Re: [asterisk-users] DTMF not being heard correctly by far end conference system

2011-01-12 Thread Duncan Turnbull
Hi Thorsten

Thanks very much, at this point my preference is rfc2833 but I will try some 
other options. 

The system is generating audible tones (that I can hear), although I think the 
audio is generated by the last sip device in the network so if thats so I don't 
have any control of it. Probably then I have to go to inband to get some 
control back, I am not sure what I lose from this, or change upstream provider 
(although the current provider works from a different system)

Cheers Duncan

On 12/01/2011, at 11:42 PM, Thorsten Göllner wrote:

 As far as I can remember you should take a look at the used codec and this 
 here:
 http://www.voip-info.org/wiki/view/Asterisk+sip+dtmfmode
 
 Some codecs do not feel happy with some seetings for dtmfmode. Perhaps you 
 may comapre these on your 2 boxes.
 
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Re: [asterisk-users] Usage Reports

2010-12-30 Thread Duncan Turnbull
Freepbx really needs its own list but it doesn't seem to have one

But - if you have mysql setup and records being logged then the reports should 
show you usage on a daily, weekly, monthly level. Make sure you built asterisk 
with cdrs logged into mysql - its in the addons 

Cheers Duncan

On 30/12/2010, at 8:36 PM, Ben Schorr wrote:

 We’re using FreePBX 2.8 and there is a Reports tab but it doesn’t seem to 
 actually do anything.  Is there some secret/trick to getting a report out of 
 it that will tell us which extensions are placing calls?  I’ve tried every 
 query on the form that I can think of.  Is the reporting disabled by default 
 or ???
  
 Any tips/pointers appreciated.
  
 Ben M. Schorr
 Chief Executive Officer
 __
 Roland Schorr  Tower
 www.rolandschorr.com
 b...@rolandschorr.com
  
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Re: [asterisk-users] Pop-up for MS Outlook 2007 recommended

2010-10-25 Thread Duncan Turnbull
I think there is a new version of Outcall, the pop up was pretty good, but the 
dialout wasn't ideal in Win 7 , and I believe thats fixed now with good 
integration with 2007 and 2010 

http://code.google.com/p/outcall/

You can buy commercial options from Biocom - who make Outcall
http://www.bicomsystems.com/products/C/P/319/288/

Cheers Duncan

On 26/10/2010, at 8:24 AM, unsero...@aol.com wrote:

 Did you already check Bria? I have not tested it yet but it seems to be very 
 powerful.
 Unfortunately there is no trial version available.
 
 If you will give it a try I would be very interested in your opinion.
 
 http://www.counterpath.com/bria-for-microsoft-outlook.html
 
 Oliver
 
 
 
 -Original Message-
 From: Bruce B bruceb...@gmail.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Mon, Oct 25, 2010 9:10 pm
 Subject: Re: [asterisk-users] Pop-up for MS Outlook 2007 recommended
 
 Great suggestion but unfortunately for this client a proven technology is 
 needed and we don't mind paying a bit for it so once the time is available we 
 might do this the way you suggested.
 
 Thanks
 
 On Mon, Oct 25, 2010 at 2:20 PM, Danny Nicholas da...@debsinc.com wrote:
 From: asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B
 Sent: Monday, October 25, 2010 1:14 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Pop-up for MS Outlook 2007 recommended
  
 Hi Everyone,
  
 Which paid or unpaid commercial plugin is available out there for Asterisk 
 that would do Outlook contacts pop-up that is proven to work great with MS 
 Outlook 2007 and Asterisk 1.6. It would be a bonus to do Dialout as well 
 through the Outlook.
  
 Thanks,
 Bruce
  
 Not specifically what you are looking for, but it is very simple to use 
 Apache/Ajax to make AMI links to launch calls from anywhere.  I would invest 
 30-240 minutes into this method before bothering with the other stuff that is 
 out there.  Also, will make it easier when you eventually jump to 1.8/1.10.
 
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Re: [asterisk-users] .call files with application/data are not generating correct CDR

2010-08-22 Thread Duncan Turnbull
You often don't get cdrs or at least useful ones unless you run the call files 
through a Local channel

You maybe already doing this

Can you check the Master.csv and see if it also is recorded incorrectly there. 
Is this just an issue with mysql cdrs or something else. In my setups which use 
freepbx I haven't had an issue with cdrs and call files if using Local channels 
to call

Cheers Duncan

On 23/08/2010, at 2:11 AM, Andy Beak wrote:

 Hi,
 
 The exact problem that I'm experiencing is described at 
 http://www.spinics.net/lists/asterisk/msg122364.html in an earlier 
 posting to the mailing list, but I could find no replies to it.
 
 I installed Asterisk using Ubuntu's apt-get and then fixed the mysql 
 conf (which doesn't load if you use the default apt-get install 
 asterisk-mysql) by building it from scratch.
 
 I'm using Asterisk as an automated voice messaging system so need to be 
 able to dynamically make .call files which point to different mp3 files.
 
 My calls are now being logged to the mysql database but even if I answer 
 a call it still logs as Not Answered with a duration of zero.
 
 Setting unanswered to either yes or no makes no difference in cdr.conf - 
 the call is still logged as Not Answered if I pick it up.
 
 Really the only way around this I can see is to check the lastapp field 
 instead of the disposition.
 
 Lastapp is set to Dial if the call was really not answered and 
 MP3Player if the call was answered.
 
 I see that there is a known bug in Asterisk and it is suggested to use 
 extension.conf to set up a context rather than using call files.  The 
 problem is that I need to be able to change the MP3 that is played.
 
 Has anybody managed to solve this problem?
 
 Thanks,
  Andy
 
 
 
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Re: [asterisk-users] Opensource Speech recognition for Asterisk

2010-08-21 Thread Duncan Turnbull
The Lumenvox works fine in my limited use, easy to setup, good dictionary 
options but it always depends on your circumstance. 

http://www.lumenvox.com/partners/digium/Asterisk.aspx

Most of it is being really careful in planning the customer experience. The 
technology is secondary to the business analysis focussing on why and what the 
caller wants and making the most easy and efficient method of getting them 
there. 

Voice recognition is a pain for people with accents and poor lines and when 
people have written bad call flows but by making sure you get someone to an 
operator really quickly if you can't work out what they said then you can 
alleviate a few issues. 

The primary advantage of voice recognition is to give more choice to the caller 
and route them through more quickly. If you can't do that or don't need that 
complexity then don't use it 

Cheers Duncan

On 22/08/2010, at 11:09 AM, Zeeshan Zakaria wrote:

 Then may be these big multi-billion dollar corporations should use one of 
 them, with whom we all deal regarding various services, and who put us 
 through these voice recognition time-wasting activity in a hope that the poor 
 caller will eventually give up, or will wait painfully long until one of 
 their agent will get time to attend call in person.
 
 Your experience could be different and better then most, and you have 
 certainly complete right of your own opinion.
 
 Zeeshan A Zakaria
 
 --
 www.ilovetovoip.com
 
 
 On 2010-08-21 6:57 PM, Paul Belanger paul.belan...@polybeacon.com wrote:
 
 On Sat, Aug 21, 2010 at 6:21 PM, Zeeshan Zakaria zisha...@gmail.com wrote:
  I yet have to see ANY...
 
 I disagree, while not Open Source like the OP requested, both Nuance
 and Microsoft Speech Server are nothing to laugh at.
 
 --
 Paul Belanger | dCAP
 Polybeacon | Consultant
 Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
 blog.polybeacon.com
 
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Re: [asterisk-users] Call agent when queue is empty and there is a voicemail left

2010-08-12 Thread Duncan Turnbull
So in broad terms

You need to know when the queue is empty, and when there is voicemail (in a 
generic queue mailbox presumably) and also that you haven't already delivered 
the voicemail, and probably that when you deliver the mail its been 
successfully been heard and actioned.

Are you also emailing the voicemail as what happens once you have played it - 
do you delete it assuming its actioned or leave it there, 

So you need a trigger when the queue is empty, could be an AGI checking on 
hangup or an event from the manager to check the queue count.

I haven't done much with AGIs but would think that using a DeadAGI on hangup 
(in 'C' as Steve suggests for speed) you could check queue size, check 
voicemail then trigger an event that triggers the calls. You may not want the 
call direct in the AGI as you want that to finish but you could either use an 
async agi or write a call file or use the manager interface

Do you actually want them to get the voicemail prompts or do you just want to 
find the voicemail messages and play them directly. 

So I don't know if it helped or added more questions but good luck

Cheers Duncan

On 12/08/2010, at 8:42 PM, Jonas Kellens wrote:

 Anyone has an idea of implementation ?!
 
 Jonas.
 
 
 On 08/10/2010 09:04 AM, Jonas Kellens wrote:
 
 Hello list,
 
 situation :
 
 1. incoming calls come into a queue
 2. there is 1 agent logged in into the queue (not always the same agent)
 3. when the caller is in the queue, he has the option to quit the queue and 
 leave a voicemail message
 
 what I want :
 
 when there are no more callers in the queue with who the agent has to be 
 connected, Asterisk needs to call the agent and connect him with his 
 voicemail IF there is a voicemail left.
 
 
 The caller can press 8 to exit the queue and go to the part in the 
 dialplan that does voicemail. I thought about creating a call file when a 
 voicemail is left.
 
 But how will I know when to connect the agent with the voicemail ?
 How will I know the queue is empty and so the call file may be executed ?!
 
 
 I'm not asking for dialplan examples, just a description of how you guys see 
 the above implementation.
 
 
 Kind regards,
 
 Jonas.
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Re: [asterisk-users] AMD setup in Astersik

2010-08-07 Thread Duncan Turnbull
You can include the label of the context in the custom area instead of 
including a different context

i.e. [ext-queues](+)

http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf

Not sure if it affects the order of processing or if that matters

Cheers Duncan

On 8/08/2010, at 7:43 AM, Rushikesh wrote:

 Hi,
 
 You can use /etc/asterisk/extensions_override_freepbx.conf  file if you 
 dont want your dialplan to get overridden.
 
 
 Regards,
 Rishi
 
 On Saturday 07 August 2010 07:36 PM, Tino wrote:
 
 In my Asterisk server following things have been done to detect 
 answering machines before the answered call connects to the agents in 
 queue.
 
 In extension_additional.conf
 
 ==
 [ext-queues]
 include = ext-queues-custom
 exten = 5000,20,Macro(user-callerid,); changed the priority to 20
 ...
 ==
 
 In extension_custom.conf  added following amd dialplan
 
 ===
 [ext-queues-custom]
 exten = 5000,1,Answer()
 exten = 5000,n,AMD(2500|1500|300|5000|120|50|4|384)
 exten = 5000,n,GotoIf($[${AMDSTATUS} = MACHINE]?machine:human)
 exten = 5000,n(machine),Verbose(3, We found an answring machine)
 exten = 5000,n,Set(AMP=${CALLERID(num)})
 exten = 5000,n,Set(date=${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
 exten = 5000,n,System(not showing the actual command)
 exten = 5000,n,Goto(ext-queues,5000,20)
 exten = 5000,n(human),Verbose(3, We've got a human on the line!)
 exten = 5000,n,Goto(ext-queues,5000,20)
 ===
 
 This setup is working fine but the problem is that when i reload 
 freepbx,  extension_additional.conf will go to its original form
 and the changes made will be lost. Is there any way to make the 
 changes in extension_additional.conf conf permanent . Or is there any 
 alternative method for this ?
 
 
 
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Re: [asterisk-users] FYI: Seen the 2600Hz announcement?

2010-08-03 Thread Duncan Turnbull
No its a split

FreePBX is still the same, V3 is still the same, this is a fork from some guys 
who had got involved (or maybe paid some money)

Cheers Duncan

On 4/08/2010, at 2:56 AM, Tzafrir Cohen wrote:

 On Tue, Aug 03, 2010 at 02:28:15PM +0100, Alan Lord (News) wrote:
 http://gigaom.com/2010/08/03/2600hz-project/
 
 So practically FreePBX V3 was renmed 2600Hz / BlueBox ?
 
 -- 
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 icq#16849755  jabber:tzafrir.co...@xorcom.com
 +972-50-7952406   mailto:tzafrir.co...@xorcom.com
 http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
 
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Re: [asterisk-users] dial extension and play sound file from shell on asterisk server?

2010-04-08 Thread Duncan Turnbull
Have a look at the call files examples of voipinfo

http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out

Its not too hard to do what you want

Cheers Duncan

On 8/04/2010, at 11:00 PM, Brian J. Murrell wrote:

 I want to use Asterisk as a general message delivery system here.
 
 That is, I want to be able to have a (shell, perl, etc.) script on my
 Asterisk server dial an extension, wait for it to be answered and then
 play a sound file and then hang up, or even wait for a response or
 reactions to some IVR.
 
 Certainly if I had a SIP library, I could have the script simply look
 like a SIP extension but that seems like it should be an unnecessary
 added complication.  Instead I am looking for a more direct API to
 Asterisk which allows a process to interact like a phone but not
 actually be one.
 
 Is this at all possible?  Anyone done anything like this?
 
 b.
 
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Re: [asterisk-users] Asterisk system for church call center

2010-03-29 Thread Duncan Turnbull
Hi Frank

I have found Freepbx on top of Asterisk a good solution for the church I look 
after and the rest of my customers, the callcentre functions you need are built 
in it and if they have someone technical then they can expand what they are 
doing

It has both queues and ring groups (which are often all they need) 

I would imagine you just send them to an IVR or mailbox to ask for their name 
and details then move them into a higher priority second queue 

Elastix has Sugar in it I believe and looks okay but I like debian/ubuntu as a 
base distribution rather than Centos so haven't gone that way.

However most of my customers still struggle with the concepts involved in 
telephony so while they are happy to look at it while I am there they forget 
quickly how to drive it or lose their nerve, especially the church which is 
faith focussed rather than tech focussed ;-)

Because of that I think you want the easiest system for you to maintain 
remotely and elastix or freepbx is pretty easy. It also allows you to say this 
is available, this is not which is useful in narrowing down requirements. 

Cheers Duncan
On 30/03/2010, at 9:58 AM, Frank Church wrote:

 I have been asked by my church to recommend a VoIP system which can do
 the following.
 
 They do internet radio shows which are sometimes broadcast on radio.
 
 They are looking for a system which does the following for about 5
 agents, exactly as they have described it.
 
 1. Take incoming calls
 
 2. Put them on hold if there is no one to handle the call immediately,
 or transfer them to an available agent
 
 3. Take down their details, and number, (if this can be retrieved and
 saved from the caller id, thats better)
 
 4. Get them to hold on after taking their details if they still want to hold
 
 5. Call them back when the backlog is cleared up.
 
 I have a fairly good grasp of the hardware and programming part of
 Asterisk, having compiled it more than a few times and implemented
 A2Billing phone card and call shop system with it.
 
 But the type of software suited to the Call Center side is where my
 knowledge gap lies.
 
 I am looking for solutions based on the usual Asterisk distributions
 like AsteriskNow, trixbox, elastix etc, whether ready packaged or
 requiring additional customization.
 
 
 The matter of whether they will use soft phones, or regular phones
 with headsets is also something to consider. Soft phones with good
 GUI's may be preferred if more cost effective for them, although my
 personal preferences are with hard phones.
 
 Any recommendations - the ease of software for the end users is the
 main thing for me, and integration with the database for taking
 customers details is the main thing for me. One of the distributions
 with SugarCRM comes to mind here.
 
 Sorry for cross-posting, but ready made and commercially supported
 systems are not ruled out, if they come within their budget.
 
 Regards
 
 
 Frank Church
 
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Re: [asterisk-users] No RTP from asterisk?

2010-02-28 Thread Duncan Turnbull

On 1/03/2010, at 2:41 PM, Peter Serwe wrote:

 I checked the firewall, iptables -L showed no rules whatsoever.  No other 
 traffic has indicated it was blocked, iptables was set in allow all 
 everywhere mode.
 
 I went ahead and turned it off, still don't have RTP.  No audio either 
 direction via lines registered.
 
 G729 is completely disabled from all trunk groups and users, only using G711 
 at this point.
 
 Peter
 
 
The asterisk rtp debug should show if asterisk is sending audio or receiving 
packets but its not nearly as useful as tcpdump. 

tcpdump udp port 5060 -s0 -A will give you all the SIP. 

But just dumping all traffic between asterisk and the host will give you a view 
on RTP - you should see it take off when a call is setup if its not blocked

You should see a SIP Invite to setup a call with the audio destination - this 
should be your asterisk box and the far end depending on who is doing what. You 
should look at the address and also whether both sides are providing a mutually 
acceptable audio formats. If there are no agreed audio formats you won't get 
rtp. The c= in the session setup indicates the addresses each site is using for 
media. 

Cheers Duncan


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Re: [asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory

2010-02-10 Thread Duncan Turnbull
The other way on Debian/Ubuntu is just to test the existence of the dir and 
create it if needed

If you add this to the /etc/init.d/asterisk near the start you should be fine

if ! [ -d /var/run/asterisk ] ; then
mkdir /var/run/asterisk
chown $AST_USER.$AST_GROUP /var/run/asterisk
exit 0
fi

Set the ownership as required 

Cheers Duncan

On 11/02/2010, at 7:50 AM, Brian wrote:

 On Wed, 2010-02-10 at 11:24 -0600, Jason Parker wrote:
 Brian wrote:
 Each time the server is rebooted Asterisk duly
 deletes the manually created /var/run/asterisk directory - quite why it
 does this I just don't know - perhaps it is a bug?
 
 
 Your assumption is incorrect.  Some Linux distributions will empty /var/run/ 
 on 
 boot, just as they do with /tmp/.  
 Thanks Jason - that had never dawned on me, but I've just tested it and
 indeed it does.
 
 I do believe you're right, however, in 
 suggesting that there is a bug in Asterisk.  It appears that Asterisk 
 creates 
 /var/run/asterisk/ during install and assumes that it will always exist.
 Agreed - that would make sense that by default it thinks the directory
 is there. The workaround / fix is to take out the (!)
 from /etc/asterisk/asterisk.conf and allowing the default setting of:
 astrundir = /var/run to come into play. It then puts the .pid and .ctl
 in the root of /var/run
 
 Some of the sample init scripts (Debian) create that directory before 
 starting 
 Asterisk.  This should be done in all of them (or in Asterisk itself, 
 maybe?).
 The one I had didn't - but I could have added it. I just wanted to be
 sure I was doing the right thing.
 
 Please report an issue on http://issues.asterisk.org/
 Done - but I'm a bit embarrassed as it seems so trivial.
 
 Thank you for your help.
 
 
 
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Re: [asterisk-users] Mitel integration

2010-01-27 Thread Duncan Turnbull
Having looked at the outputs into PMS they are very simple stop start records. 
Line by line text that can easily be recreated. They have about 4-5 fields, 
origin number, destination, time of call,  duration, or similar things

Usually they go out via a serial port or TCP port expecting a terminal to 
receive them so plugging into them will quickly show you what you need.

Its not that you need to match the Mitel, you need to match the PMS. Best to 
talk to them but I have looked at it for a couple of customers who are still 
deciding and helped them fix their PBXs when they broke and its pretty straight 
forward. You just need to be able to output to either a serial or TCP port .

Cheers Duncan

On 28/01/2010, at 6:01 AM, Jeff LaCoursiere wrote:

 
 
 On Wed, 27 Jan 2010, Mark Wiater wrote:
 
 the mitel 3300 sends SMDR on TCP 1752.  It spews software and hardware 
 logs in the same manner, different ports.
 
 This particular model (need to get the model number) has a serial 
 connection.  I'm all for putting a serial sniffer between them (if they 
 let me!), but was really hoping someone had already done this and could 
 give me a headstart.
 
 I'll investigate the ethernet options, though, as that would make more 
 sense anyway!  If the PMS will talk over ethernet I'll try to pretend to 
 be a 3300.
 
 Cheers,
 
 j
 
 
 On 1/27/2010 11:00 AM,  Steve Howes said:
 On 27 Jan 2010, at 15:48, Jeff LaCoursiere wrote:
 Sounds good to me, but without the spec I'm stuck in a catch 22!
 
 tcpdump? (assuming IP). Bet its fairly simple plain text or something.
 
 Steve
 
 
 
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[asterisk-users] Upgrading an AA50 and finding zap drivers - lack of Digium support

2009-12-23 Thread Duncan Turnbull
Hi there

I have a client who has an AA50 from DIgium. I am really challenged getting any 
support as the client doesn't have any of the original registration or 
subscription info, someone did the install and left without any records. I 
thought okay we can ask Digium, but you can't get help wthout registering your 
product and you can't register your product without your subscription info. 

I did get one response which was to email customer services and eventually 
found an email address for them but that seems to have fallen on deaf ears. 
Perhaps my expectations are too high but it was an email a week ago and no 
response, not even to say go away.

Its really disappointing and amazingly hard to believe such a system exists. 
Perhaps no one else in the world loses their registration info.

I need to upgrade the AA50 to the latest firmware, and just get some general 
support for the setup, as it doesn't seem to have picked up Zap and I think 
perhaps the CF card has died. Does anyone have any pointers as who to talk to. 
I see lots of digium people on the list so I am hoping someone can help me. 

Thanks very much

Merry Xmas

Cheers Duncan
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Re: [asterisk-users] Upgrading an AA50 and finding zap drivers - lack of Digium support

2009-12-23 Thread Duncan Turnbull
Thanks very much Kevin

But its not that clear - in fact the support email address isn't listed on the 
support page - everything leads to logging in with your registered product. Its 
incredibly frustrating and I recommend you try looking and seeing how it works 
for yourself if you haven't got your registration details. Having some method 
to handle missing details would have sorted this all out but I get stuck in a 
big loop.

You are probably right I should have called but I would have thought email 
would be possible at least. It wasn't a rush for a while. Also I am not in the 
US so I don't tend to call overseas as standard as most companies have ways of 
getting to support and need emails with details anyway.

I agree a 4 day holiday is approaching but a week ago it wasn't and apart from 
an automated response not even a whiff of a reply

Cheers Duncan


On 24/12/2009, at 9:16 AM, Kevin P. Fleming wrote:

 Duncan Turnbull wrote:
 
 I did get one response which was to email customer services and eventually 
 found an email address for them but that seems to have fallen on deaf ears. 
 Perhaps my expectations are too high but it was an email a week ago and no 
 response, not even to say go away.
 
 The contact information for Digium's support department is clearly
 listed on our web site. There are multiple methods, including just using
 the plain old PSTN to call us :-)
 
 Its really disappointing and amazingly hard to believe such a system exists. 
 Perhaps no one else in the world loses their registration info.
 
 I need to upgrade the AA50 to the latest firmware, and just get some general 
 support for the setup, as it doesn't seem to have picked up Zap and I think 
 perhaps the CF card has died. Does anyone have any pointers as who to talk 
 to. I see lots of digium people on the list so I am hoping someone can help 
 me. 
 
 Given the serial number of the unit the support department will be able
 to research the registration and get you squared away; the only issue is
 that we are rapidly approaching a four-day holiday weekend at Digium so
 you would likely not get any response until early next week.
 
 -- 
 Kevin P. Fleming
 Digium, Inc. | Director of Software Technologies
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 skype: kpfleming | jabber: kpflem...@digium.com
 Check us out at www.digium.com  www.asterisk.org
 
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Re: [asterisk-users] GUI for hunt groups?

2009-10-28 Thread Duncan Turnbull
Freepbx comes with setup of ring groups and queues with different hunt 
strategies

Also it has Flash Operator Panel which gives you the state of the system 
in real time graphical format

No money - just a small bit of installation time and learning how to use it

Cheers Duncan

Ken D'Ambrosio wrote:
 Hi, all.  I've got an Asterisk box installed that I'd really like to
 leverage -- and installing a GUI for hunt groups would be awesome.  So
 long as I can have a trial copy, I could even pay money.  It would have to
 be able to make use of both SIP and ZAP extensions.

 Suggestions?

 (Note: I wouldn't much care about the GUI, myself, but my boss is all over
 one.)

 Thanks!

 -Ken


   

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Re: [asterisk-users] Today's problem: Inbound call routing

2009-10-09 Thread Duncan Turnbull
Usually that message comes up because the caller is anonymous and 
freepbx doesn't like anonymous calls by default.

There is an option to accept anonymous calls, or set the incoming trunk 
to accept calls from the specific IP address

Of course it could be something else

Cheers Duncan

Ben Schorr wrote:
 Sorry, I'm brand new at Asterisk (and/or FreePBX).  I'm going to have to
 figure out what all those things are before I can show them.

 I'll have to get back to you.

 Ben M. Schorr
 Chief Executive Officer
 __
 Roland Schorr  Tower
 www.rolandschorr.com
 b...@rolandschorr.com


   
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
 boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
 Sent: Friday, October 09, 2009 9:54 AM
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Today's problem: Inbound call routing

 On Fri, Oct 09, 2009 at 06:15:43AM -1000, Ben Schorr wrote:
 
 -Original Message-
 
 Too simple, apparently, when I dial the number the caller gets a
 recording that it's a non-working number and this is what I see
   
 in
   
 the
   
 CLI:

 Extension '8085255935' in context 'default' from '808xxx'
   
 does
   
 not
   
 exist.  Rejecting call on channel 0/1, span 1

   
 That is a pretty clear error message.
 
 Yes, I thought so.  But how do I fix it?

   
 So...other than creating the inbound route and assigning it to
   
 an
   
 extension I apparently have to do something else.  Any
   
 suggestions
   
 as
   
 to what that might be?
   
 You manage your dialplan with FreePBX. This mailing list supports
 
 Asterisk. I
   
 have no problem with questions about FreePBX systems. But they
 should also be phrased as Asterisk questions. This is a FreePBX
 
 question.
   
 I see, so this isn't an Asterisk problem it's a FreePBX problem?
   
 Creating an inbound route is FreePBX speak. This is a FreePBX
 
 question.
   
 Please ask an Asterisk question.

 For instance, show a dialplan trace, show the respective dialplan,
 
 show the
   
 respective channel configuration.

 --
Tzafrir Cohen
 icq#16849755  jabber:tzafrir.co...@xorcom.com
 +972-50-7952406   mailto:tzafrir.co...@xorcom.com
 http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] Upgrading Asterisk in Trixbox installation

2009-08-31 Thread Duncan Turnbull
I am a big fan of ubuntu LTS and freepbx and recently I saw mention of a 
custom module to add auto configuring endpoints for linksys (but i cna't 
find it again right now)

Trixbox had too much stuff whereas the source install of just what you 
want is nice and clean

Cheers Duncan

Jeff LaCoursiere wrote:
 On Mon, 31 Aug 2009, ilker Aktuna wrote:

   
 Thank you.
 That was quick and helpful :)

 Then I'll just make and make install
 What should I backup, in case of rollback requirement ?
 

 That's a bit tougher.  At the least /usr/lib/asterisk/modules, 
 /etc/asterisk, and /usr/sbin/asterisk...  someone else may need to chime 
 in here...

 I've always been a fan of trixbox, and I have done a lot of installations, 
 but when it comes down to it all I really want it for is for a quick 
 installations of asterisk and FreePBX.  I don't think I actually use any 
 of the trixbox-only features.  I've also been enamored with Ubuntu of 
 late, and have dumped CentOS.  YMMV, but you might consider starting over 
 with a clean build of the linux of your choice, and doing asterisk + 
 addons + FreePBX from source.

 j

   
 Thanks.


 - Original Message -
 From: Jeff LaCoursiere j...@jeff.net
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Monday, August 31, 2009 11:15 PM
 Subject: Re: [asterisk-users] Upgrading Asterisk in Trixbox installation


 
 On Mon, 31 Aug 2009, ilker Aktuna wrote:

   
 Hi,

 My Trixbox 2.8.0.1 installation includes the following Asterik version:
 1.6.0.9-samy-r27

 I am having some problems with it and I think they might be solved if I
 use the latest Asterisk version.
 Is it a good idea to update Asterisk in Trixbox externally ?
 
 I've done it in the 1.4 branch.

   
 Is it safe ?

 
 Should be, as long as you stay within the same branch.  That being the
 case, I would stick with 1.6.0.14 if I were you.  Make sure you don't
 make samples :)

 j

   
 If so, which version should I prefer ?
 1.6.1.5 or 1.6.0.14 ?

 Thanks,
 ilker
 
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Re: [asterisk-users] outbound calls not ringing

2009-08-19 Thread Duncan Turnbull
Generally with FreePBX the ring options are set in the General Options - 
you can set the Dial options which are normally tr, but I guess that 
isn't working for you.

The SIP files you could edit would have custom in their name, otherwise 
your changes will be overwritten when you reload freepbx

You could put this in sip_general_custom.conf which will be included

Cheers Duncan

John A. Sullivan III wrote:
 Oops! - You're using FreePBX - someone who knows more about FreePBX will
 have to help you as I don't.  May I also suggest that you bottom post in
 future responses rather than top post; that makes it a little easier to
 follow.  Good luck - John

 On Wed, 2009-08-19 at 16:59 +, Ott Rose wrote:
   
 here is my sip.conf. i don't see it.
 ;;
 ; Do NOT edit this file as it is auto-generated by FreePBX. All
 modifications to ;
 ; this file must be done via the web gui. There are alternative files
 to make;
 ; custom modifications, details at:
 http://freepbx.org/configuration_files   ;
 ;;
 ;

 [general]

 ; These files will all be included in the [general] context
 ;
 #include sip_general_additional.conf

 ;sip_general_custom.conf is the proper file location for placing any
 sip general
 ;options that you might need set. For example: enable and force the
 sip jitterbuffer.
 ;If these settings are desired they should be set the
 sip_general_custom.conf file.
 ;
 ; jbenable=yes
 ; jbforce=yes
 ;
 ;It is also the proper place to add the lines needed for sip nat'ing
 when going
 ;through a firewall.  For nat'ing you'd need to add the following
 lines:
 ; nat=yes , externip= , localhost= , and optionally fromdomain= .
 ;
 #include sip_general_custom.conf

 ;sip_nat.conf is here for legacy support reasons and for those that
 upgrade
 ;from previous versions.  If you have this file with lines in it
 please make
 ;sure they are not duplicated in sip_general_custom.conf, if so remove
 them
 ;from sip_nat.conf as sip_general_custom.conf will have precedence.
 #include sip_nat.conf

 ;sip_registrations_custom.conf is for any customizations you might
 need to do to
 ;the automatically generated registrations that FreePBX makes.
 ;
 #include sip_registrations_custom.conf
 #include sip_registrations.conf

 ; These files should all be expected to come after the [general]
 context
 ;
 #include sip_custom.conf
 #include sip_additional.conf

 ;sip_custom_post.conf If you have extra parameters that are needed for
 a
 ;extension to work to for example, those go here.  So you have
 extension
 ;1000 defined in your system you start by creating a line [1000](+) in
 this
 ;file.  Then on the next line add the extra parameter that is needed.
 ;When the sip.conf is loaded it will append your additions to the end
 of
 ;that extension.
 ;
 #include sip_custom_post.conf


 
 From: jsulli...@opensourcedevel.com
 To: asterisk-users@lists.digium.com
 Date: Wed, 19 Aug 2009 12:17:15 -0400
 Subject: Re: [asterisk-users] outbound calls not ringing

 sip.conf

 On Wed, 2009-08-19 at 15:55 +, Ott Rose wrote:
   
 we are using Aastra 57i

 i don't see that setting. where is it at?

 
 From: jsulli...@opensourcedevel.com
 To: asterisk-users@lists.digium.com
 Date: Wed, 19 Aug 2009 11:07:21 -0400
 Subject: Re: [asterisk-users] outbound calls not ringing

 On Wed, 2009-08-19 at 13:54 +, Ott Rose wrote:
   
 I put a post on here about my issues with outbound calls not
 
 ringing
 
 but i haven't resolved it. so i am trying again.

 When i dial any outside number i dont get a ring tone at all.
 
 when
 
 the
 
 person picks up and starts to talk i can hear them fine. it
 
 sounds
 
 great. How do I start to troubleshot this?
 
 snip
 What type of phones are giving you the problem? If I recall
   
 correctly,
 
 our SIP phones had this problem depending on how the destination
   
 handled
 
 signaling. We resolved it by adding progressinband=no (as
   
 opposed to
 
 the default never - at least I think it is the default) but this
 produces the problem of duplicate ring tones at times. Hope this
   
 helps
 
 - John
 -- 
 John A. Sullivan III
 Open Source Development Corporation
 +1 207-985-7880
 jsulli...@opensourcedevel.com

 http://www.spiritualoutreach.com
 Making Christianity intelligible to secular society


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Re: [asterisk-users] Skype for Asterisk???

2009-08-17 Thread Duncan Turnbull
I am using the beta and its pretty good for remote access for clients

It would help if they had some discount structure for volume

Cheers Duncan


Pascal Bruno wrote:
 Not sure if anybody noticed, but it seems like Skype For Asterisk is out.

 $66 per channels, pretty pricey

 http://store.digium.com/productview.php?product_code=1SFA0001
 

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Re: [asterisk-users] Call File Channel

2009-08-12 Thread Duncan Turnbull
If you use a Local channel to dial it then it will fall under the same rules

Channel: Local/numbertod...@the-context-you-want

This gets a CDR produced, it does pay to check everything works the same 
but it should be fine

Cheers Duncan

David Gibbons wrote:

 Context: is what the call is dumped into after it is answered, at 
 extension Extension:. I don’t think it’s related to how the call is 
 placed.

 I can dial the local extension SIP/170 but I’m not sure where that 
 gets me.

 Basically I want to have the same failover that I have for all other 
 outgoing calls on these automatic calls…

 Thanks

 Dave

 *From:* asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny 
 Nicholas
 *Sent:* Wednesday, August 12, 2009 5:17 PM
 *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
 *Subject:* Re: [asterisk-users] Call File Channel

 Ok. Here’s how you would do that:

 Channel: SIP/170 (some local extension)

 CallerID: SIP/104 (another local extension)

 MaxRetries: 1

 WaitTime: 60

 retryTime: 5

 Context: your_context

 Extension: s

 This should create an extension call using your context. The context 
 can then dial out as you write it.

 

 *From:* asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *David 
 Gibbons
 *Sent:* Wednesday, August 12, 2009 4:10 PM
 *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
 *Subject:* Re: [asterisk-users] Call File Channel

 Thanks Danny,

 I do have a dial cmd with multiple arguments in my normal outgoing 
 context. I guess my question really is:

 How do I tell the call file using “Channel: XXX” to use my outgoing 
 context instead of Zap/g1/xx or sip/trunk_x/xx directly?

 -Dave

 *From:* asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny 
 Nicholas
 *Sent:* Wednesday, August 12, 2009 5:05 PM
 *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
 *Subject:* Re: [asterisk-users] Call File Channel

 Exten = s,1,Dial(SIP/trunk_x/#1SIP/trunk_y/#2ZAP/g1/#3,60)

 

 *From:* asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *David 
 Gibbons
 *Sent:* Wednesday, August 12, 2009 3:59 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* [asterisk-users] Call File Channel

 I know I’m missing something here (been a long day)…

 How can I specify more than one channel in a call file?

 I want to dial SIP/trunk_x and fail to SIP/trunk_y and fail to ZAP/g1…

 Thanks

 Dave

 

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Re: [asterisk-users] Anonymous Connection form IP to use specific Context

2009-07-09 Thread Duncan Turnbull
If you create a peer definition and put the host address in it and the 
context you want it to go to you should be fine

Cheers Duncan

David Klaverstyn wrote:

 Hi All,

  

 I never saw a reply to this question.  Is anyone able to assist?

  

 Regards

 David.

  

 *From:* asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *David 
 Klaverstyn
 *Sent:* Friday, 19 June 2009 2:28 PM
 *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
 *Subject:* [asterisk-users] Anonymous Connection form IP to use 
 specific Context

  

 Hi All,

  

 How can I force an anonymous SIP connection from a certain IP address 
 to use a specific context rather than the default one defined in sip.conf.

  

 I am using Asterisk 1.6.0.9

  

 Regards

 *David Klaverstyn*

  

 

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Re: [asterisk-users] about monitored calls storing

2009-06-29 Thread Duncan Turnbull
Trixbox I think uses FreePBX

FreePbx has an option for each extension to set it to record all calls. 
It will record the extension in the file name and you can view it 
through the recordings app if you want a web view.

There are all stored in a common dir /var/spool/asterisk/monitor - you 
can probably mod the code for the recording if you want more info in the 
filename

Cheers Duncan

peace keeper wrote:
 Hello all,
  how can I possibly make the monitoring for all calls through the 
 asterisk, and for those file to be stored with the name of the 
 initiator, in additional to know to whom this call is going, could 
 this functionality be implemented via configurations!

 in other words, could I configure the asterisk so that the 
 administrator to be able to hear calls coming from who going to whom, 
 as a having a record for each call,
 I am using trixbox v2.6.2.1

 should that functionality be implemented by an external application , 
 such as one written using asterisk-java !!!

 any help is appreciated?
 thanks in advance,

 

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Re: [asterisk-users] how to sniff RTP and SIP traffic only

2009-06-29 Thread Duncan Turnbull
For Linux use tcpdump on the host you are after

tcpdump udp and port 5060 or portrange 1-16000 -s0 -i eth0

where 5060 is your SIP port and 1-16000 are your rtp ranges
-s0 means snap length of 0 so capture all the packet rather than cutting 
off at a point

And refine it by adding the host you are targetting and -w to write to a 
file.

Then you can import the file in wireshark and use the voip utlities to 
listen to it fairly easily or use tcpdump -r to read it back and clean 
it out a bit more

Cheers Duncan

Xavier Cardil wrote:
 Hi, do somebody knows how to sniff RTP and SIP traffic only for a 
 faster debugging ?

 Thanks.
 

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Re: [asterisk-users] GSM mobile trunks

2009-06-23 Thread Duncan Turnbull
Yip the VoiceBlue SIP units are very good but a bit pricey

Gordon Henderson wrote:
 On Tue, 23 Jun 2009, Sasa Bobek wrote:

   
 Hi all,
 We have been planing for a long time to set up GSM mobile trunks for
 termination, and were planing on going with analog GSM adapters connected to
 a VoIP gateway.  Should we be concerned with such a set-up as far as voice
 quality and other issues are concerned?  Any experiences with GSM terminal
 chipsets?
 

 Why not SIP based GSM devices? e.g. Portech?

 Gordon

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Re: [asterisk-users] asterisk and openvpn and sip

2009-06-18 Thread Duncan Turnbull
Usually this is a routing error with openvpn setup and asterisk thinking 
it needs to route someway other than the vpn. If the originating packets 
have an external ip address asterisk might send them back out another route

Have a look using tcpdump on the server to see where the returned 
packets are destined

Cheers Duncan

Giorgio Incantalupo wrote:
 Hi all,

 I'm trying to connect one phone to a remote asterisk server via openvpn. 
 First of all, I put the vpn server on the box hosting asterisk and the 
 vpn client on another box, both with public ips.
 Then I set the client ip as my phone IP gateway and the remote pbx ip as 
 the registrar and outbound proxy.

 I see in the phone log register packets are sent but nothing in return. 
 Asterisk console shows it tries to give back the packets but they seem 
 to be lost somewhere.

 I made some tests with my pc setting its gateway with the vpn client IP 
 and I can reach the pbx machine (ping, ssh,...) but sipsak gets no response.
 It seems ping and ssh response packets are correctly routed but sip 
 packets aren't.

 I tried to set nat=yes in sip.conf but without result.
 Is there any asterisk parameter to set to make it work with openvpn?

 Any help really appreciated.

 Thank you.

 Giorgio

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Re: [asterisk-users] Automatic Calling Feature?

2009-06-11 Thread Duncan Turnbull
Not too hard to do,

you can have a script generate a list of call files which automatically 
ring the callers in the list and play a message

http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out

Cheers Duncan

Christopher Stamper wrote:
 Right now, my organization is using a commercial service 
 (OneCallNow.com), that gives telephone notifications to all numbers in 
 a predefined list. Example:

 -Admin records a voice message
 -Service calls each number in the list, and plays the message back to them

 It's a pretty handy service, albeit a bit pricey. I've been wondering 
 if Asterisk could do this for me? I don't really want to have to write 
 scripts, but it would be great if it's already a feature.

 I don't have an Asterisk PBX running yet, but when I do it will 
 probably have multiple T1 PRI lines, making it possible to dial all 
 these numbers (100+) in a reasonable amount of time.

 Anyone know of a way to do this?

 -- 
 Christopher Stamper

 Email: christopherstam...@gmail.com mailto:christopherstam...@gmail.com
 Web: http://tinyurl.com/2ooncg
 gTalk: http://tinyurl.com/6e359r
 Skype: cdstamper
 

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[asterisk-users] Best way to get 60+ analogue extensions.

2009-03-15 Thread Duncan Turnbull
Hi All

I am looking at a replacement for a hotel PBX which requires at least 60 
analogue extensions.

I tend to use Sangoma equipment but haven't tried this many analogue 
extensions before. I am interested in anyone's experience of which 
server platform literally fits and copes well with multiple cards, and 
the choice of Digium vs Sangoma or something else.

I can see the Digium AEX2400 with 24 lines, physically they are all very 
deep, if I had 3 of these in a server it would seem straight forward 
assuming the motherboard doesn't haven't anything get in the way
Equally the Digium TDM2400P supports 24 lines and physically requires 
similar space

The Sangoma A400 provides 24 ports but uses two slots, having 3 of these 
in a server looks like I need to pick the server carefully.

I may need an ISDN PRA inbound but am working hard to have the inbound 
lines via SIP, but if I do that means at least 4 slots on this plan.

I am just interested in any recommendations for server hardware and card 
combinations that are currently in use.

Also if anyone has provided call data out to the RMS system ( 
http://www.rms-global.com/Our-Products/RMS-Hotel/ ) I would be keen to 
hear how it worked.

Thanks very much

Cheers Duncan

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Re: [asterisk-users] Best way to get 60+ analogue extensions.

2009-03-15 Thread Duncan Turnbull
Thanks very much Rob  Stephen

The channel banks look good. I am not sure if they are easily availble 
in NZ but we can get some in I am sure.

Xorom make very positive comments about their astribanks and that you 
can have multiple channel banks on a server so they look pretty good (if 
they are honest). I can't tell the manufacturer of the other channel 
banks you were referring  to.

In Wellington, NZ, PRAs are pretty expensive and a 25Mbit/sec 
symmetrical fibre connection to a SIP provider is a better deal. On some 
of my other customers we have 15 SIP lines without issue using G711 and 
consuming about 80-100k per line if that. But I take the point so will 
revisit it in the design. Another reason for SIP is the Telepermited 
options available are limited over here, so to connect you really want 
to have an approved device in case you have any issues. But with SIP via 
a Provider you abstract that layer which is cleaner.

If we need to have one E1 then having more for the Astribanks sounds fine.

Cheers Duncan

Rob Hillis wrote:

Duncan Turnbull wrote:
  

Hi All

I am looking at a replacement for a hotel PBX which requires at least 60 
analogue extensions.

I tend to use Sangoma equipment but haven't tried this many analogue 
extensions before. I am interested in anyone's experience of which 
server platform literally fits and copes well with multiple cards, and 
the choice of Digium vs Sangoma or something else.
  



You have several options here, however due to the power requirements, I
wouldn't recommend you use either the Sangoma or Digium analogue cards
here - providing ring voltage to that many extensions is likely to
over-tax the power supply in the server.

I'd either be looking at three channel banks (3 24 channel channel banks
would give you a total of 72 analogue channels) or two Xorcom Astribanks
which would likewise give you up to 64 channels.

The Astribanks are probably a cheaper way to go since they connect to
your server via USB rather than T1/E1 ports.  However, I haven't had any
experience with multiple Astribanks connected to the same server, so
there may be issues there that I'm not aware of.  Channel banks are
certainly the proven and reliable technology, but will be significantly
more expensive since they connect to your Asterisk server via T1/E1 links.

  

I may need an ISDN PRA inbound but am working hard to have the inbound 
lines via SIP, but if I do that means at least 4 slots on this plan.
  



You'd need to be very sure of the bandwidth and quality of connection to
your VoIP provider to go with SIP for more than half a dozen channels. 
This kind of connection can easily be far more expensive than a
traditional T1/E1 line, so I wouldn't be pushing so hard for SIP.

If you were to use channel banks, you would most likely end up with a
four port T1/E1 card and would only be using three of those channels,
leaving a spare one for an incoming T1/E1 line.

If you were to use Astribanks, you would have plenty of space in the
server to include a T1/E1 card.

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Re: [asterisk-users] top posting again [was: Re: CDR Design]

2008-12-05 Thread Duncan Turnbull
I like the discussion, I doubt it will end.

I prefer top posting because I reply to all my customers that way, my 
mail client isn't that smart and I think technology should meet the 
needs rather than force you to adopt work arounds.

I can fully understand though others preferring it, but I don't.

All the presented evidence so far suggest bottom posting is a work 
around to a list archive function that is less than ideal or a 
politeness to get around a way of doing things that doesn't really apply 
so much anymore. I would have thought someone could make a better list 
archive model, I don't believe bottom posting is intuitive and therefore 
being picked up by many newcomers to the game.

An alternate is to get a filter that sorts the whole thing out depending 
on preferences ;-), but who can be bothered.

I haven't seen a signup requirement to this list requiring bottom 
posting, and neither have I on the many other lists I am on. In fact if 
I look at most of my lists the majority of posters over time have tended 
to top posting. Doesn't mean its right but it appears to be happening.

Cheers Duncan

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Re: [asterisk-users] Cisco 7906g SIP

2008-10-17 Thread Duncan Turnbull
Hi Salvatore

Have you checked the tftp logs in any event? Its important to check the 
tftp logs and see if anything is being requested.

I have had this before but usually its still trying to grab its first 
couple of files, and from that you can get an idea of where its getting 
stuck. If it says upgrading it means its trying to change from one 
version to another and failing, so you need to go backwards to a version 
it can cope with.

If its not asking for any files then usually what I have done is to go 
to the lowest SIP version 2 or 3 for changing from the call manager to 
SIP and reset the phone to factory defaults and try and get it to start 
the change again

Cheers Duncan

Sasa wrote:
 Hi Duncan,
 yes I have a tftp server (I use also Cisco 7941G that use tftp server for 
 upload configuration) and I know this function, but now my problem is that 
 the phone is stopped on the initial screen that show 'upgrading' and MAC 
 address and the process not continued.
 Thanks.
 
 --
 
Salvatore.
 
 
 
 - Original Message - 
 From: Duncan Turnbull [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Tuesday, October 14, 2008 8:52 PM
 Subject: Re: [asterisk-users] Cisco 7906g  SIP
 
 
 Hi Salvatore

 Do you have a TFTP server that serves the phone configuration files?
 This is very separate to the phone, i.e. on a server/pc somewhere, and
 will log all the file requests it receives. You can check this
 irrespective of the phone

 Have you checked whether tftp requests are being made, usually they come
  before the system goes into the upgrading state.

 I have had that before and it was caused by having different load files
 from that specified in the OS79XX.TXT file which for my phones usually
 have P003-08-6-00 but for upgrading I start from P0S30202

 For SIPDefault.cnf you also need the image version to match
 #Image Version
 image_version:P0S3-08-6-00 ;

 But for conversion I first go to this image
 image_version:P0S30202 ;

 And I go from that to this

 image_version:P0S3-06-2-00 ;

 then to the current version


 And I have these files on my tftpserver which are the respective firmwares

 -rwxr-xr-x 1 root root 753560 2007-04-23 14:36 P0S3-08-6-00.sb2
 -rwxr-xr-x 1 root root459 2007-04-23 14:36 P0S3-08-6-00.loads
 -rwxr-xr-x 1 root root 130228 2007-04-23 14:36 P003-08-6-00.sbn
 -rwxr-xr-x 1 root root 129824 2007-04-23 14:36 P003-08-6-00.bin
 -rwxr-xr-x 1 root root 486974 2007-04-27 14:51 P0S3-06-2-00.sbn
 -rwxr-xr-x 1 root root 486570 2007-04-27 14:51 P0S3-06-2-00.bin
 -rwxr-xr-x 1 root root 392214 2007-04-27 14:51 P0S30202.bin

 I can't recall if I need all the 08-6 versions

 Cheers Duncan


 Sasa wrote:
 Hi Duncan,
 I have tried more times to make the reset phone but is displays always 
 and
 only  'upgrading' and MAC address and I cann't access the phone
 configuration.
 Thanks.

 --

Salvatore.



 - Original Message - 
 From: Duncan Turnbull [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Tuesday, October 14, 2008 11:41 AM
 Subject: Re: [asterisk-users] Cisco 7906g  SIP


 Hi Salvatore

 You need to look at the logs of the tftp server, not the phone.
 Hopefully you can see the ip address of the phone asking for files

 If there is nothing at all being requested from the tftp server then you
 probably want to reset the phone to defaults again.

 Usually it stalls when you have some mismatches in the config files. But
 it almost always asks for the default files.

 From the files requested you can determine whether its asking for SIP
 or SCCP files, and if SIP which version of firmware for the phone

 Cheers Duncan

 Sasa wrote:
 Hi Dave,
 I don't view nothing in tftp server because the phone is stopped on 
 start
 screen with displayed 'upgrading' and MAC address..I don't understand
 what
 happened after the reset. phone
 Regards.

 --

Salvatore.



 - Original Message - 
 From: David Gibbons [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Monday, October 13, 2008 4:29 PM
 Subject: Re: [asterisk-users] Cisco 7906g  SIP


 Hi Salvatore,

 I'm talking about the tftp logs on the tftp server:

 Something like 'tail -f /var/log/tftp' or 'tail -f /var/log/messages'
 should do the trick.

 Dave

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Sasa
 Sent: Monday, October 13, 2008 9:57 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Cisco 7906g  SIP

 I cann't view phone log files because, after reboot, the phone is
 stopped
 on
 this screen ( 'upgrading' with MAC address) !
 Regards.

 --

   Salvatore.

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Re: [asterisk-users] Cisco 7906g SIP

2008-10-14 Thread Duncan Turnbull
Hi Salvatore

Do you have a TFTP server that serves the phone configuration files? 
This is very separate to the phone, i.e. on a server/pc somewhere, and 
will log all the file requests it receives. You can check this 
irrespective of the phone

Have you checked whether tftp requests are being made, usually they come 
  before the system goes into the upgrading state.

I have had that before and it was caused by having different load files 
from that specified in the OS79XX.TXT file which for my phones usually 
have P003-08-6-00 but for upgrading I start from P0S30202

For SIPDefault.cnf you also need the image version to match
#Image Version
image_version:P0S3-08-6-00 ;

But for conversion I first go to this image
image_version:P0S30202 ;

And I go from that to this

image_version:P0S3-06-2-00 ;

then to the current version


And I have these files on my tftpserver which are the respective firmwares

-rwxr-xr-x 1 root root 753560 2007-04-23 14:36 P0S3-08-6-00.sb2
-rwxr-xr-x 1 root root459 2007-04-23 14:36 P0S3-08-6-00.loads
-rwxr-xr-x 1 root root 130228 2007-04-23 14:36 P003-08-6-00.sbn
-rwxr-xr-x 1 root root 129824 2007-04-23 14:36 P003-08-6-00.bin
-rwxr-xr-x 1 root root 486974 2007-04-27 14:51 P0S3-06-2-00.sbn
-rwxr-xr-x 1 root root 486570 2007-04-27 14:51 P0S3-06-2-00.bin
-rwxr-xr-x 1 root root 392214 2007-04-27 14:51 P0S30202.bin

I can't recall if I need all the 08-6 versions

Cheers Duncan


Sasa wrote:
 Hi Duncan,
 I have tried more times to make the reset phone but is displays always and 
 only  'upgrading' and MAC address and I cann't access the phone 
 configuration.
 Thanks.
 
 --
 
Salvatore.
 
 
 
 - Original Message - 
 From: Duncan Turnbull [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Tuesday, October 14, 2008 11:41 AM
 Subject: Re: [asterisk-users] Cisco 7906g  SIP
 
 
 Hi Salvatore

 You need to look at the logs of the tftp server, not the phone.
 Hopefully you can see the ip address of the phone asking for files

 If there is nothing at all being requested from the tftp server then you
 probably want to reset the phone to defaults again.

 Usually it stalls when you have some mismatches in the config files. But
 it almost always asks for the default files.

 From the files requested you can determine whether its asking for SIP
 or SCCP files, and if SIP which version of firmware for the phone

 Cheers Duncan

 Sasa wrote:
 Hi Dave,
 I don't view nothing in tftp server because the phone is stopped on start
 screen with displayed 'upgrading' and MAC address..I don't understand 
 what
 happened after the reset. phone
 Regards.

 --

Salvatore.



 - Original Message - 
 From: David Gibbons [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Monday, October 13, 2008 4:29 PM
 Subject: Re: [asterisk-users] Cisco 7906g  SIP


 Hi Salvatore,

 I'm talking about the tftp logs on the tftp server:

 Something like 'tail -f /var/log/tftp' or 'tail -f /var/log/messages'
 should do the trick.

 Dave

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Sasa
 Sent: Monday, October 13, 2008 9:57 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Cisco 7906g  SIP

 I cann't view phone log files because, after reboot, the phone is 
 stopped
 on
 this screen ( 'upgrading' with MAC address) !
 Regards.

 --

   Salvatore.



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Re: [asterisk-users] Cisco 7906g SIP

2008-10-14 Thread Duncan Turnbull
Hi Salvatore

You need to look at the logs of the tftp server, not the phone. 
Hopefully you can see the ip address of the phone asking for files

If there is nothing at all being requested from the tftp server then you 
probably want to reset the phone to defaults again.

Usually it stalls when you have some mismatches in the config files. But 
it almost always asks for the default files.

 From the files requested you can determine whether its asking for SIP 
or SCCP files, and if SIP which version of firmware for the phone

Cheers Duncan

Sasa wrote:
 Hi Dave,
 I don't view nothing in tftp server because the phone is stopped on start 
 screen with displayed 'upgrading' and MAC address..I don't understand what 
 happened after the reset. phone
 Regards.
 
 --
 
Salvatore.
 
 
 
 - Original Message - 
 From: David Gibbons [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Monday, October 13, 2008 4:29 PM
 Subject: Re: [asterisk-users] Cisco 7906g  SIP
 
 
 Hi Salvatore,

 I'm talking about the tftp logs on the tftp server:

 Something like 'tail -f /var/log/tftp' or 'tail -f /var/log/messages' 
 should do the trick.

 Dave

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of Sasa
 Sent: Monday, October 13, 2008 9:57 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Cisco 7906g  SIP

 I cann't view phone log files because, after reboot, the phone is stopped 
 on
 this screen ( 'upgrading' with MAC address) !
 Regards.

 --

   Salvatore.




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Re: [asterisk-users] Cisco 7906g SIP

2008-10-07 Thread Duncan Turnbull
Are you sure you have set the 7960 to SIP?

By default they use SCCP, so you need to go through the process of 
changing them over, which ideally would just be done with the edits you 
have already in the load files but generally means going back to an 
early version of the SIP code then working upwards from there.

You can check the current hardware in the status, if its SIP it will be 
something like POS-0806... (I haven't got a phone handy to check) but 
there is a reasonable amount of info on voipinfo about the process

Cheers Duncan

Sasa wrote:
 Hi, I have a problem with Cisco 7906G and SIP protocol use with Asterisk 
 1.2.26.
 I have uploaded in my tftp server the firmware 
 'cmterm-7911_7906-sip.8-0-4SR1' that use 'SIP11.8-0-4SR1S.loads' and in 
 SEPmacaddress.cnf.xml I have:
 
 loadInformationSIP11.8-0-4SR1S/loadInformation
 
 ..but in tftp log server I have:
 
 Oct 07 11:56:22 asterisk1.local atftpd[6230.-1208161360]: Serving 
 CTLSEPmacaddress.tlv to 192.168.0.155:49152
 Oct 07 11:56:22 asterisk1.local atftpd[6230.-1208161360]: Serving 
 SEPmacaddress.cnf.xml to 192.168.0.155:49153
 
 ..and in asterisk CLI I have:
 
 -- Starting Skinny session from 192.168.0.155
 Device SEPmacaddress is attempting to register
 
 Now when 7906G started is loaded:
 
 load file: sccp11.8-3-2s
 boot load id: tnp06.3-0-1-31.bin
 
 ..why isn't loaded sip firmware ??
 Thanks in advance.
 
 --
 
Salvatore. 
 
 
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Re: [asterisk-users] Verbosity best practice

2008-09-18 Thread Duncan Turnbull
Its a good question

I have lots of disk space so leave it high, I would rather have the 
detail if I need it

It probably would seem sensible to revisit stable systems after a year 
and lower the verbosity, but then since I can afford the space I am not 
too fussed.

Cheers Duncan

Olivier wrote:

 Hello,

 When managing a stable system, which verbosity level do you adopt ?
 Leaving a higher level helps to catch root cause, if for any reason, 
 things go wrong.
 Leaving a lower level saves resources if you need (have) to backup logs.

 What are current best practices ?
 Do you change verbosity level during system lifecycle ?

 Regards



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Re: [asterisk-users] Wi-SIP 802.11f - Inter Access Point Protocol HANDOFF

2008-08-31 Thread Duncan Turnbull
Its not so hard if the APs are purely just converting ethernet to 
wireless. If there is any authing on the AP then it would be tougher. 
And a centralised DHCP issuer is important i.e. just one address range 
across all APs so when moving APs there is no dhcp change, no auth 
change, just a client reconnecting to the SSID. I guess this is not 
exactly what you are talking about but organising such protocols to work 
properly does become much more complex, and no we didn't have so much 
joy that way.

We have a wireless ISP in Wellington, New Zealand called CafeNET and 
thats all the APs do i.e. wireless to ethernet for large zones of the 
city homed back to a central controller. I have walked along our main 
central city street, Lambton Quay and carried on a conversation moving 
between at least 3 APs. I usually walked about 200 - 350m  depending my 
destination so could be using quite a few APs

The asterisk box in question is not blocked by the ISP.

Driving is different, but walking is okay, and the street noise masks 
the other occasional glitches, so I may think its doing better than it 
is, it can be noisy and hard to hear in any event with a mobile as well.

I don't do it so much any more because the cellphone charges got lower 
and I got tired of two devices, especially one that ran out of 
batteries. But I did for a few months to prove the point you are asking 
about.

Cheers Duncan

Michael Graves wrote:

On Sat, 30 Aug 2008 11:51:49 -0500, Karl Fife wrote:

  

Has anyone ever really, truly, actually held on to a Wi-SIP call while
moving from the range of one AP to the range of another AP in the same
network?  

Let's say a 'YES' only counts if you had a bona-fide handoff.  In other
words, you began in place 'A' (within range of AP#1 but OUTSIDE the
range of AP#2), AND THEN MOVED to place 'B' (in range of AP#2, but
completely outside the range of AP#1) WITYOUT dropping the call.  

Supposedly it's possible with compliant hardware using 802.11f -
Inter-Access Point Protocol (IAPP), but given how ALL standards ALWAYS
work together PERFECTLY, 100% of the time :-), I'm guessing that it
doesn't work.  Can anyone speak to this from experience?

-Karl



Karl,

I'm guessing that it was not common. 802.11f handoffs reportedly take
100ms which is considered too long for streaming applications like
voice and video.

The 802.11r standard was only agreed upon and released days ago. This
specifies FAST BSS transition specifically to saisfy such applications.
Not sure if any hardware supports this as yet.

http://en.wikipedia.org/wiki/IEEE_802.11r

Michael
--
Michael Graves
mgravesatmstvp.com
http://blog.mgraves.org
o713-861-4005
c713-201-1262
sip:[EMAIL PROTECTED]
skype mjgraves
fwd 54245




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[asterisk-users] AA50 using multiple outbound routes

2008-08-04 Thread Duncan Turnbull
Hi All

I have an AA50 without inbound DDIs but each line has a separate number 
so based on analogue port it can be routed to different people. The 
challenge with this method is it appears to only allow the dial plan to 
use 1 outbound route so if all the analogue ports are split into 
individual lines it can only use 1 line for outbound, it won't allow it 
to step to other ports.

This seems a less than ideal design and it could be I am missing 
something. If anyone knows how to make the outbound calls on an Digium 
AA50 appliance step through all its available ports (or at least a 
selected subset of them) I would love to know.

Thanks very much

Cheers Duncan

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Re: [asterisk-users] TDMoE with Telco

2008-08-03 Thread Duncan Turnbull
You can use TDMoE to get an E1 running but its really designed to 
replicate an E1 end to end

Its a standard and there is equipment out there that does it, e.g. from 
RAD and a few others. I didn't have any joy using the Asterisk code to 
get it going but it should in theory work. Its completely different to Dundi

The challenge it is a protocol and needs two boxes talking TDMoE at each 
end. Telco's do not have this as an option, or at least none do that I 
have found

Cheers Duncan

Michael Graves wrote:

 --Original Message Text---
 *From:* Yacine Boukaba
 *Date:* Sun, 3 Aug 2008 18:54:08 +0100

 Hello, is it possible with TDMoE to replace classic digital T1/E1 
 interfaces like digium and sangoma cards connected to a telco. Or 
 TDMoE is only possible for connecting two asterisk boxes using their 
 NIC interfaces. if TDMoE can work with an T1/E1 connected with telco 
 how we can get the remote mac address of the telco interface ? 
 ThanksNo virus found in this incoming message.
 Checked by AVG - http://www.avg.com
 Version: 8.0.138 / Virus Database: 270.5.10/1586 - Release Date: 
 8/1/2008 6:59 PM

 I thought that TDMoE was largely depricated in the wake of DUNDi?

 Michael
 --
 Michael Graves
 mgravesatmstvp.com
 http://blog.mgraves.org
 o713-861-4005
 c713-201-1262
 sip:[EMAIL PROTECTED]
 skype mjgraves
 [EMAIL PROTECTED]




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Re: [asterisk-users] gui issue in asterisk aa50

2008-07-15 Thread Duncan Turnbull
I had an issue where I put a comma in the prepend digits string pn  
call plans and then the call plan menu would no longer load.
It parses the menu from the text file so I used the file editor to  
clear the offending line and my menu came back. Not sure if thats your  
issue but I was surprised I could enter text that broke the menus

Cheers Duncan



On 16/07/2008, at 10:27 AM, Sydney Web Hosting [EMAIL PROTECTED] 
  wrote:

 HI all,

 I am having issues with the gui on my AA50.

 under Voice Menus  Add new Step  Go to Time based rule.

 It allows me to select “Go to Time based rule” from the menu but  
 no options come up when selected.

 I’ve tried all browsers but no luck.



 Thanks
 David.






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Re: [asterisk-users] astrundir not used

2008-07-08 Thread Duncan Turnbull
Are you using ubuntu?

Usually I have to edit the Makefile in the else section of Global 
variable declaration based on architecture
# ASTVARRUNDIR=$(localstatedir)/run
ASTVARRUNDIR=$(localstatedir)/run/asterisk

This seems to do it

Cheers Duncan

on 07/09/08 04:53 Cyril SCETBON said the following:
 hi,

 I'im using asterisk 4.1.21 and astrundir is configured as followed in 
 /etc/asterisk/asterisk.conf :

 [global]
 astetcdir = /etc/asterisk
 astmoddir = /usr/lib/asterisk/modules
 astvarlibdir = /var/lib/asterisk
 astagidir = /usr/share/asterisk/agi-bin
 astspooldir = /var/spool/asterisk
 astrundir = /var/run/asterisk
 astlogdir = /var/log/asterisk

 when I start asterisk it creates his pid file and the ctl socket in 
 /var/run and not in /var/run/asterisk

 How can I fix it ? Is it a known issue ? I did not get this error with 
 asterisk 1.4.10

 Thanks
   

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Re: [asterisk-users] sip extension compromised, need help blocking brute force attempts

2008-06-30 Thread Duncan Turnbull
Try some of the shell scripts in the asteriskcookbook recipe heap

http://asteriskcookbook.com/wiki/index.php/RecipeHeap

Specifically
http://asteriskcookbook.com/wiki/index.php/Asterisk_Brute_Force_Prevention

Cheers Duncan

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton
Sent: Tuesday, 1 July 2008 07:33
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] sip extension compromised,need help blocking 
brute force attempts

iptables -A INPUT -p tcp -s 74.52.112.162 -j DROP
Good luck.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of spectro
Sent: June 30, 2008 12:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] sip extension compromised, need help blocking
brute force attempts

Hello, yesterday one of the extensions on my asterisk server got
compromised by brute-force attack. The attacker used it to try pull an
identity theft scam playing a recording from a bank your account has
been blocked due to unusual activity, please call this number...

Attacker managed to make lots of calls for around 8 hours before I
detected it and changed the password for that extension. As of this
morning it is still attempting to brute force the password for that
extension again. I need a way to block that IP from connecting to my
asterisk server, please advice.

--- sip debug ---
Using INVITE request as basis request -
[EMAIL PROTECTED]
Sending to 74.52.112.162 : 5060 (NAT)
Found user '211'
Reliably Transmitting (NAT) to 74.52.112.162:5060:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP
74.52.112.162:5060;branch=z9hG4bK3b28fa36;received=74.52.112.162;rport=5060
From: ASLPLS sip:[EMAIL PROTECTED];tag=as130a4d39
To: sip:[EMAIL PROTECTED];tag=as0c69057b
Call-ID: [EMAIL PROTECTED]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
llow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0
--- sip debug ---

That box is currently running Trixbox 1.2.3. I have iptables disabled.
If anybody can give me a simple ruleset that allows all traffic except
ip 74.52.112.162 to port 5060 I will really appreciate it.

Are there mechanisms in Asterisk to detect and automatically block
these brute force attempts?

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Re: [asterisk-users] one way audio after call transfer

2008-04-30 Thread Duncan Turnbull
I had a similar issue in 1.2 after transfer and we were using SIP only 
but an upgrade cured it

We are now on 1.4.18 still without issues

Cheers Duncan


Rilawich Ango wrote:

Hi all,

  Recently, I experienced one way audio after call transfer.

incalling call (PSTN)  A -- GXP2000 thro' zap --blind transfer-- destination 
B
Finally A and B reach each others, but there is only one way audio.
Anyone get the same experience before?  How to solve the problem?

Asterisk vesion:
Asterisk 1.4.15
zaptel 1.4.7
asteriks-addon 1.4.5

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Re: [asterisk-users] Cisco to Asterisk migration

2008-04-25 Thread Duncan Turnbull
Hi Femi

We have about 50 Cisco 7960s on one site off Asterisk 1.4.18

Its all SIP and it doesn't stress a P3 system much at all.

I am not sure what phones you are using - the 7960s are not hard to configure, 
a bit of process to convert from the Cisco Skinny to
SIP (using SIP v8.6) but everything seems to work well. The 7961s or 7971s use 
an XML config which is probably 

Everything loads off the TFTP server. We are using the Linksys POE Switches 
SFE2000P which seem okay but don't always like to be
fully loaded 

Things I would work on are automating or simplifying the provisioning (doesn't 
change that much once its done), firmware upgrades,
and getting to know the config files well. 

Cheers Duncan

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Femi
Sent: Friday, 25 April 2008 21:34
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Cisco to Asterisk migration

Hi Guys,
I have client with a Cisco 2690 call manager solution that wants to upgrade
but cannot stomach the costs of continuing with Cisco

The installation will go up to 100 users
The client currently has about 40 Cisco phones and would like to continue
with these phones with the odd Polycom

I'm looking at plugging in an Asterisk box and using the existing Cisco box
as a PSTN gateway only

Has anyone on the list done this?
Any pitfalls or tips you would like to share?


Thanks

Femi


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[asterisk-users] Best practice security for internet access to Asterisk

2008-01-29 Thread Duncan Turnbull
Hi All

For the scenario of a single asterisk server that needs to serve clients 
on the net, as well as local office clients, I would be very interested 
in people's views of the best method to handle security to prevent net 
based attacks while still allowing the client access.

Some of the challenges I see are:
- preventing brute force and bot type attacks
- monitoring for unusual events and notifying and acting appropriately
- limiting damage if someone does get in
- avoiding a Denial or degradation of service on your asterisk platform
- making it easy for staff to use

Some of this can be done with
- firewall control - but its hard to limit where your clients will come 
from, besides restricting ports
- scripts monitoring logs, I saw a recipe for checking password failures 
then blocking that ip after x failures, I imagine this could get quite 
sophisticated
- using separate restrictions for offnet users but this kind of makes it 
harder for the staff members.
- using a proxy in front of asterisk for SIP, to limit the available 
extensions and minimise the scanning impact on the asterisk box. I am 
hoping this could detect and prevent illegitimate or poorly formed 
requests or unknown user agents. Staff should be using a standard set.
- using iax softclients to shift the attack requirements - I don't know 
much about how well these work
- running all clients over a vpn e.g open vpn, but this is not so good 
for wireless handsets or other devices that can't do a vpn

I am interested in all views and recommendations

Thanks very much

Cheers Duncan

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Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!

2007-12-16 Thread Duncan Turnbull
We build and maintain 7 Asterisk boxes for our customers, I have 
recently moved 3 to 1.4. I also use iaxmodem and on the last one 1.4.14 
I was getting iax thread errors - which was reported as a bug in much 
earlier versions but apparently fixed.  When 1.4.15 came out (two days 
later) it solved this problem, for me at least. I didn't dig any further 
but it did moderate my confidence somewhat.

We run everything on ubuntu server 6.06 LTS and also use freepbx as the 
interface with some minor customisations. It works very well and we are 
now shifting some others to 1.4 but the issue is if anything goes wrong 
its too costly to fix, as part of maintenance we keep them uptodate. The 
main blocker for 1.4 was freepbx but now it supports 1.4 and seems to 
manage the transition really well.

However being a small self employed group of two the main reason to 
stick with what works is the risk of cost. We don't generally do major 
upgrades without charging but there isn't any seriously missing 
functionality yet, and the effort involved to be sure it will be hassle 
free is significant. The clients have to see value in the upgrade.

We also work with people still on version 1.0, because the risk of 
change to a working system is too high

This seems to be the same issue already mentioned but my take on it is 
most people can't cope with any risk on production machines unless there 
is some significant gain. Its been a year now, generally I would think 
that means its probably starting to become stable but a year isn't very 
long really. Give it another year and the new installs will mostly be 
1.4 and the migration process will be a lot more trusted. I don't think 
a year is really long enough to expect much more than where you are at. 
The debian stable, unstable, and testing model would be useful here, 
debian stable is so reliable it just rocks, if there was a version like 
that it would be fantastic (of course you trade access to the latest 
features for it) . We find ubuntu server a great balance between debian 
stability and getting the latest options.

Is there a performance analysis of 1.2 vs 1.4 around or a clear business 
analysis of the distinctions in value for each?

Cheers Duncan

Lyle Giese wrote:

 Olle E Johansson wrote:

All I can say is with 1.6, if a change is made that causes something  
that worked in 1.4 not to work in 1.6, please think twice, three  
times or four times before making the change, or making the change  
in such a way that it won't break dialplan stuff from 1.4.



Our policy is to never remove any functionality between two versions.  
We replace the functionality with new functionality and print out  
warnings whenever you use the deprecated functions. We also add this  
to the documenation in the software and the UPGRADE.TXT file. So the  
functionality that you lost in 1.4 was old 1.0 functions that was  
marked as deprecated in 1.2 and removed in 1.4.

We might want to be more informative about those changes. We need to  
make a clear list of things you need to start changing as a user of  
1.4 to prepare for lost functionality in 1.6. This information already  
exist, but should maybe be a bit more public.

In some cases we do have to change in a dramatic way and can't  
preserve the old functionality to solve a bug in the software. This  
requires thorough discussion in the developer group and is something  
we really want to avoid at all costs. If this happens, it's clearly  
documented in the software.

Thank you for your feedback, it's important to us.

/O

  

 Along that this same line, I ran 1.0.something for a long time and it 
 was working just fine for my SOHO.  I had a channel bank to interface 
 pots lines from the local Telco and feed the analog phones in the 
 house.  Over time, I replaced most of those analog phones with SIP phones.

 An unfortunate incident caused us to lose that server and several sip 
 phones.  When I recovered enough to rebuild *, I tried 1.4 and it 
 would not compile completely and zaptel did not load properly.  I 
 download 1.2 and it worked with the same configs as 1.0, but the 
 quality was poor.  That was due to hardware issues.

 I purchased a new motherboard and rebuilt using a newer Asterisk 1.4 
 with the then current libpri and zaptel and the call quality came 
 back.  But I had a hard time with syntax changes.  Basically I was 
 jumping from 1.0.x to 1.4.x in one leap.

 My biggest gripe is that everything loaded and seemed to work.  A day 
 later we found this did not work and discovered a syntax change.  A 
 day later something else did not work, an other syntax change.  Why 
 isn't there some pre-processor to check the syntax of the config 
 files?  Would have saved me a whole bunch of time I didn't have to 
 spare and still don't.

 Lyle
 As it is syntax problems or changes are not noticed or logged until 
 Asterisk tries to execute them. If there is a chunk of code that is 
 only hit once a week???  It almost 

Re: [asterisk-users] 7960 Queue Issue

2007-11-04 Thread Duncan Turnbull
The freepbx system has a primary number option in its ring group dialing 
which if selected as a ring strategy means it won't ring any further if 
the primary number is engaged. This is useful in follow me setups.

I haven't dug into how its implemented but it works for ring groups and 
follow me on freepbx (asterisk 1.2 and 1.4)

An article on the concepts.
http://freepbx.org/2007/06/03/ring-group-and-follow-me-ring-strategies-1-of-2

It maybe useful to help figure out a way around your issue.

Cheers Duncan

on 11/05/07 14:09 Nick Brown said the following:
 Thanks Eric, this is the case. A bit of a shame that it removes the
 functionality for the member to see calls that have not come from a queue
 however there is not much choice in the matter.

 FWIW to get this option a firmware upgrade was required (Now running
 POS3-08-8-00).

 Cheers.


 On 5/11/07 11:57 AM, Eric Merkel [EMAIL PROTECTED] wrote:

   
 On 11/4/07, Nick Brown [EMAIL PROTECTED] wrote:
 
 Morning All,

 Quick question that has me stumped. Have a queue with several members
 (Statically defined in queues.conf at this stage for testing) who use Cisco
 7960's.

 The queue is configured to use rrmemory and generally this works correctly.
 However if a member is already on a call their phone will still ring (The
 7960 can show multiple incoming calls for one line). I really don't want
 members who are on calls to get more calls. Especially when we start logging
 out members who don't answer.

 Asterisk shows;
 -- Called 1014
 -- SIP/1014-08f2e4d0 is ringing
 -- Local/[EMAIL PROTECTED];1 is ringing
 -- Nobody picked up in 15000 ms

 Short of disabling the feature to show multiple incoming calls on the 7960's
 (Which I don't know if it can be done anyway), has anyone got any
 suggestions?

   
 Yes, you can turn off this in the phone. Go into call preferences on
 the phone and turn off call waiting. Not optimal but can be done.

 -Eric

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Re: [asterisk-users] Dell PowerEdge 860, Sangoma A108

2007-10-08 Thread Duncan Turnbull
Hi Helen

Sounds good, I think Troy will need me to setup the notification list to 
the winners though so it might pay to send me those details directly

Should be better rugby this weekend for one of us ;-)

Cheers Duncan

on 10/09/07 14:20 Paul Hales said the following:
 We have used a quite a few dell 860's in our installs with Digium cards
 (Te120's) without any issues.

 PaulH


 On Mon, 2007-10-08 at 15:28 -0700, Girts Graudins wrote:
   
 Hello everyone,

 I'm considering getting me a quad-core Dell PowerEdge 860 to run
 Asterisk.  Anyone else using this model?  Any tales of woe and sorrow
 I should know about?

 Then, in a couple of weeks, I'm thinking of getting a Sangoma A108 and
 giving that a try.  Same question with that one - any quirks I should
 be aware of? 


 Girts
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Re: [asterisk-users] FreePBX (2.3) - Good? Bad? Ugly?

2007-09-13 Thread Duncan Turnbull
I am yet to use 2.3 but have 2.2 on 8 ubuntu based installations with Asterisk 
1.2.18 or greater

FreePbx is really useful as an interface to all the config files, stats etc, 
its also really great if your customers need some
control

The documentation has recently been updated and there is a lot of life in the 
project so I would recommend it

Just note, that like everything you still need to put some time into 
understanding what you are doing and how to get around the
systems

Cheers Duncan

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay R. Ashworth
Sent: Friday, 14 September 2007 8:57 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] FreePBX (2.3) - Good? Bad? Ugly?

On Thu, Sep 13, 2007 at 04:32:27PM -0400, Jay R. Ashworth wrote:
 I'm about to (finally) do my first Asterisk install; SMB, 4 FXO, 4-6
 stations, mostly IP (I'm looking at the Grandstream 201, to start), and
 maybe X-lite on a couple of laptops via VPN.
 
 We've got a 4xFXO box we bought off eBay, which unfortunately I can't
 find to quote a model number off of, but I *think* it's a Grandstream
 as well.
 
 I've looked at several of the packages that turn Asterisk from a PBX
 construction kit into an *actual* PBX, and so far FreePBX looks like
 the one that matches my mental model of a small phone system best.
 
 Anyone have any first hand experiences with it that they'd like to
 share?

And I inadvertantly thread-jacked someone.  Sorry.  Fixed.

Cheers
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth  Associates http://baylink.pitas.com '87 e24
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Re: [asterisk-users] Slow list

2007-07-17 Thread Duncan Turnbull
This message arrived today 18 July NZ time

Full headers below but most of my mail is like this - the offending bit seems 
to be: INXS.digium.internal which took 4 days to
deliver it 

Cheers Duncan

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-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel van Baak
Sent: Sunday, 15 July 2007 8:40 a.m.
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Slow list

On 16:28, Thu 05 Jul 07, Philipp Kempgen wrote:
 Since the list was switched over to API-Digital almost
 every message I get is older than a week. Coincidence?
 Is anyone else having trouble?
 
 Regards,
   Philipp

I got this message today July 14
Yes, I have the same.
-- 

Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD

Why is it drug addicts and computer afficionados are both called users?


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Re: [asterisk-users] Slow list

2007-07-17 Thread Duncan Turnbull
I thought initially it was a pretty poor generalization about postgrey and our 
capabilities until I realized that this was sent a
few weeks ago when this probably wasn't an as obvious issue. But it clearly is 
an issue now.

I have checked my mail servers for failures, implicitly greylisting is working 
as the mails are coming from digium constantly - just
a long time delayed, if postgrey was an issue there would still be retries and 
there have been none in over a week - as long as my
logs go back, any decent mail server should have retried in much less than a 
week.

Anyway - a discussion and investigation of issues is made pretty hard with 4 
-10 day gaps in it. Since every other list works on
time (+- a few hours) its looking like Digium from my view.

I imagine someone will have sorted it before I see my own post, fingers crossed.

Cheers Duncan 

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Walt Reed
Sent: Friday, 6 July 2007 6:33 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Slow list


On Thu, Jul 05, 2007 at 01:40:50PM -0400, Doug Lytle said:
 Well, this is now the third active thread on this subject, but I guess
 you won't see this message for a while.  Has anyone dissected the
 headers of a delayed message yet?  We should be able to tell for sure
 where the holdup is.  All of the messages are coming through on time
 for me, so it won't do much good for me to look.
 
 
 Looks like mail is getting held up between INXS.digium.internal and 
 lists.digium.com
 
 INXS.digium.internal received it the first of July, lists.digium.com 
 received it on the 4th.
 
 drdos.info (ME) received it from lists.digium.com on that same day (Today).

What you can't see without looking at the mail server logs on both ends
is delivery attempts. Greylisting for example can totally hose you over
depending on the implementation. Greylisting without whitelisting is
irresponsible.  How many tries did the digium server make before the
message finally got through??? That's what we need to know. Only Digium
can say.

Before poking Digium too much, I would look at exactly what YOUR mail
servers are doing that may potentially be the real cause of the delays.

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RE: [asterisk-users] SugarCRM Integration

2007-06-02 Thread Duncan Turnbull
If you look through the Trixbox without Tears by Ben Sharif - google for it, 
it's a good read for things you can do for asterisk

 

Ch 31 has this below

 

I would search the tribox and sugar forums for more info - really its just 
using click to dial from sugar, and potentially CID
lookup - I am not sure if it is using 

 

Cheers Duncan

 

SugarCRM is a contact management software that comes bundled with Trixbox.

To set up SugarCRM, first, you need to open the SugarCRM application

http://your.trixboxip.address/crm using the default username of Admin and the

password of password.

For security reason you should change the Admin password. To do this, click on 
'My

Account' in the upper right-hand corner, then click on the 'Change Password' 
button

underneath 'Users: Administrator (Admin) in the center-left of the screen.

Change it to a new password and confirm your new password and click 'Save.'

Now it's time to set up your contacts. I will start off setting up a couple of 
my internal

extensions.

Click on 'My Account' again and then click the 'Edit' button.

Change 'Asterisk Phone Extension' to your Asterisk extension. My extension is 
2001.

While you are at it, change your time zone and date format as well.

Click 'Save' to save that information.

Let's add another one.

Click on the contacts tab and then select 'Create Contact' from the left hand 
Shortcuts

menu.

Add another extension, in my case I chose my daughter's extension 2002:

Firstname: Norsurya

Last name: Sharif

Home: 2002

Click 'Save' to save that information.

Add another and another if you want to, using the method above.

At this point, you may find that you are unable to make a phone call through 
SugarCRM.

This is due to a little bug in the popup_picker.php (this bug may have been 
fixed by the

time you read this, but at the time of writing, this bug exists).

To fix this bug, you need to edit popup_picker.php by doing the following:

From your Linux CLI, log in as root.

cd /var/www/html/crm/modules/Contacts

nano Popup_picker.php

Browse down to line 121 and change it from:

$number = preg_replace ( /[^\d\*]/, , $number );

To

$number = preg_replace ( /[^\d\*]/, , $display_number );

TRIXBOX Without Tears Page 136 of 209

You should now be able to dial from SugarCRM to your other internal extensions 
and to

the outside world.

Note: You can add multiple users who will each have their own 
settings/contacts/etc.

 

 

  _  

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Crazy Boy
Sent: Saturday, 2 June 2007 7:25 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SugarCRM Integration

 

Yes. You are right. You can integrate Sugar with Trixbox very easily. You can 
customize it also.

Thanks,
Chandra

Joseph Bajin [EMAIL PROTECTED] wrote:

I'd like to know as well about this.

On 6/1/07, Diego Quintana Cruz wrote:
 Hi folks,
 I was wondering if there's a guide on how to configure sugarCRM
 Integration with Asterisk. I was looking in google and all i found was
 about Trixbox, which has sugarcrm integrated by default.

 Regards,
 --
 Diego Quintana a.k.a. RouterMaN
 Ingeniero de las Telecomunicaciones
 Linux Registered User #382615 - http://counter.li.org/
 SIP # 1-747-633-6676 Ext. 1011
 FWD # 764839 Ext. 1011
 http://routerman.blogsome.com
 http://gst.telecom.pucp.edu.pe
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  _  

Park yourself in front of a world of choices in alternative vehicles.
Visit
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Z3JlZW4tY2VudGVy  the Yahoo! Auto Green Center.

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RE: [asterisk-users] Delays on E1 Delivered via SHDSL

2007-05-30 Thread Duncan Turnbull
I doubt it's the PRI itself

SHDSL isn't part of the internet per se, its just an access technology.

SHDSL is just synchronous DSL which can be used to deliver E1s over.

ISDN PRI's are delivered in a 2Mbit/sec G703/G704 frame and will give you lots 
of alarms if they are having any issues

It could be your toll provider at the end of it is routing calls in ways that 
cause delays, but less likely to be the PRI

Cheers duncan

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Sent: Thursday, 31 May 2007 12:18 p.m.
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Delays on E1 Delivered via SHDSL

I have an Asterisk system with a TE110P installed and connected to an ISDN
E1 PRI that is delivered via a 2mb SHDSL connection. I am experiencing
delays (the type of delay you would get on an international call) during
calls. I am wondering if anyone could advise, would the problem be with any
part of the Asterisk system or is the problem with the fact that the ISDN is
delivered over the internet?



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RE: [asterisk-users] Bottom line on fax reception

2007-05-28 Thread Duncan Turnbull
I use Asterisk and Hylafax and iaxmodem to send 600+ faxes per week with no 
baby sitting, I receive about 20 and it requires no baby
sitting

Hylafax and iaxmodem with Asterisk works very well - check out iaxmodem and 
hylafax lists for much bigger examples

Cheers Duncan

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of shadowym
Sent: Tuesday, 29 May 2007 7:34 a.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Bottom line on fax reception

Sorry but I must have missed it if someone else responded.  If the built in
fax reception doesn't work very well what about the 3rd party stuff
mentioned on the Asterisk Wiki?

-Original Message-
From: Steve Totaro [mailto:[EMAIL PROTECTED] 
Sent: Monday, May 28, 2007 8:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Bottom line on fax reception

Someone already answered this question.  The answer is no, it does not work
by your definition of production ready.

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
 

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users- 
 [EMAIL PROTECTED] On Behalf Of shadowym
 Sent: Monday, May 28, 2007 11:20 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [asterisk-users] Bottom line on fax reception
 
 Anybody??
 
 -Original Message-
 From: shadowym [mailto:[EMAIL PROTECTED]
 Sent: Thursday, May 24, 2007 9:35 AM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Bottom line on fax reception
 
 
 
 So what is the bottom line?  Does it work or not.  I've heard stories
it
 works, it doesn't work, it kinda sorta works when it's not raining out 
 side.
 Everything under the rainbow.
 
 What's the bottom line with recent updates on 1.2.x?  Is it production 
 ready for fax?  By production ready I mean that it just works all the 
 time
and
 doesn't need any babysitting.  Do I have to worry about dropped lines, 
 sometimes not detecting incoming fax toneyada yada.
 
 I know I don't have to use fax on Asterisk but I really want to for 
 various reasons.  Mostly incoming but outgoing is a nice to have.  
 Should I
use an
 addon package and if so which one?  Any help would be appreciated.
 
 
 
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RE: [asterisk-users] WiFi SIP phones

2007-05-23 Thread Duncan Turnbull
I have a recent dual gsm /wifi from e28 via Skyvoice. (http://myskyvoice.com/) 
Its built to use voip or gsm and is about the same
price as existing wifi phones. My main hassle is it doesn' yet do WPA - WEP's 
okay and they say WPA is only a firmware load away ;-)
, and it has a browser to login if you need to.

 

So far so good and then to some degree I am not sure I would use a wifi only 
phone again

 

That said wifi voip is still occasionally flaky but I much prefer it to soft 
clients on the laptop.

 

Cheers Duncan

 

  _  

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves
Sent: Thursday, 24 May 2007 2:50 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] WiFi SIP phones

 

I travel a lot for work. I frequently find hotels that have wifi, free or 
otherwise available. But I've yet to find it anywhere near
sufficient to support voip applications. At least not good enough to compel me 
to not use my cell phone. If you have control of the
host LAN then you can ensure it meets the needs of a wifi SIP phone, otherwise 
why bother.

Has anyone ever seen anyone making a voip call on a wif handset ata public 
hotspot? While that would score many geek points I doubt
it would work in many places.

About 18 mo ago I bought the Hitachi Cable WIP5000 handset. It was seriously 
flawed so I resold it after a few months and settled on
the Aastra desk phone. I do wish the cordless handsets were a little more like 
a Panasonic cordless phone...more buttons...easier to
program, etc.

Michael

On Wed, 23 May 2007 21:59:03 -0400, Justin Moore wrote:

On 5/23/07, Michael Graves [EMAIL PROTECTED] wrote:
 I must say that I've VERY happy with my Aastra 4801 CT phones. I think that
 they're DECT. Each can have up to six cordless handsets. Technically its a 9
 line phone, but if you use G.729 you can only sustain two calls at once. I
 can have a call on the portable and easily take another on the base.

I am also an extremely happy user of an Aastra 480i CT. Awesome phone.
However, I was under the impression that the OP was looking for a WiFi
phone that could be carried from place to place, but I may be wrong...

-- 
Justin Moore
aka wantmoore
---
www.wantmoore.com
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RE: [asterisk-users] HUDlite Server on Debian Etch/Asterisk 1.4

2007-05-20 Thread Duncan Turnbull
I have the same challenge and issue, the server dies shortly after being fired 
up, although I am using Asterisk 1.2

Even with strace its very trying to work out whether the messages are errors or 
importance or just run of the mill

All advice and options appreciated

Cheers Duncan

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
Sent: Thursday, 17 May 2007 11:37 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] HUDlite Server on Debian Etch/Asterisk 1.4

On Wed, May 16, 2007 at 03:22:35PM +0200, Jack wrote:
 Hi,
 
 has anyone managed to get hudlite server working on a Debian Etch
 based installation of Asterisk 1.4?
 
 So far I managed to eliminate all error messages, but the process is
 killed directly after starting the hudlite server without showing any
 error messages.
 
 I would be very happy if anyone can give me some hints or point me to
 a installation guide.

What I would do in such a situation, is run everything under strace.

However, recall that you're dealing with a proprietary program here. The 
only ones who have the full information to help you are Fonality.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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RE: [asterisk-users] DTMF not working using *98, but OK on inbound routes?

2007-05-20 Thread Duncan Turnbull
I have this happening with a Cisco 7960 - I can't see what the difference is, I 
have asterisk 1.2.13 and a number of 7960s which
happily work, as well as some 7961s which also work. 

However one 7960 doesn't, although it dials quite happily but that's probably 
due to dtmf being put into SIP rather than inband. Why
one works and the other doesn't I don't yet know. 

Cheers Duncan

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug
Sent: Thursday, 17 May 2007 2:40 p.m.
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] DTMF not working using *98,but OK on inbound routes?

Has anyone seen anything like this:

I dial *98.  Asterisk says Password?   I punch in
the password, and the system doesn't recognize the
tones.

However, if I dial my own number and ignore the
incoming call, it goes to voicemail, and then
I can get into voicemail.

I have a sneaking suspicion that Asterisk is
somehow not recognizing the DTMF tones somewhere
along the way.

This happens intermittently with Linksys ATAs and
Polycom phones.  Using a Cisco 3640 VOIP router.

Any ideas on what to check?


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