Re: [asterisk-users] asterisk server stress test

2015-08-20 Thread Steven Howes
On 20 Aug 2015, at 11:12, Sevana Oy sa...@sevana.fi wrote:
 Curious why didn’t you try AQuA http://sevana.biz/products/aqua/ to score 
 the quality? Using voice files for tests has more representation to my 
 opinion.


Spot the salesman? ;)

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Re: [asterisk-users] packages.digium.com

2015-03-12 Thread Steven Howes
On 11 Mar 2015, at 17:53, Matthew Jordan mjor...@digium.com wrote:
 On Wed, Mar 11, 2015 at 10:28 AM, Steven Howes
 steve-li...@geekinter.net wrote:
 Anyone know where it’s gone?.. Appears to have been down all day.
 The hamsters should be running in their wheels again now.

Cheers Matthew. Give them some food from me.

Steve
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[asterisk-users] packages.digium.com

2015-03-11 Thread Steven Howes
Anyone know where it’s gone?.. Appears to have been down all day.

Steve
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Re: [asterisk-users] (no subject)

2015-02-09 Thread Steven Howes
On 9 Feb 2015, at 15:32, Francisco Leonardo Mota francisco.m...@rnp.br wrote:
 Submission.
 
 Thanks,

Uh, no problem?..

Steve
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Re: [asterisk-users] Subscribe event ua-profile

2014-11-10 Thread Steven Howes
On 10 Nov 2014, at 13:01, Norman Laidla norman.lai...@telegrupp.ee wrote:
 Well, pants. It actually is causing a problem, because the phone doesn't use 
 any other methods to register to Asterisk. This is a bit of a big issue.

What softphone is it? It sounds like rather odd behaviour.
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Re: [asterisk-users] Asterisk 13 stable?

2014-10-28 Thread Steven Howes
 The Asterisk 13 is already stable for production environment?

It’s only been out a couple of days - hard to make judgements just yet. But it 
is out of Beta so should be of reasonable quality - give it a try :)

Steve
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Re: [asterisk-users] Asterisk Phone ( Telecom feature )

2014-10-07 Thread Steven Howes
On 7 Oct 2014, at 09:24, Dania Asi da...@futuretrendsest.com wrote:
 Kindly note that I asked about the capability of the phones and now I am
 asking about the way I can do it to my client's phones, because he is asking
 for a demonstration.

Yet you’ve not even told us the phones in use. You can’t just expect a mailing 
list to do your work for you. You need to look at the handsets and see what 
they support, and how they support it. You don’t even say if the handsets are 
SIP or not, the PSTN connectivity is pretty irrelevant.

 Sales Executive Engineer

That explains that one.

Steve
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Re: [asterisk-users] Custom SIP-header not present in call Asterisk to Asterisk

2014-09-02 Thread Steven Howes
On 2 Sep 2014, at 09:03, Jonas Kellens jonas.kell...@telenet.be wrote:
 So just before hanging up, I add a custom SIP-header :
 
 exten = s,n,SIPAddHeader(X-My-Hangup: MaxChan)
 exten = s,n,Hangup()

SIPAddHeader only works for INVITE as far as I know.

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Re: [asterisk-users] Custom SIP-header not present in call Asterisk to Asterisk

2014-09-02 Thread Steven Howes
On 2 Sep 2014, at 10:38, Jonas Kellens jonas.kell...@telenet.be wrote:
 Then how can I let another Asterisk server know the custom reason of hangup ? 
 If it is not possible with custom SIP-header, then how ?

As far as I know that’s going to require a source change. May not be the case 
with PJSIP though - not used that yet.

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Re: [asterisk-users] AMI Elastix

2014-08-18 Thread Steven Howes
On 18 Aug 2014, at 09:27, Усин Айбек prince...@gmail.com wrote:
 I have trouble with connection to AMI 1.1 wich enabled on Elastix 
 
 Asterisk Call Manager/1.1
 Action: Login Username: admin Secret: qweasd123
 
 Response: Error
 Message: Missing action in request

You are missing the newline characters.. Action/Username/Secret should be on 
their own lines.

Read the AMI spec.

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Re: [asterisk-users] The plain old PBX functionality

2014-08-08 Thread Steven Howes
On 8 Aug 2014, at 06:05, Gergo Csibra csi...@gmail.com wrote:
 back in the old analog telephony days there was digital PBX-es and
 digital system phonesets. This phonesets have had many individual
 illuminatable buttons connected with extensions. The PBX can show on
 the buttons if some extension is ringing (blinks) or busy (constant
 light), and the user can transfer the call with one touch (pressing
 one of this button).
 
 I search this functionality in Asterisk. What versions, and what
 extension functions (or other settings), and what VoIP phones can do
 this?

It’s called presence, it’s in every version of Asterisk you’re likely to find. 
Most SIP handsets support it. All of it depends on how you configure it.

Steve
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Re: [asterisk-users] Unregistered ports on SPAxxxx

2014-08-05 Thread Steven Howes

On 5 Aug 2014, at 17:10, Mike Diehl mdiehlena...@gmail.com wrote:
 All of my SPA112's are running 1.3.2(014).  My SPA8000's are running 5.1.10.

If you do firmware upgrade your 8000s, don’t go past 6.1.3 or it’ll go badly… 
Freezing and requiring power-cycle, clocks stopping (and showing minus 
figures!) and major struggles downgrading again. Had about a dozen of them 
doing the same, eventual downgrade to 6.1.3 and it’s all happy.

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Re: [asterisk-users] Attack on Sip server.

2014-06-27 Thread Steven Howes
On 27 Jun 2014, at 15:37, Anurag Rana anuragrana31...@gmail.com wrote:
 There are lot of requests coming in and I am not able to stop it because I am 
 unable to detect the IP address. 
 I used wireshark to capture the packets.

If you can capture the packet, surely you have the IP? If they intend to get 
the response then the IP header can’t be forged.

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Re: [asterisk-users] Elastix Architecture

2014-05-02 Thread Steven Howes
On 2 May 2014, at 10:07, upendra uppi...@gmail.com wrote:
 Am new to Elastix and wanted to try build new modules in the Elastix , so i 
 want to know how the PHP is running ?? as i see no Apache server inside ?? so 
 wanted to know how its running ? which server and architecture?

This is not an Elastix mailing list, and even if it was I doubt there is 
sufficient information there for anyone to help you. There are multiple 
appliances and software versions, without you saying what you’ve actually got, 
it’s going to be hard to help.

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Re: [asterisk-users] Numbers hackers call

2014-03-26 Thread Steven Howes
On 26 Mar 2014, at 15:05, Michelle Dupuis mdup...@ocg.ca wrote:

 I see a lot of attempts by hackers to call 00972595301123​ or 
 011972595115207​ or variations but that same 972595 is often present.
 
 Can someone break down that dial string with an explanation?  The 011 look 
 like an overseas call (from Americas), while the 972595XX is unclear...

It’s an international call to +972595XX, tried with the 00, 001 and no 
prefix What is confusing?

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Re: [asterisk-users] Numbers hackers call

2014-03-26 Thread Steven Howes

On 26 Mar 2014, at 16:20, Michelle Dupuis mdup...@ocg.ca wrote:

 If this is to 972 area code then the next digits should be 0X or 0XX but they 
 are not.  This differs from what I found documented for that area code - I 
 thought someone from the region might add to the discussion.  Not sure if 
 this reflected a premium service etc.  (But someone jumped in with an 
 explanation)

I never mentioned the 972 area code. It’s a country code - and as others have 
said it’s been mapped to a Palestinian mobile network. I’ve added this to my  
bar list - I’ve seen quite a lot of toll fraud to Palestine (and the middle 
east in general in recent months). If you’re referring to country code, then 
the 0 of the local number is dropped when dialled internationally, see:

https://en.wikipedia.org/wiki/Telephone_numbers_in_Israel

 I'm guessing you have nothing to add to the discussion?  

Think what you will.

Steve
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Re: [asterisk-users] Spammer direct replying to those posting on the users list

2014-03-25 Thread Steven Howes
On 25 Mar 2014, at 14:16, Digium's Asterisk Development Team 
asteriskt...@digium.com wrote:
 We apparently have a spam bot subscribed to the list and replying
 *directly* to anyone who posts on the list.

There’s plenty of people harvesting the list archives too, I get loads of spam 
about gateways etc :(

S
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Re: [asterisk-users] IAXModem or T38Modem?

2014-03-25 Thread Steven Howes
On 25 Mar 2014, at 15:00, Jeff LaCoursiere j...@jeff.net wrote:
 On 03/24/2014 05:50 PM, Thorolf Godawa wrote:
 
 But your carrier has to support T38, when we began to evaluate this some
 years ago, this was not true for all.
 Would you share the provider you are using?  I have had almost zero luck so 
 far.

What country are you in? Every carrier we’ve tried in the UK supported T38 
without problem (so far.. I expect there is a few that don’t). Not sure I can 
name specific carriers as this is the non-commercial list though.

Steve
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Re: [asterisk-users] Enterprise VoIP Trunk

2014-03-06 Thread Steven Howes
Probably should post this to the asterisk-biz list. This is the non-commercial 
discussion list. Post to the -biz list and you’ll probably have loads of sales 
droids happy to help :)

Steve
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Re: [asterisk-users] SIP OPTIONS storm?

2014-02-14 Thread Steven Howes
On 13 Feb 2014, at 18:10, Tim Nelson tnel...@rockbochs.com wrote:
 I recently experienced an odd situation. I have an Asterisk 11.5.0 system 
 (Box A) with a SIP peering to another Asterisk 1.8.23.0 system (Box B). At 
 some point, Box A started sending over 65Mbps of SIP OPTIONS packets to Box 
 A. I do have qualify=yes for the peer on both sides, and the qualifyfreq is 
 not set (aka default of 60secs).

Just because Box B was receiving 65MBps doesn’t mean box A was sending them. I 
suspect it’s probably the same one repeated, due to some kind of network 
problem. Do you have a pcap so you can look for the ID in the packets to see if 
they are the same? Would be good if you can prove A sent them too (traffic 
stats from SNMP monitoring or something).

S
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Re: [asterisk-users] how can I get authenticate from my own server?

2014-01-17 Thread Steven Howes
On 17 Jan 2014, at 02:18, Sean Darcy seandar...@gmail.com wrote:
 I'm  used to seeing fraudulent attempts to authenticate, But now I'm getting 
 them from the server itself.
 
 I have an asterisk server behind a firewalled router. The local subnet is 
 10.10.10.0/24, the server is 10.10.10.100.
 
 Now I'm seeing in the log lots of:
 
 Failed to authenticate device *00sip:*00@10.10.10.100:5060;tag=9c565e6e
 
 How can this happen?

I’d get an actual SIP trace rather than relying on the logs. If you get it at 
IP level, it’s a little harder to spoof (i.e. sometimes the SIP headers contain 
nonsense)

Steve
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Re: [asterisk-users] Allowing calls to m...@mydomain.org securely on Asterisk 11 box?

2014-01-14 Thread Steven Howes
On 14 Jan 2014, at 02:19, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote:
 Thanks for your feedback Paul. The not having outbound trunks is going to be 
 a challenge. 

Why? it’s what contexts were invented for.

Steve
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Re: [asterisk-users] Multiple IAX2 Trunks Load balancing

2013-12-13 Thread Steven Howes
On 13 Dec 2013, at 07:48, Muhammad Usman replyus...@gmail.com wrote:
 Hi - I have 2 Asterisk servers connected using 05 IAX2 trunks. I want to load 
 balance incoming calls over IAX2 trunks. If any trunk goes down the calls 
 traffic will be shared with other available trunks. When it gets Up the 
 script is supposed to perform as desired i.e in load balance mode.

Sounds wonderful.

S
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Re: [asterisk-users] Hack

2013-10-18 Thread Steven Howes
On 18 Oct 2013, at 04:06, John T. Bittner j...@xaccel.net wrote:
 Today I was hacked but caught it very quickly. This is the weird part, they 
 hacked an IP Auth based account by simply knowing the account name.
 
 How is this possible? I am running Asterisk 11.5.0. Now it’s my fault I used 
 a dictionary based account name but how did they bypass the set ip I had 
 under the account for this host.

Did the IP show under sip show peer xxx? If it's realtime it's possible to set 
it and need to prune it / sip reload.

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Re: [asterisk-users] RTP port ranges

2013-09-13 Thread Steven Howes
On 13 Sep 2013, at 11:44, A J Stiles wrote:
 In the Windows world, where you usually don't get the Source Code, you never 
 know what is running on your computer; in which case, you are never sure that 
 there isn't a daemon listening on a particular port number, so it is wise in 
 that case not to leave ports open unnecessarily.  (Though not half as wise as 
 just not running un-audited software in the first place .)

Netstat will tell you what's running on Windows, just like on other platforms.

Steve
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Re: [asterisk-users] Introducing Sippy Cup: SIPp Load Testing Made Easy

2013-08-27 Thread Steven Howes

On 27 Aug 2013, at 15:34, Ben Klang wrote:
 But what's REALLY useful is Sippy Cup's ability to dynamically generate PCAP 
 audio.  If you've ever needed to drive an IVR from SIPp you're probably 
 familiar with the pains - it usually requires capturing an actual call, 
 isolating the RTP, and then giving it to SIPp to play back.  Sippy Cup makes 
 that easier by actually generating uLaw silence interspersed with 
 appropriately timed RFC4733 DTMF.  That alone has saved us tremendous time 
 when tweaking our load test scenarios.

That's awesome.

Steve


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Re: [asterisk-users] Cut off last character of EXTEN

2013-08-20 Thread Steven Howes

On 20 Aug 2013, at 12:25, Pat Collins wrote: 
 Here ya go:
  
 112233# use ${EXTEN:0:6})
 123# use ${EXTEN:0:3})
 123456789# use ${EXTEN:0:9})

I think 'variable length' implied 'unknown length'...

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Re: [asterisk-users] Strange issues with newly rebooted machine

2013-08-07 Thread Steven Howes
On 6 Aug 2013, at 19:28, Mike Diehl wrote:
 We got it fixed!  Our co-lo is in the process of doing a network
 reconfiguration/relocation and had changed their MTU to 1400 during
 the transition.  Once we did the same, everything started to work.

PMTU should take care of that. Are you blocking ICMP somewhere?

S

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Re: [asterisk-users] Microsoft CRM Integration

2013-07-16 Thread Steven Howes
On 16 Jul 2013, at 04:10, Klaverstyn, David C wrote:
 I’m hoping someone can recommend a method to integrate Microsoft CRM with 
 Asterisk.  Preferably an open source product otherwise a commercial product.

Hi,

You've not said what you're trying to integrate... Creating tasks for calls, 
contact lookups, automatic case creation. Either way, all possible with ODBC 
and FreeTDS.

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Re: [asterisk-users] SIP. Call-limit dialstatus

2013-07-03 Thread Steven Howes
On 3 Jul 2013, at 12:28, I.Pavlov wrote:
 [2013-07-03 15:22:27] NOTICE[29728]: chan_sip.c:6003 update_call_counter: 
 Call to peer '0014' rejected due to usage limit of 1
 -- Couldn't call 0014
   == Everyone is busy/congested at this time (0:0/0/0)
 -- Executing [0014@sub_pbxdialco:50] NoOp(SIP/1295-01f8, 
 CHANUNAVAIL) in new stack
  
 I think that isn’t correct. Is it possible to change dialstatus and 
 CDR(disposition) to BUSY-value when call-limit reached?

You could look for CHANUNAVAIL in dialplan and run Busy(), bit of a workaround 
but may work I've not actually used the Busy() app recently, but I assume 
the CDRs would work with that?

S

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Re: [asterisk-users] Jabber

2013-05-23 Thread Steven Howes
On 23 May 2013, at 10:49, bilal ghayyad wrote:
 Facebook and Whatsapp sort-of support XMPP, so we can use Jabber to 
 communicate with them. But, how much jabber channel in asterisk is stable and 
 updated?

You can find out the support status from menuselect

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States
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Re: [asterisk-users] Asterisk 1.6 + Cisco AS5300

2013-03-07 Thread Steven Howes
On 7 Mar 2013, at 10:12, Mickael Monsieur wrote:
 Do you have an explanation?

Put a SIP debug on and we may be able to find one..

Steve

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Re: [asterisk-users] POKE from command line

2013-02-26 Thread Steven Howes
On 26 Feb 2013, at 16:52, Gary Carr wrote:
 Is it possible to issue the POKE to a end point from the CLI? Our asterisk 
 servers is not seeing some end points drop off and I would like to create a 
 script to manually check end points.

http://www.geekinter.net/iaxping.txt

May be of use to you. Just dug it out of my subversion repo of useful bits so 
make need some poking (excuse the pun) to get it running. No warranty etc etc.

Steve
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Re: [asterisk-users] Dialplan / check / tool

2013-02-18 Thread Steven Howes
On 18 Feb 2013, at 17:03, Christopher Harrington wrote:
 On Mon, Feb 18, 2013 at 10:54 AM, Steve Edwards asterisk@sedwards.com 
 wrote:
 ) If the AGI is [...] a compiled executable (C, Fortran, Cobol, assembler, 
 etc.) 
 I'd like to see an AGI written using Fortran or Cobol. 

Don't tempt me ;)

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Re: [asterisk-users] RPM updates

2013-02-06 Thread Steven Howes
On 28 Jan 2013, at 13:55, Steven Howes wrote:
 Who do I need to poke to get the yum repository / RPM files updated? The 
 dahdi RPMs are not up to date with the CentOS kernel versions any more, it's 
 making doing an installation a bit tricky due to dependancies, I'd rather not 
 roll back / remove new kernels if I don't have to..


Cheers for the replies regarding alternative repos. I'm looking to keep using 
the Digium ones, but they're still broken. Guess I'll just have to wait until 
someone at Digium notices :S

Steve
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Re: [asterisk-users] RPM updates

2013-02-06 Thread Steven Howes
On 6 Feb 2013, at 20:06, Rusty Newton wrote:
 - Original Message -
 From: Steven Howes steve-li...@geekinter.net
 On 28 Jan 2013, at 13:55, Steven Howes wrote:
 Who do I need to poke to get the yum repository / RPM files
 updated? The dahdi RPMs are not up to date with the CentOS kernel
 versions any more, it's making doing an installation a bit tricky
 due to dependancies, I'd rather not roll back / remove new kernels
 if I don't have to..
 Cheers for the replies regarding alternative repos. I'm looking to
 keep using the Digium ones, but they're still broken. Guess I'll
 just have to wait until someone at Digium notices :S
 I'm not involved in the build process for RPMs, but it sounds like they are 
 waiting on the dahdi-linux 2.6.2 release to finish the new set of RPMS. I'd 
 throw out an estimate of 1-3 weeks.  


Hi Rusty, thanks for the update. Sounds like it's being done to save wasting 
time building both the old and new dahdi-linux against the new kernel. I can 
see why that might be done. Makes it a bit awkward if we cant build a PBX by 
just yum installing for a few weeks. I'll have to try rolling back to an out of 
date kernel on the box involved - my project is on around the 4 week mark so 
it'd be cutting it a little fine otherwise.

Cheers for the update.

Steve
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Re: [asterisk-users] CallerID external call after Attended Transfer

2013-02-04 Thread Steven Howes
On 4 Feb 2013, at 12:53, Jonas Kellens wrote:
 I call with my cellphone to our public telephone number
 Our receptionist answers the incoming call and does an attended transfer to 
 my colleague ( A )
 Colleague answers and the receptionist tells him that I am on the other side.
 Receptionist transfers the call and I am connected to my colleague ( B )
 
 
 My question is about the CallerID that the colleague sees on his IP-phone.
 
 In step A the colleague sees the CallerID of the receptionist, which I normal.
 In step B, after I am connected to my colleague, the colleague still sees the 
 CallerID of the receptionist (and not my cellphone number).
 
 How come my colleague does not see my cellphone number ? What is the correct 
 setting ( IP-phone ? Asterisk ? ) to obtain this functionality.

It's called connected line ID (it sends clid updates when things change). 
Asterisk supports it in recent versions (i believe 1.8 is sufficient) - your 
handsets may or may not (their method of transfer, and their ability to process 
the updates can affect it's workability). Given you've not mentioned your 
handsets, we cant make that judgement for you.

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Re: [asterisk-users] CallerID external call after Attended Transfer

2013-02-04 Thread Steven Howes
On 4 Feb 2013, at 13:45, Jonas Kellens wrote:
 The IP-phones in this case are Yealink T32G.
 
 What setting is needed in this IP-phone ?

Quick google doesn't turn up any results. Handsets probably dont support it.

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[asterisk-users] RPM updates

2013-01-28 Thread Steven Howes
Hi All,

Who do I need to poke to get the yum repository / RPM files updated? The dahdi 
RPMs are not up to date with the CentOS kernel versions any more, it's making 
doing an installation a bit tricky due to dependancies, I'd rather not roll 
back / remove new kernels if I don't have to..

Cheers

Steve
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Re: [asterisk-users] DIDForSale spam

2013-01-10 Thread Steven Howes
On 10 Jan 2013, at 02:10, Jai Rangi wrote:
 I have removed yours right away. 
 
 Yes, I agree, But just like any company we have purchased/collected email 
 from different source. Also just like any company we are not perfect, we make 
 mistakes. 

Then buy your addresses from different sources, remove EVERYONE from that 
source - it's clearly unclean data.

And fck off, that too.

Steve
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Re: [asterisk-users] Verizon SIP trunking Field Trial

2013-01-03 Thread Steven Howes
On 3 Jan 2013, at 15:13, Michael L. Young wrote:
 So, I am asking the community for any input.  I have read on here and seen on 
 IRC that some in the community are successfully using Asterisk with Verizon 
 SIP.  Verizon was going to check and see if they have any notes about that 
 and those particular setups.  Can anyone help share any information or 
 tidbits on how they were able to sucessfully work with Verizon?

I *think* Verizon require IPSEC for the signalling, so it may be worth reading 
up on configuring IPSEC in Linux (or acquiring additional hardware) whilst 
you're looking at the Asterisk part. This could have just been for a specific 
product / contract or something, I don't recall the details exactly.

S
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Re: [asterisk-users] Top Posting

2013-01-02 Thread Steven Howes
On 2 Jan 2013, at 15:54, Eric Wieling wrote:
 On Sun, Dec 30, 2012 at 2:54 PM, Benny Amorsen benny+use...@amorsen.dk 
 wrote:   
 There is a sanction. People like me will score top posters lower and
 soon not see their posts at all.
 I'm the opposite.  I'm likely not to scroll down 10 pages to see the comments 
 at the end. 

Wouldn't need to if people trimmed their posts properly.

S
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Re: [asterisk-users] Top Posting

2012-12-31 Thread Steven Howes
We should all top *AND* bottom post!

On 31 Dec 2012, at 06:03, isr...@gmail.com wrote:
 Just my pitch in to post
 From a blackberry you can only top post there is no way of bottom posting 
 So if I would have to wait to get to a computer to bottom post I would just 
 never answer

We should all top *AND* bottom post!

(tongue firmly in cheek here..)

S
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Re: [asterisk-users] ODBC Connection Problem

2012-12-10 Thread Steven Howes
On 10 Dec 2012, at 16:13, Christopher Harrington wrote:
 On Mon, Dec 10, 2012 at 5:23 AM, Chandrakant Solanki 
 solanki.chandrak...@gmail.com wrote:
 Password= c3podb@2012
 
 In case you didn't realize you were sending this out publicly to a publicly 
 archived and searchable list, you might want to change that password now. 

Hostname address is RFC1918, he'll probably be ok ;)

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Re: [asterisk-users] Why all the 401 Unauthorized

2012-10-23 Thread Steven Howes
Hi,

SIP registrations typically try to register, are them prompted for a password 
(via a 401 message) it then sends a new request with authentication . This is 
normal.

Steve

On 23 Oct 2012, at 13:26, Jerry Geis wrote:

 I have a connection between two asterisk boxes, both running 1.4.43
 
 The connection is alive and good and working. however, I see a bunch of
 401 Unauthorized messages using wireshark - then it eventually registers again
 just fine.
 
 Why would it not successfully register the first time - every time?
 
 Jerry
 
 
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Re: [asterisk-users] realtime sip peers status

2012-10-22 Thread Steven Howes
On 22 Oct 2012, at 15:21, Ishfaq Malik wrote:
 On Mon, 2012-10-22 at 19:17 +0500, Control Oye wrote:
 I have successfully setup Asterisk realtime. Now I can create
 extensions dynamically. But when I put this command on cli mode
 
 sip show peers
 
 it returns no result.
 
 can any one guide me to fix this problem.
 
 The extensions you have created will not show up in the cli command of
 sip show peers until the sip extensions have tried to connect to the
 asterisk server.

You may also need to cache realtime peers for some of the stats you're probably 
after. There are plenty of guides online for this. Google is your friend.

Steve
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Re: [asterisk-users] asterisk as IVR using 3g usb modem

2012-10-18 Thread Steven Howes

On 18 Oct 2012, at 16:50, Mitul Limbani wrote:
 U would have to write a dahdi module for this 3G modem to help asterisk 
 understand it as standard gsm channel.
 
Look up chan_datacard (i think that's what it's called from memory).

Steve

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Re: [asterisk-users] using analog phones

2012-08-20 Thread Steven Howes
On 20 Aug 2012, at 01:37, Noam Birnbaum wrote:
 A client wants to keep their old Inter-Tel KTS analog phones for budget 
 reasons. Two questions:
 
 1. How could they use these with FreePBX?

You'd need an ATA or an analogue card. And how well they work, depends on if 
they are 'true' analogue, or have extra digital pairs etc.

 2. Would they be losing any features that they currently have with their 
 analog PBX?

You'd need to find out what features they have on their PBX, and compare this 
to FreePBX.

Steve
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Re: [asterisk-users] Problem provisioning Cisco SPA303

2012-08-01 Thread Steven Howes
On 1 Aug 2012, at 07:05, Support wrote:
 I've looked on Cisco's website and Googled around, but I can not find a true 
 example of a provisioning file for this device.  Anything I could find would 
 be 
 enough for me to make a template.

Download the SPC tool from the Cisco site. It includes the capability to 
generate examples.

You can also do http://phoneip/admin/spacfg.xml to get the full config.

Reading up on examples for the spa941/942/504 might help - they're a bit more 
common.

S
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Re: [asterisk-users] HP DL360 G5 better than HP DL360 G7 ?

2012-08-01 Thread Steven Howes

On 1 Aug 2012, at 12:39, D Tucny wrote:
 HP refuse to let you put your data at risk, refusing to activate write 
 caching without a charged battery attached or NV cache.
 
 I personally would like to have the option to override things like this at my 
 own risk, but, HP don't give you that option.

You can enable it in the ACU (if you install it). There's an override in there.

S


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Re: [asterisk-users] Digium IP Phones D40

2012-06-25 Thread Steven Howes
On 25 Jun 2012, at 16:58, Jeff LaCoursiere wrote:
 Actually we get that complaint a lot too (Polycom ring volume).  We
 typically install in hotel environments, and in their back office the
 environment can be noisy, as well as in their restaurants.
 
 I imagine in a typical office environment this wouldn't be an issue...

The old Cisco/Linksys SPA9xx series used to be ear-bleedinly loud. The newer 
SPA5xx series have been seriously toned down. I'm guessing it's all 'safety' 
related :S

Steve
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[asterisk-users] Fwd: Flashphoner

2012-04-27 Thread Steven Howes
Thought this deserved a name and shame!

;)

Steve

Begin forwarded message:

 From: Pavel Ismailov pavel.ismai...@gmail.com
 Date: 27 April 2012 06:58:07 GMT+01:00
 To: steve-li...@geekinter.net
 Subject: Flashphoner
 
 Hello!
 
 My name is Pavel Ismailov 
 and I`m CEO of www.flashphoner.com project.
 
 We noticed that you quite active in Asterisk-user 
 mail list, and would like to offer you buy signature
 in your messages for some monthly price.
 
 Is it interested for you?
 
 --
 Thanks, 
 Pavel Ismailov
 skype: pavel.ismailov
 www.flashphoner.com

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Re: [asterisk-users] Company info

2012-04-20 Thread Steven Howes
Can't tell if this is a transparent attempt at advertising, or...?

S

On 19 Apr 2012, at 22:09, Josué Conti wrote:

 This is your website:
 
 http://www.convergia.com/
 
 Thanks in advanced for any informations.
 
 Best Regards
 
 Josue
 
 Em 19 de abril de 2012 17:11, Josué Conti josueco...@gmail.com escreveu:
 Dear all, 
 Please let me know if anybody have informations about a company called 
 Convergia, like your products, ASR/ACD or more details.
 
 With Best Regards
 
 Josue
 
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Re: [asterisk-users] Voicemail crashs asterisk

2012-04-03 Thread Steven Howes
On 3 Apr 2012, at 16:42, Vik Killa wrote:
 #disasterisk fail
 
 #freeswitch win


#unhelpful comment

S

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Re: [asterisk-users] concurrent channels limit

2012-03-30 Thread Steven Howes
On 30 Mar 2012, at 10:14, Syco wrote:
 Finally the problem is: I cannot manage more than 80 concurrent calls.

What happens on the 81st call?..

S

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Re: [asterisk-users] unsubscribe

2012-03-30 Thread Steven Howes
On 30 Mar 2012, at 10:04, Sean McMaster wrote:
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Re: [asterisk-users] use of Read cmd with AGI

2012-03-02 Thread Steven Howes

On 2 Mar 2012, at 11:35, Kamlesh Kumar wrote:
 $agi = new AGI();
 $agi- exec('Background','press_one0press_two0press_zero0');
 $agi- exec('Read','NUMBER,,1,3');
 $agi- verbose (You have entered.$NUMBER);

You need to use AGI to read the Asterisk variable.. Asterisk variables don't 
magically become PHP ones.. Or you get Asterisk to process it instead using 
${NUMBER}.

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Re: [asterisk-users] India Pune Pri call problem

2012-02-13 Thread Steven Howes
On 13 Feb 2012, at 12:06, virendra bhati wrote:
 You can't set callerid for outgoing calls in case of PRI.

Why not? Every PRI I have used supported it. Is this a carrier-specific thing?

S
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Re: [asterisk-users] Stumped about adding a semi-colon to a variable

2012-02-11 Thread Steven Howes

On 11 Feb 2012, at 13:41, Kevin P. Fleming wrote:
 At this time, there's no way to do it directly in the dialplan

In extensions.conf
[globals]
SEMICOLON=;

Then use ${SEMICOLON}

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Re: [asterisk-users] Major changes between 1.4/1.6.1.8/10?

2012-02-09 Thread Steven Howes
On 9 Feb 2012, at 11:08, Gilles wrote:
 Does someone of a good site/blog that keeps track of new releases of
 Asterisk, and explains what the major changes/features when they do
 occur?

Why not just use the latest version?..

S

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Re: [asterisk-users] Major changes between 1.4/1.6.1.8/10?

2012-02-07 Thread Steven Howes
On 7 Feb 2012, at 14:27, Gilles wrote:
 On Mon, 06 Feb 2012 10:24:42 -0600 (CST), Richard Mudgett
 rmudg...@digium.com wrote:
 The UPGRADE.txt and CHANGES files do just that.  They have been a part
 of the Asterisk source files for a long time.
 
 Thanks for the info. The problem is that the ChangeLog files
 
 http://downloads.asterisk.org/pub/telephony/asterisk/releases/
 
 are very long to read, and make no distinction between tiny
 features/bug fixes and major changes, so non-experts are unable to
 tell them apart.

The upgrade files may be more to your tastes than changes files. There is no 
comparison chart that I know of. Just use the latest version that has a 
support-lifetime suitable to your needs.

S
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Re: [asterisk-users] Securing Asterisk

2011-07-27 Thread Steven Howes

On 27 Jul 2011, at 17:11, CDR wrote:
 This is turning into a political issue such as the one in Washington
 and the impending default on US debt.

No, YOU are turning this into a political discussion.

 The point is that a minor change
 in the code would have a dramatic effect on security, and carry a
 lower impact on CPU that using Iptables. The simplicity of the change
 cannot understated. The hackers do not continue sending packets with
 new REGISTER attempts unless they see a response. The would move on.

Much as they do after you firewall them out. Have you ever tried? No? Too busy 
blaming others is suspect.

 Digium is being monarchical about this.

Why do you keep blaming Digium? Asterisk is made by a community.

 It looks like a loss of contact with reality.

Couldn't agree more.

 The vast ecosystem of Digium is made of hundreds
 of people like me. I am being forced now to place Opensips in front of
 Asterisk, in port 5060, set Asterisk to listen at Port 5061, and block
 access to 5061 from outside. Instead of a minor change, I have to
 bring a second application to the picture.

There, problem solved.

 The reason why I find useless using iptables and a rule that bans an
 IP address if it communicates more than a threshold of times, is
 simple. I have customers that hit me 10+ times per seconds from the
 same IP. It would look like a hacker, and it is not.

Which is why you don't use packet count, you look in the logs for auth failures.

 I use a cluster of Asterisk in the same box, a big server, and each asterisks 
 listens
 in its own network interface, and responds from it. It does work. But
 iptables or fail2ban would not work in a wholesale scenario.
 Any way, thanks for your attention.

Sure it would. If they're hacking one, you can block them from the lot.. I see 
no problem. Just make it look at all of the logs.

S
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[asterisk-users] Asterisk 1.6.2.19 RPM

2011-07-01 Thread Steven Howes
Hi All,

Asterisk 1.6.2.19 was released on the 28th, does anyone know if there a 
timescale for this reaching the RPM repository? We're badly affected by a bug 
in previous versions that has only recently become apparent to us. It's in a 
situation where rebuilding from source isn't too practical so we need to rely 
on RPMs. Failing that, is there a way I can build an RPM for myself on another 
box and have the right compile-time options (i.e. matching what would be used 
for a real RPM)?

Thanks in advance..

Steve
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Re: [asterisk-users] Re : Re : Re : Re : Direct RTP with Asterisk

2011-06-20 Thread Steven Howes
On 20 Jun 2011, at 16:33, Sagbo Romaric wrote:
 Any client behind his NAT can talk with another behind his NAT.

Still not possible.. The internet doesn't really work like that. SIP even more 
so.

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Re: [asterisk-users] VOICEMAIL CONFIGURATION

2011-06-15 Thread Steven Howes
On 15 Jun 2011, at 11:20, mahesh katta wrote:
 i DID SOME VOICE MAIL CONFIGURATION. SO HOW CAN YOU RETRIVE THAT VOICEMAIL. 
 WHEN I RETRIVE THE VOCIE MAIL ITS NOT GETTING ANY MSG

A lot of filesystems are case sensitive. Maybe you wrote your configuration in 
caps? This would also explain why you couldn't provide anything from the logs ;)

S
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Re: [asterisk-users] Voxbone numbers

2011-06-03 Thread Steven Howes
On 3 Jun 2011, at 11:57, devr devr wrote:
 My query now is willl all voxbone numbers show up as the operator as Voxbone 
 SA as above.  I wanted to find out who the service provider is on some 
 numbers, I suspected the service  to be voxbone but the operator shows as 
 other companies.
 
 My idea on how voxbone works is that voxobone is a intermediatary enabler 
 with the actual hardware with third parties in which case the operator will 
 show as the hardware owner. Is this acuratrate?  

If you're trying to find the carrier for a number in the UK, start here:

http://www.ofcom.org.uk/static/numbering/index.htm

Won't cover ported numbers though.

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Re: [asterisk-users] Question about null routing calls to DIDs we don't handle

2011-06-02 Thread Steven Howes
On 1 Jun 2011, at 22:50, Jesse Thompson wrote:
 We are managing an Asterisk installation for residential VOIP service, and we 
 are having a problem where all inbound calls to DIDs which are assigned to us 
 by our wholesaler but not yet assigned to a downstream customer get caught in 
 a routing loop.

Put this line:

_NXXNXX = Dial(SIP/${EXTEN}@upstream,120); // numbers not handled here get 
routed upstream

in the 'local' context instead of the other one? Letting a carrier use you 
as a carrier seems like quite a bad idea generally..

S
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Re: [asterisk-users] receive faxes

2011-05-04 Thread Steven Howes
On 4 May 2011, at 15:01, vip killa wrote:
 screw that i just got hylafax to work with IAXMODEM...i refuse to pay 
 digium a dime... supposed to be open-source right?


There is so much wrong with that sentence, I don't know where to start.

On 4 May 2011, at 16:02, vip killa wrote:
 Honestly Digium's Asterisk is not a quality project. Though it has lead the 
 way in innovative open-source VoIP, it's a flawed and chaotic project. Hence, 
 I refuse to pay Digium. Digium seems to make a bazillion dollars off of 
 these flaws by selling commercial support/addons anyway... so that should be 
 worth some bad karma points.

Don't use Asterisk then.

S
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Re: [asterisk-users] Asterisk Export Fax from Wave file

2011-04-21 Thread Steven Howes

On 21 Apr 2011, at 13:46, A J Stiles wrote:
 You *might* be able to recover the document, *if and only if* the recording 
 quality is high enough.  Easiest way to try it is to call up a fax machine  
 (either an actual real one, or a copy of Hylafax)  from Asterisk and play the 
 wav file down the line to it.

But fax requires two-way negotiation right?..

S
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Re: [asterisk-users] Call Center Reporting

2011-04-18 Thread Steven Howes
On 18 Apr 2011, at 11:06, bilal ghayyad wrote:
 I am using Asterisk for Call Center (so agents login, logout, ready, not 
 ready, ... etc). To be able to have a good call center reporting, on what I 
 have to depend? On the CDR of Asterisk or there is another way?
 
 Is there a good open source tool to be used for Asterisk call center 
 reporting?

http://www.google.com/search?q=asterisk+call+cener+reporting

There are certainly some nice commercial ones. Can't comment on OSS stuff. But 
given it's for a call centre, I'd be tempted by the you don't get something 
for nothing approach.

S
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Re: [asterisk-users] changing port 5060 to 5061

2011-04-11 Thread Steven Howes
On 11 Apr 2011, at 10:03, darin iv wrote:
 please send me the ways to change asterisk port from 5060 to 5061 i
 need to configure it because we are already using 5060 port in router
 then we cant use it again we have to configure other sip server so
 please suggest me a way..

Are you trying to be a pain in the arse? You've posted this far too many times, 
and ignored responses.

If you ask a question and someone is kind enough to reply, it's polite to at 
least acknowledge what they suggested.

S
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Re: [asterisk-users] voicemail odbc Length is .....

2011-04-11 Thread Steven Howes
On 11 Apr 2011, at 15:28, vip killa wrote:
 I'm using voicemail ODBC with Asterisk 1.6.2.17.2.
 Why do I see Length is 186545 or something similar but a different number 
 in Asterisk CLI everytime someone leaves a message? 

Because not all messages are the same length. I'd guess it's length in bytes?..

S


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Re: [asterisk-users] asterisk SIP MESSAGE method support

2011-04-07 Thread Steven Howes
On 7 Apr 2011, at 14:32, Deka, Rajib IN MAA SL wrote:
 Is the following is the link for getting the source,
 http://svn.asterisk.org/svn/asterisk/trunk/

Please try not to reply to the entire digest..

S

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Re: [asterisk-users] Call recording - methodology

2011-04-06 Thread Steven Howes

On 6 Apr 2011, at 11:54, Silver Thorne wrote:
 Does anyone know of any opensource or otherwise solutions out there that I 
 can try out?

Asterisk. Google it. If you're too lazy, Google MixMonitor. If you're too lazy 
for that:

http://www.voip-info.org/wiki/view/MixMonitor

S
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Re: [asterisk-users] voicemail call back loop

2011-04-06 Thread Steven Howes
On 6 Apr 2011, at 17:46, vip killa wrote:
 I have externnotify = /var/lib/asterisk/agi-bin/vm_notify.pl so that when 
 someone is left a voicemail it will call the person's mobile phone and prompt 
 them with the new message. The perl script simply originates a call to a 
 persons mobile phone and connects it to their voicemail using VoiceMailMain. 
 Problem is when user hangs up from checking their messages, it runs the 
 externnotify again causing an infinite loop. Has anybody encountered this 
 problem or is there an option to not have it run externnotify after checking 
 messages?

Look at the docs. Externnotify sends mailbox + mailbox count. Make your script 
exit if it's 0 messages.

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Re: [asterisk-users] is downloads.asterisk.org down?

2011-03-31 Thread Steven Howes
On 31 Mar 2011, at 13:52, Sebastian wrote:
 


http://www.downforeveryoneorjustme.com/downloads.asterisk.org

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Re: [asterisk-users] asterisk and fail2ban

2011-03-28 Thread Steven Howes
On 28 Mar 2011, at 14:19, vip killa wrote:
 Yes I followed directions on that page
 Running Asterisk 1.6.1.22, anybody else experiencing this?

How often does fail2ban check the logs? It can only block that often, so if 
more attempts happen in that time period it can't do anything until it knows.

S
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Re: [asterisk-users] What is the most stable version of asterisk?

2011-03-24 Thread Steven Howes
On 24 Mar 2011, at 16:38, Gordon Henderson wrote:
 1.2 has been the most stable version for me.
 
 Same setups with 1.4 +DAHDI has never been as stable with random crashes and 
 re-starts - however they're not predictable and sometimes months apart. I had 
 one instance of 1.2 run for over a year without a hiccup.

I've got a 1.4 process that's been running 2 years, 6 weeks..

S
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Re: [asterisk-users] Fwd: asking for some help

2011-03-24 Thread Steven Howes
On 24 Mar 2011, at 16:46, tahar .H wrote:
 so plz is there any one who can give me a puch to learn this extraordinary 
 Asterisk plz(video things will be better :))

Learn to ask questions. Learn to read books. Learn to use google.

S
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Re: [asterisk-users] Asterisk using as a SIP client

2011-03-23 Thread Steven Howes
On 23 Mar 2011, at 10:40, Nikhil wrote:
 I am planning to use asterisk as a IP phone(Porting asterisk into a hardware).

Interesting..

 Is there any limitations if I use asterisk as a SIP client?,and asterisk has 
 any advantages if use like this?

It's not really designed as a SIP client. It's probably possible, but it's very 
sledgehammer.

S
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Re: [asterisk-users] Multi-Tenant Hosted PBX system with Reseller functionality

2011-03-22 Thread Steven Howes
On 22 Mar 2011, at 01:09, Outback Dingo wrote:
 Even worse... now it smells of a scam

At least their website isn't hideous...

Oh..wait.. ;)

S


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Re: [asterisk-users] SIPAddHeader not working

2011-03-15 Thread Steven Howes

On 15 Mar 2011, at 09:08, Jonas Kellens wrote:
 I also notice the presence of a Remote-Party-ID SIPheader... Where does 
 this come from ?! Not from my dialplan...

sendrpid in your sip.conf

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Re: [asterisk-users] SIPAddHeader not working

2011-03-15 Thread Steven Howes

On 15 Mar 2011, at 11:30, Jonas Kellens wrote:
 On 03/15/2011 12:24 PM, Steven Howes wrote:
 
 On 15 Mar 2011, at 09:08, Jonas Kellens wrote:
 I also notice the presence of a Remote-Party-ID SIPheader... Where does 
 this come from ?! Not from my dialplan...
 
 sendrpid in your sip.conf
 Not really :
 
 [3259]

What about in [general]? 'sip show peer' x might help--
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Re: [asterisk-users] SIPAddHeader not working

2011-03-15 Thread Steven Howes
On 15 Mar 2011, at 15:21, Jonas Kellens wrote:
 On 03/15/2011 12:39 PM, Steven Howes wrote:
 
 On 15 Mar 2011, at 11:30, Jonas Kellens wrote:
 On 03/15/2011 12:24 PM, Steven Howes wrote:
 
 On 15 Mar 2011, at 09:08, Jonas Kellens wrote:
 I also notice the presence of a Remote-Party-ID SIPheader... Where does 
 this come from ?! Not from my dialplan...
 
 sendrpid in your sip.conf
 Not really :
 
 What about in [general]? 'sip show peer' x might help
 vps2301*CLI sip show peer 3259 
snip
  Send RPID: Yes

That's why it's sending RPID. Somewhere in your sip.conf you have sendrpid=yes 
(wether that's peer specific, or in general).  Try setting it to 'no' on the 
peer?

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Re: [asterisk-users] SIPAddHeader not working

2011-03-14 Thread Steven Howes

On 14 Mar 2011, at 15:58, Jonas Kellens wrote:
 dialplan :
 
 exten = 67121212,1,NoOp()
 exten = 67121212,n,Set(CALLERID(all)=3259 3259)
 exten = 67121212,n,SIPAddHeader(P-Preferred-Identity: 
 sip:3259\;user=phone)
 exten = 67121212,n,SIPAddHeader(Privacy: id)
 exten = 67121212,n,Dial(SIP/3259/67121212)
 
 
 CLI :
 
 INVITE sip:67121...@sip.voip.tld SIP/2.0
 Via: SIP/2.0/UDP 192.168.1.106:5063;branch=z9hG4bK-1b5cdb51
 From: VC sip:vo...@sip.voip.tld;tag=cb415736707fb109o2
 To: sip:67121...@sip.voip.tld
 Remote-Party-ID: VC sip:vo...@sip.voip.tld;screen=yes;party=calling
 Call-ID: 2a80707a-bdb7c895@192.168.1.106
 CSeq: 101 INVITE
 Max-Forwards: 70
 Contact: VC sip:voip2@192.168.1.106:5063
 Expires: 240
 User-Agent: Linksys/SPA941-5.1.8
 Content-Length: 399
 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
 Supported: replaces
 Content-Type: application/sdp

That's the invite from the phone, not from Asterisk... no?

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Re: [asterisk-users] Asterisk 1.8 paging with ploycom

2011-03-14 Thread Steven Howes
On 14 Mar 2011, at 16:24, satish patel wrote:
 I test page application and it works but i am worried about i have 200 SIP 
 phone. Do you think asterisk page application can handle that number of page 
 ? 

Do they support multicast?

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Re: [asterisk-users] Testing from where number is...

2011-03-03 Thread Steven Howes
On 3 Mar 2011, at 20:53, Danny Nicholas wrote:
 Not having an in-depth knowledge of how EU numbering works

Sadly there is no 'EU numbering'. Europe isn't a country, thus doesn't share 
any dial plan. There appears to be some tendency towards having a '7' at the 
front of a mobile number, but it's far from universal. Each telco/regulating 
body will probably supply a list of codes. OFCOM in the UK do. Failing that, 
putting the country code (e.g. +44) into google gives a fairly rough outline 
for each country.

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Re: [asterisk-users] asterisk security....again

2011-02-28 Thread Steven Howes
On 28 Feb 2011, at 10:33, Rizwan Hisham wrote:
 The problem I have been experiencing since last month is that some of my 
 customers are getting calls with Asterisk Unknown caller id. Most of them 
 in the middle of the night. And my asterisk server has no record of these 
 calls. The customers were getting irritated as you can imagine. I guessed the 
 only way to receive incoming calls by by-passing the registration server is 
 thru sip-uri calls directly to customers. I have updated the customers atas 
 to not accept any calls from sources other than the registration server. 
 Thats all fine now. But the question is how can anyone know the direct sip 
 uri addresses of our customers.

'asterisk security' is a misleading subject line. Guessing someone just scanned 
some IP addresses and made calls. You need what's called a 'firewall'.

 My guess is that someone has been sniffing my server's sip traffic. In that 
 case what should i do to get rid of the sniffers?

It's hard to sniff without being on a network. Most likely they brute forced 
something?.. Get a firewall, and something to look for brute force attacks.

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Re: [asterisk-users] Asterisk Using as a SIP Client

2011-02-17 Thread Steven Howes
On 17 Feb 2011, at 10:04, Nikhil wrote:
 Do I need to modify chan_phone application to make it works or it is 
 available in net.

Why not use a proper sip client?

S
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Re: [asterisk-users] Asterisk and Cisco Call Manager Express (CME)

2008-10-24 Thread Steven Howes
On 24 Oct 2008, at 03:57, David Gibbons wrote:
 Dare I ask why you want to do this?

Cheaper than buying an AIM-CUE? And certainly more flexible.

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Re: [asterisk-users] : Parking Issue

2008-10-22 Thread Steven Howes

On 22 Oct 2008, at 20:29, Craig Van Ham wrote:


 HI all,

 snip

This appears to be the same message you posted earlier.

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Re: [asterisk-users] Call files

2008-10-16 Thread Steven Howes
On 14 Oct 2008, at 18:05, Christian Victor wrote:
 Steven Howes schrieb:
 Have created a system that involves using call files in the outgoing
 spool folder. On some occasions it retries which is fine is there
 any way to view calls waiting retries from the CLI? Using 1.4 btw.
 Have googled to no avail (although it is near the end of the day so I
 might be being a muppet!)

 One solution would be to look INTO the callfiles. The content of the
 file changes if there is a retry involved. I don't know by heard how
 exactly it changes but afair there is a line stating the retry time.

AHA! I did not think of that. Sure someone said a while ago it just  
sucked them up and removed them. Thanks for your help.

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Re: [asterisk-users] app_confcall build issues

2008-10-16 Thread Steven Howes
On 16 Oct 2008, at 14:57, jonathan augenstine wrote:
 I am trying to build app_confcall and it is failing.  Are there  
 known build issues with this module.  I am running Asterisk 1.6.0- 
 beta9.

Ah yes. 'failing'. I bet that is all it says eh? its not like  
compilers give descriptive errors is it.

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[asterisk-users] Call files

2008-10-14 Thread Steven Howes
Hi All,

Have created a system that involves using call files in the outgoing  
spool folder. On some occasions it retries which is fine is there  
any way to view calls waiting retries from the CLI? Using 1.4 btw.  
Have googled to no avail (although it is near the end of the day so I  
might be being a muppet!)

Thanks in advance.

Steve

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Re: [asterisk-users] Asterisk voicemail

2008-10-14 Thread Steven Howes
On 14 Oct 2008, at 11:00, Chris Rowson wrote:
 Hi folks,

 I'm working on a solution using the Asterisk voicemail component and
 wondered if anyone knew the answer to this question please?

 I understand that Asterisk saves voicemail to
 /var/spool/asterisk/voicemail/context/mailbox/INBOX/  but I
 wondered if * creates the file in memory (or tmp/or wherever) and  
 then
 loads the completed file into that directory, or if it writes the  
 file
 to the directory directly, appending it till the recording is
 finished?

 Sorry to reply to my own post! I notice a tmp directory at
 /var/spool/asterisk/voicemail/context/mailbox/tmp/ I'm wondering
 if this is where the file is created, and then moved to the INBOX
 folder perhaps?

Just do an `asterisk -r` and leave a voicemail. It answers all your  
questions.

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Re: [asterisk-users] Tribox

2008-10-06 Thread Steven Howes
Hi

triXbox.org can answer these questions. Google may also give a  
balanced view. But yes, i can assure you, people are using Trixbox  
from Fonality.

Steve

On 6 Oct 2008, at 10:24, broadband Voice wrote:

 Anyone using Tribox from Fonality. I understand its open source and  
 free. Can I use it for a call center functionality? Thanks.
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Re: [asterisk-users] Creating Asterisk Binary Package

2008-09-29 Thread Steven Howes
Just copy the src folder and do `make install` on each machine?
Then tar and copy the /etc/asterisk folder if config is important too.

On 29 Sep 2008, at 08:41, Jim Boykin wrote:

 Is there a script to create an Asterisk binary package after it is
 compiled on one system.

 We do not want to compile Asterisk of each system where we want to
 run. I am sure there is a way but I could find it.

 Thanks
 Jim

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Re: [asterisk-users] Asterisk on VMware Workstation 6

2008-09-25 Thread Steven Howes
Hi,

Agreed. Asterisk on a VM appears to work sometimes, only if magic is  
involved. It is not the way to run anything for a business.

Steve

On 25 Sep 2008, at 02:36, Dean Collins wrote:

 Mike,

 Buy an asterisk appliance like http://www.taa.com/products-vdex-40.html 
  problem solved.

 If you are worried about good call quality it’s either a dedicated  
 pc or a dedicated appliance, one or the other.




 Cheers,

 Dean

 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 
 ] On Behalf Of Michael J. Liberatore
 Sent: Wednesday, 24 September 2008 8:28 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Asterisk on VMware Workstation 6

 Hi, i am running a small personal asterisk server for my business,  
 and instead of getting a dedicated machine to run linux which would  
 waste power and money i decided to run it on my windows xp sp2  
 machine.  The machine is barely used but it does have some crucial  
 programs i need to run in windows so reformating or dual booting is  
 not an option.

 Its basically a iax2 connection to my voip provider and a sip  
 connection to my phone.  It does work well, but the calls especially  
 the voicemail are all garbarled alot.  Its definetly not the  
 provider or internet connection because i use this provider for many  
 clients asterisk setups and i also even setup a temp. asterisk setup  
 on this very pc to test to make sure it was infact vmware causing  
 the problem.

 I upgraded from vmware player to the latest vmware workstation  
 hoping that would fix the problem since its a better system but it  
 hasnt.  I also installed and compiled the vmware tools when  i  
 installed workstation version.

 Is this a known issue with vmware?  Is there a way to correct the  
 issue either on the windows/vmware side or on the asterisk/linux  
 side?  Any other ways to do this project?  i looked into astwind or  
 something but either couldnt get it to work or it was unreliable.

 thanks

 mike

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Re: [asterisk-users] PRI TE110P Configuration

2008-09-25 Thread Steven Howes

On 25 Sep 2008, at 18:38, Shyju K wrote:

 I was configuring asterisk with TE110P Card.When run zttool
 It is showing  Blue Alarm/Yellow Alarm/Recovering and the
 card's LED is blinking RED and GREEN.
 I have connected 12,45 Lines from ISDN modem(RAD ASMi-52)
 to 12,45 of the PRI card respectively.

 I am using Airtel's(India) ISDN connection

 Plese Help  me to sort it out..

Config?

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Re: [asterisk-users] Voicemail cutting out after about 30 seconds

2008-09-24 Thread Steven Howes
Hi,

We saw this between Asterisk and an Audiocodes gateway. Whilst the  
voicemail is being recorded asterisk is not sending *ANY* rtp. Silence  
detection will always detect silence if it listens to this side of the  
conversation. Adjusting the threshold wont work, you need to find the  
timeout for the gateway which is doing it (not always easy...)

Steve

On 24 Sep 2008, at 12:30, Chris Bagnall wrote:

 Greetings list,

 I've had problems on a few of our asterisk boxes where voicemail  
 tends to cut out after about 30 seconds, despite the maximum message  
 length being set at 240s (4m).

 I've tried reducing the silence detection threshold from 128 down to  
 32, which has helped, but not resolved the issue entirely.

 The calls in question are being delivered to the boxes via IAX or  
 SIP, so it's not a Zaptel gain issue.

 Has anyone else had similar problems? Is it safe to drop the  
 silence detection threshold even lower?

 Thanks in advance.

 Regards,

 Chris



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