
AstriCon 2010 site.  There is a link to the 2009 data on the left  
column.

You can find the agenda for last week here:  
http://www.astricon.net/2009/astricon/agendaAtaGlance.aspx

The videos will be going up in the next few weeks; I'll post a blog  
post and probably here on the list when they start to appear.

JT



On Oct 19, 2009, at 8:58 AM, Zoa wrote:

>
> I missed the talk that was given on wideband codecs @ astricon last  
> week.
> I tried to lookup the speaker on astricon.net, but that website seems
> horribly broken at the moment, showing only a tmcnet video, whatever
> page i click on.
>
> Would somebody have the contact details for that speaker ?
>
> Greetings,
>
> Zoa
>

---
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Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW -  Huntsville AL 35806  -   USA
direct: +1-256-428-6083 http://www.digium.com/




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Re: [asterisk-users] Drop Call on ICMP Port Unreachable?

[John Todd]

The idea of using ICMP unreachables as a method to shut down RTP  
streams (and corresponding signaling sessions) is a good one, and I'd  
like to see discussion on it.

There is the rtptimeout option in sip.conf which will possibly solve  
some of those symptoms (and has dangerous side-effects, when clients  
don't send CNG and are "on hold") but of course it is not the same  
mechanism and using ICMP unreachables is a better solution.

For those about to suggest it, the SIP session timers don't solve this  
problem in many cases, since signaling and RTP go to different places  
and ICMP unreachable on RTP doesn't imply that the signaling will also  
fail.

JT



On Oct 7, 2009, at 7:05 PM, Dan Mahoney, System Admin wrote:

> One of our users recently had a powerfail while connected to our  
> meetme
> gateway.  (Asterisk 1.4.17 on debian 4.0)
>
> Through the course of it, asterisk never hung up.  His system came  
> back
> up, and started sending ICMP port unreachables, but the stream went  
> on,
> flooding him with "silence" media stream packets (there was nobody  
> else in
> the conference).
>
> Is asterisk aware of ICMP unreachables?  Is there a tunable I can  
> set to
> make it be?
>
> I found a thread here that discusses it briefly:
>
> http://lists.digium.com/pipermail/asterisk-users/2005-March/ 
> 086626.html
>
> However, there's no real resolution there.
>
> If it's not aware of it, how difficult would it be to add?
>
> -Dan Mahoney
>
> -- 
>
> Dan Mahoney
> Techie,  Sysadmin,  WebGeek
> Gushi on efnet/undernet IRC
> ICQ: 13735144   AIM: LarpGM
> Site:  http://www.gushi.org
> ---

---
John Todd   email:jt...@digium.com
Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW -  Huntsville AL 35806  -   USA
direct: +1-256-428-6083 http://www.digium.com/




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[asterisk-users] Enterprise users going to VoiceCon?

[John Todd]

Are there any enterprise Asterisk end-users going to VoiceCon?   
(sorry, this is a situation where I'm interested in only in end-users  
- no consultants or integrators.)  Contact me off-list, please, thanks!

JT

---
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445 Jan Davis Drive NW -  Huntsville AL 35806  -   USA
direct: +1-256-428-6083 http://www.digium.com/




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[asterisk-users] AstriCon 2009: 30 days, 5 reasons & discount code

[John Todd]

We're down to slightly less than a month between now and AstriCon!   
October 13-15 is drawing close.  If you've not booked your travel  
reservations to Phoenix, now is the time to do it!

Sept 23rd is the cutoff date for the room discount, and we've  
requested another block of rooms for attendees.

The Renaissance is the exclusive hotel of AstriCon 2009.  It is where  
most of the attendees and exhibitors stay, and is a great place for  
networking!  Please make sure you book your hotel rooms right now to  
ensure you get the AstriCon fantastic discounted room rate.   
Availability is on a first-come first-served basis and the cut off  
date for the special rate is September 23 2009.

   Renaissance Glendale Resort & Spa
   http://cwp.marriott.com/phxgr/astricon09/
   Special rate - $144/night

There are other hotels in the area, so if the hotel fills (as it did  
last year) there are other options.  But getting a room at the  
Renaissance is probably your best bet, since you won't have to trudge  
across the arid parking lots or drive to another venue.  There are  
lots of restaurants close-by in the new entertainment center - I  
didn't go to the same place twice last year!

To answer the question that seems to be on everyone's lips: yes,  
AstriCon looks to be as big than last year, if not significantly  
bigger.  I know the economic situation is weighing on everyone's mind,  
but Open Source Asterisk installations are up and what is hurting the  
"big guys" is putting some wind under our wings.  AstriCon is where  
you'll see lots of people who are winning deals, creating revenue, and  
building the market of the PBX that is now the most-installed platform  
in North America (we're hoping to say world-wide VERY soon.)

Now, to encourage having everyone book a TINY bit in advance instead  
of all at the last minute (who, you? book at the last minute?  I know  
I'm not talking about you.)  I'll again announce that we have a  
discount code that you can use on your sign-up, which will give you a  
15% break on the conference price.  The code is "AC09" and you'd enter  
it on the registration page (http://www.astricon.net/ 
attendRegister.aspx) to get your discount.

Top 5 Reasons to attend AstriCon:

  * The talks!  This is yet another stellar line-up of talks this  
year, with a wide array of fascinating examples of how Asterisk is  
being used to solve novel problems.  How can you make your business or  
project more profitable and effective?  These talks focus on those  
questions, and more.  The sub-tracks on cloud computing and government/ 
large enterprise implementations are creating quite a bit of interest  
this year, and the speakers have extensive practical advice to  
dispense on all topic areas.

  * Trade information - Open Source isn't just the software.  To a  
large degree, our user and development community members cooperate  
with each other to solve all kinds of problems.  Ask others about  
their experiences, and offer your own in the informal setting around  
the conference.  The market around Open Source software doesn't just  
have code as its only currency!  The conference is for information  
exchange, and this just might be the most valuable thing you take home.

  * Vendor area - check out the new technologies from hardware  
vendors, software vendors, and service providers! The market changes;  
make sure you know what the most current methods and products are.

  * Put names to faces - that person you've been talking with for a  
year on IRC or IM but have never met?  That consultant whose emails  
you've been reading on the mailing list?  That customer you've been  
trying to get pay attention to you?  Chances are good they'll be at  
AstriCon, and having that face-to-face conversation is sometimes the  
trigger you need to get a project going.

  * Meet entirely new people - the best experiences at AstriCon come  
from the most unexpected places.  That person you sit beside in a  
talk, the lunch table you share with others, the person in the  
elevator with you - the interactions you have will expose you to new  
people whose projects will amaze and interest you, and possibly even  
lead to your changing your methods or finding new business.

I really hope I see you there!

JT


---
John Todd   email:jt...@digium.com
Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW -  Huntsville AL 35806  -   USA
direct: +1-256-428-6083 http://www.digium.com/




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Re: [asterisk-users] RESET CDR

[Todd Routhier]

billsecs is a field in the CDR, it's already there.. Just don't bill based
on the duration field, bill based on the billsecs field and you should have
what you want.


On Wed, Sep 9, 2009 at 11:03 AM, B.Masoud @ SH  wrote:

> Can you provide me some code for that?
> I am NOOB
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
> Backeberg
> Sent: Wednesday, September 09, 2009 5:31 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] RESET CDR
>
> On Wed, Sep 9, 2009 at 10:12 AM, B.Masoud @ SH wrote:
> > I don't want to bill the first 30 seconds, that's all, why is that so
> > strange??? My line does not support polarity reversal, so I don't want to
> > bill for ringing the line...
> > Do you suggest different way than this?
>
> yes. Subtract 30 seconds from the billing when the call is completed.
>
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[asterisk-users] Custom CDR Help

[Todd Routhier]

So, it's been a while and I am just lately getting back into Asterisk stuff.

I am trying to remember/understand how CDR works and after lots of trial and
error and searching the archives, google etc, I am stuck and have a few
questions.

I have setup custom CDR and am trying to figure out the following:

1- Can I just set a variable like set(SomeVar=SomeValue) Then stick this is
cdr_custom.conf${SomeVar} and expect ${SomeVar} to be written out to the
CDR? My thought is that it won't work because only predetermined variables
for CDR can be set but not new variables created.

2- CDR userfield is useful to a point. The problem is we are using
Agents/Queues with the option to write the wav file for the call recording
to the userfield. So, whatever we write in the userfield gets overwritten by
the recorded wav file, file name. Is there anyway to keep the usefield from
being overwritten other than by turning off the link option in agents.conf?
Also, is there anyway to write multiple delimited items in the CDR userfield
at different times within the dialplan by appending instead of overwriting
everytime?

3- We are using agents/queues as mentioned above. Each time the queue rings
an agent another CDR record is created in addition to the other CDR records
that are created during the same call. If the queue ends up ringing 3 agents
before the call is answered, we end up with 3 useless (for our needs) cdr
records. Is there a way to keep those CDR records from being generated? We
already tried NoCDR() at the point just before the agent extension is dialed
but no joy.

Thanks in advance for any help the list can provide with one or more of
these questions. Just need a little help knocking the rust off :)

Regards,
 Todd R.
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[asterisk-users] setvar=CDR(accountcode)=${EXTEN} in sip.conf ???

[Todd Routhier]

Trying to do something like this in the sip.conf under my incoming provider
profiles:

setvar=CDR(accountcode)=${EXTEN}

It seems to show up in the CDR but it's showing up exactly like this
"${EXTEN}".

Is there a way to stuff the DNIS (number dialed) into the accountcode for
CDR?

I have already accomplished setting on a number by number basis, I just want
to do it globally for all number when they come in.

Thanks in advance.
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[asterisk-users] New Languages: Call for contributions

[John Todd]

(also posted today on http://blogs.digium.com/2009/09/02/new- 
languages/ )

Asterisk is being used all over the world, in dozens or even hundreds  
of nations, in a huge variety of linguistic settings.

Until now, the official Asterisk distribution has come in only three  
language “flavors” – English, French, and Spanish.  We are long  
overdue for getting more languages into the “main” Asterisk  
distribution, and over the past few weeks there has been quite a bit  
of work done getting licensing and practical concepts understood to  
the point where we are comfortable with expanding the number of  
available languages at the discretion of the community.

There has been a document submitted for inclusion with Asterisk which  
outlines the protocol process, practical requirements, and license  
criteria for having a new language submitted to Asterisk as part of  
the official distribution.  It should come as no surprise that we’re  
asking for all contributions to be in the Creative Commons v3.0 Share- 
Alike/Attribution licensing regime, as this is clearly the best (or  
only) method for distributing works such as audio recordings with an  
open-source package such as Asterisk.  We’re also insisting that the  
talent that creates any language files be available for others to  
hire, so that there does not become a bottleneck with new prompts for  
others who wish to expand the range of recordings.  Lastly among the  
important notes is that in the rare instances where we have new  
prompts as part of the “core” package requirements, anyone who has  
submitted a language package is under a non-binding community  
commitment to get the new prompts created in their language for  
addition.  (This is a rare event, so hopefully is not overly  
burdensome to contributors.)  This is truly a community participation  
request – there are far too many languages in the world for this to  
work without being almost entirely contributed by active Asterisk  
users and developers.

The complexities of adding new languages is significant – there are  
intricacies in the “say.c” sections of code which determine how  
numbers and dates are pronounced.  There are differences in the way  
voicemail prompts are created for playback.  New languages may not be  
functionally complete if they require code to handle certain nuances  
of sentence structure, and the inclusion of new language audio files  
does not mean that they will be sensible in that particular language  
even if accepted.  However, the first step is to get the language  
recordings in there, and then others can come in and correct the code  
once they have half the puzzle in their hands – that’s the spirit of  
open-source!

There are at least 35 language or dialect versions already existing in  
third-party repositories 
(http://www.voip-info.org/wiki/view/Asterisk+multi-language 
) and of those there are probably a quarter that have more than one  
voicing in male or female talent formats.  I’d love to see the  
majority of those find their way into Asterisk as selectable language  
options.  If you know the person that has created one of these  
language sets, please forward them the new language guideline link  
below!  I’ll be trying to contact all of the language contributors,  
but often there are linguistic barriers or dead-ends for contact data.

To read the requirements and to get started on your language  
contribution to Asterisk, see this document which will soon be part of  
the Asterisk standard distribution: Asterisk Language Submisson  
Criteria, part of issue #15771.

JT

References:

https://issues.asterisk.org/file_download.php?file_id=23667&type=bug

https://issues.asterisk.org/view.php?id=15771

---
John Todd   email:jt...@digium.com
Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW -  Huntsville AL 35806  -   USA
direct: +1-256-428-6083 http://www.digium.com/




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Re: [asterisk-users] List Access

[John Todd]

On Aug 31, 2009, at 3:01 PM, David @ULC wrote:

> To view the post and reply , I always to use below link..
>
> http://lists.digium.com/pipermail/asterisk-users/2009-August/thread.html
>
> Any better way to access the forum ?



There are a variety of other locations that subscribe to the asterisk- 
users mailing list and archive in their own formats - some quick  
Google searching will find them, but honestly I don't keep a list  
since I use the "canonical" source, which is lists.digium.com.

You may want to use Gmail and create your own archive - I know quite a  
few people like Gmail's threading capabilities for list reading/ 
archiving.  Just get a Gmail account and then filter your asterisk-*  
into a separate folder.

JT

---
John Todd   email:jt...@digium.com
Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW -  Huntsville AL 35806  -   USA
direct: +1-256-428-6083 http://www.digium.com/




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Re: [asterisk-users] Error loading module 'res_config _odbc.so'

[Todd Fulton]

Hi,Excellent!  Thank you so much for the tip.  I updated /etc/asterisk/modules.conf and un-commented the following lines:    preload => res_odbc.so    preload => res_config_odbc.soAnd I was good to go.  Much appreciated.Todd


 Original Message 
Subject: Re: [asterisk-users] Error loading module 'res_config_odbc.so'
From: Tilghman Lesher 
Date: Fri, August 28, 2009 5:47 pm
To: "Asterisk Users Mailing List - Non-Commercial Discussion"



On Friday 28 August 2009 18:46:16 Todd Fulton wrote:
> Hi,
>
> I'm getting the following at asterisk startup.  OCBC was working with
> 1.4 no problem, but now under 1.6 (I've tried 1.6.1, 1.6.2-beta3/beta4)
> I can't seem to get rid of this  anyone?
>
>  WARNING[32664]: loader.c:385 load_dynamic_module: Error loading
> module 'res_config_odbc.so':
> /usr/lib/asterisk/modules/res_config_odbc.so: undefined symbol:
> ast_odbc_clear_cache
>
>
> Any help would be greatly appreciated ...

Make sure that you load res_odbc.so first.

-- 
Tilghman & Teryl
with Peter, Cottontail, Midnight, Thumper, & Johnny (bunnies)
and Harry, BB, & George (dogs)

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Re: [asterisk-users] cannot run agi scripts

[Todd Fulton]

Hi,Did you start your agi server?  It should be listening on port 4573 in order for asterisk to be able to connect.  Once you do start the server, telnet to localhost 4573 (or whatever your agi host is) and you'll know if its available.Todd


 Original Message 
Subject: [asterisk-users] cannot run agi scripts
From: Michael Connors 
Date: Sat, August 29, 2009 5:29 am
To: asterisk-users@lists.digium.com

Hi,I am new to Asterisk, I installed it add got it working for incoming calls using a sip provider.I can for example run the following successfully:exten => 124,1,Wait(1) exten => 124,2,Playback(demo-thanks)exten => 124,3,HangupMy problem is that I can not run AGI scripts, I tried the default test-agi.agi and a more simple python based one. I am using the following to use AGI. exten => 124,1,wait(1)exten => 124,2,AGI(hello.agi)exten => 124,3,HangupThe result of this is that the call goes straight to the SIP providers voice mail system, and it does not register in my /var/log/asterisk/messages log file. My AGI scripts are stored in /usr/lib/asterisk/agi-bin, which is read, write and execute for all users, as are all the scripts in the directory.In my asterisk.conf I have the following line: astagidir => /usr/lib/asterisk/agi-binI am running Asterisk 1.4.17~dfsg-2ubuntu1Any help would be greatly appreciated.Regards, -- Michael ConnorsLeiden 2313 HZ,The Netherlands  ___
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[asterisk-users] Error loading module 'res_config _odbc.so'

[Todd Fulton]

Hi,

I'm getting the following at asterisk startup.  OCBC was working with
1.4 no problem, but now under 1.6 (I've tried 1.6.1, 1.6.2-beta3/beta4)
I can't seem to get rid of this  anyone?

 WARNING[32664]: loader.c:385 load_dynamic_module: Error loading
module 'res_config_odbc.so':
/usr/lib/asterisk/modules/res_config_odbc.so: undefined symbol:
ast_odbc_clear_cache


Any help would be greatly appreciated ...

Todd


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[asterisk-users] realtime voicemail and imap user

[Todd Fulton]

Hi,Does anyone know which database columns I would use to configure the imap user/password in a Realtime voicemail table?Todd

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