Re: [asterisk-users] ATA for large networks

[Lafras Henning]

Hi Asterisk user-list,
I know this sort of mail is frowned upon and I would not normally send 
it, but as it is very topical to the current thread, I hope it may have 
value for some.

Xietel is in the field testing phase of a modular channel bank that 
would be ideal for this application  –  100's of existing points that 
has to be converted to asterisk.

Basic features:
Fxo+Fxs ports.
Scalable from 4 to 64 ports per channel bank.
Wall mount unit
Compact 300mm x 200mm x 100mm deep.
TDMoe protocol
AIX protocol planned
1000m extension cable length.   

Benefits
*Locate 100's of ports in one location on legacy cable.
*Install new banks at point of use –
  use single cat 5 link cable instead of 20/50 pairs cat3.
*Reduce lightning risk for outdoor extensions by
   using Fiber optic between buildings.

Requirements.
*Use a sperate - dedicated TDMoE network.


Expected pricing
$2395 US for 48 ports
$1395 US for 24 ports

Availability
Q1 2009 


Regards
Lafras

www.xietel.com


Vinícius Fontes wrote:
 I installed a few Spidermux units on some clients. It works fine, but I had 
 some trouble with cordless analog phones connected to it. The ring voltage is 
 pretty low and that caused some phones to not ring at all. And the ring 
 voltage isn't configurable.
 
 Aside from that, it is a good product. Too bad it's the only TDMoE channel 
 bank (that I know of, at least).
 
 Atenciosamente,
 
 Vinícius Fontes
 Núcleo de Tecnologias Convergentes
 Canall Tecnologia em Comunicações
 Passo Fundo - RS - Brasil
 +55 54 2104-7000
  
 Convergent Technologies Core
 Canall Tecnologia em Comunicações
 Passo Fundo - RS - Brazil
 +55 54 2104-7000
 
 - Vieri [EMAIL PROTECTED] escreveu:
 
 Thanks for the feedback.

 I'm particularly curious to know if anyone has tried a TDMoE channel
 bank. Spidermux seems to be one of the few vendors available. It's the
 closest I can get to an ATA-like device (ie. no special hardware,
 just ethernet) and it also offers an easy failover mechanism to
 another Asterisk server on the LAN.
 So I'm wondering why TDMoE channel banks aren't that popular (am I
 wrong?)? Is Asterisk's native TDMoE implemntation unreliable?
 Has anyone tested Spidermux or other TDMoE channel bank manufacturer?

 Standard channel banks become expensive when one has to buy the
 T1/E1 PCI cards on the Asterisk server. Since most channel banks
 interface with T1 (24 channels) and if I have about 350 analog phones
 to connect then I'd need around 15 T1s (that's about 4 quad-pri T1
 cards which of course require 4 PCI slots and a fair amount of cash).

 Xorcom's Astribank is something in-between. It doesn't require PCI
 slots or T1 cards but connects via USB. Just like T1 channel banks,
 Astribanks don't  seem to offer an easy failover mechanism like in the
 TDMoE solution (correct me if I'm wrong). However, a potentially
 higher number of astribanks can be cheaply connected to a single
 Asterisk server via several USB ports. I'm wondering how many 24-FXS
 astribanks can be connected via USB 2.0 to a single 4-USB-port
 server.

 Of the three solutions I'd try the TDMoE device but I'm wondering why
 noone in this thread even mentioned the protocol.



   

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Re: [asterisk-users] Troubleshooting one-way voice... how to peek into SIP RTP?

[Philip Prindeville]

Andres wrote:
 I'll look into using Record() or Monitor() to capture the phone call, 
 but if there's any conversion being done by codecs then that won't 
 eliminate the possibility that the code itself is misconfigured or buggy 
 and generating a bad stream on one of the legs...

 Anyone have an idea about how to best go about troubleshooting this?


   
 Use tcpdump to capture to a file both call scenarios.  Then use 
 Wireshark to open the file.  You can then do an 'RTP- Show All Streams' 
 Analysis of the calls.  That alone would reveal whether the Audio is 
 really there or not.  You can export that G711 Payload and listen to it 
 with the Windows Media Player.
   

I'm running wireshark 1.0.3.  I've opened the captures...  How do I 
examine the streams?  I don't follow what you're saying above.

And does anyone have a plugin that would allow actual playback of the 
.pcap files' audio packets?

Thanks,

-Philip

 If you don't see the RTP in one direction then you might have a 
 signalling problem.

 Andres
 http://www.neuroredes.com

   
 Thanks,

 -Philip
  


   


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Re: [asterisk-users] test call generator

[Steve Totaro]

If you have some time, interest and desire, I would like to see how
FreeSwitch compares to the 9 calls per second lost SIP message issue.

On Sun, Sep 28, 2008 at 2:55 AM, Gnu Devel [EMAIL PROTECTED] wrote:


 I'm using Sipp to load test, but it  lost some SIP message when I
 increment Call Per Second more than 9.

 Regards

 Grey Man escribió:
  I've used both the Hammer Call Analyzer software and als to the Hammer
  XMS system which is a server that they install in your rack to do the
  packet captures and provide you with all sorts of statistics.
 
  I suspect the Empirix Hammer products would be able to take care of
  any load, monitoring or analysis scenarios you have including
  signalling and media.
 
  The price is going to be the issue. When we looked at the solution the
  Call Analyzer software was 5 figures and the XMS solution was 6.
 
  Regards,
 
  Greyman.
 
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  Version: 8.0.169 / Virus Database: 270.7.3/1694 - Release Date:
 26/09/2008 06:55 p.m.
 
 


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Re: [asterisk-users] test call generator

[Sam Tam]

I looked at the hammer thing.
It is quite complicate and quite useless too
All I want is something that will dial a list of number in schedule per hr
or per 3 hours
Collect the PDD, ASR and comparing it with other route and determine which
is the best.
If the call does not pass through then alert the admin

Obviously hammer can't do that
Sam 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Tuesday, September 30, 2008 4:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] test call generator

If you have some time, interest and desire, I would like to see how
FreeSwitch compares to the 9 calls per second lost SIP message issue.


On Sun, Sep 28, 2008 at 2:55 AM, Gnu Devel [EMAIL PROTECTED] wrote:



I'm using Sipp to load test, but it  lost some SIP message when I
increment Call Per Second more than 9.

Regards

Grey Man escribió:

 I've used both the Hammer Call Analyzer software and als to the
Hammer
 XMS system which is a server that they install in your rack to do
the
 packet captures and provide you with all sorts of statistics.

 I suspect the Empirix Hammer products would be able to take care
of
 any load, monitoring or analysis scenarios you have including
 signalling and media.

 The price is going to be the issue. When we looked at the solution
the
 Call Analyzer software was 5 figures and the XMS solution was 6.

 Regards,

 Greyman.

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 No virus found in this incoming message.
 Checked by AVG - http://www.avg.com
 Version: 8.0.169 / Virus Database: 270.7.3/1694 - Release Date:
26/09/2008 06:55 p.m.





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1.240.938.1212 (cell)



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Re: [asterisk-users] problem with my softphone

[David]

 Hello, when with my client X-lite try to register in the server that 
 say me,
 Registration error:501 Not implemented.
Google is your friend;
http://www.google.com/search?hl=enq=asterisk+register+x-litebtnG=Google+Searchaq=foq=


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Re: [asterisk-users] ATA for large networks

[Johann Steinwendtner]

Gordon Henderson wrote:
 On Mon, 29 Sep 2008, Andres wrote:
 
 In other words, I'd really appreciate feedback from voip administrators 
 (not from resellers) who have had experience testing their devices and are 
 happy with them.



 I would recommend the Linksys SPA8000 (8 port ATA).   It is as solid and
 reliable as the SPA2102.
 
 The OP has 300 phones. That's 38 SPA devices.
 
 And while you might think it's solid and reliable, I have one customer 
 using 3 of them and they're not impressed with echo on their existing 
 analog network.
 
 This is high-end channel bank territory. Multiple E1s - traditional 
 channel banks, or something like multiple 24-port Xorcom units or the 
 like...
 
 Gordon

Does each FXS port need one channel on the E-1 interface, or is there some
sort of concentration ? I can imagine that in a hotel environment, 1 FXS port
to one E1 channel is a waste of resources.

Thanks

Hans


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[asterisk-users] Cisco 7911g

[Sean Lowry]

I have some oddness with this phone. 


The Phone registers with Asterisk (1.4.21), however when I try to make a
call it users the default context and not the one that should be applied
when it registers. 

 

Below are the snippets of the sip.conf and then the debug about the
registration. The config on the phone is the default one found @
http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configu
ration+files+for+SIP

 

Any help on this issue would be really appreciated.


Regards

Sean 

 

[general]

port = 5060 ; Port to bind port

bindaddr = 0.0.0.0  ; Address to bind to

externip = X.X.X.X ; Address that we're going to put in SIP
messages if we're behind a NAT

;localnet = 255.255.255.0; Internal NETWORK address

;localmask = 255.255.255.0   ; Internal netmask

context = bum   ; Default for incoming calls

srvlookup = yes ; Enable SRV lookups on outbound calls

;pedantic = yes ; Enable slow, pedantic checking for
Pingtel

;tos=lowdelay

;tos=184

tos=reliability

maxexpirey=3600 ; Max length of incoming registration we
allow

defaultexpirey=360  ; Default length of incoming/outoing
registration

;notifymimetype=text/plain  ; Allow overriding of mime type in
NOTIFY

videosupport=yes; Turn on support for SIP video

;disallow=all   ; Disallow all codecs

;allow=alaw

allow=g729

 

[7469]

username=7469

secret=11223344

type=peer

context=sip

fromuser=7469

host=dynamic

nat=no

canreinvite=no

callerid=Test Phone 7469

 

--- REGISTRATION ---

--- Transmitting (no NAT) to x.x.x.x:5060 ---

SIP/2.0 100 Trying

Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bKe590c006;received=x.x.x.x

From: sip:[EMAIL PROTECTED];tag=001906af068d0002cce99518-da3b5d4e

To: sip:[EMAIL PROTECTED]

Call-ID: [EMAIL PROTECTED]

CSeq: 101 REGISTER

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Contact: sip:[EMAIL PROTECTED]

Content-Length: 0

 

 



ast-office*CLI 

--- Transmitting (no NAT) to x.x.x.x:5060 ---

SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bKe590c006;received=x.x.x.x

From: sip:[EMAIL PROTECTED];tag=001906af068d0002cce99518-da3b5d4e

To: sip:[EMAIL PROTECTED];tag=as73be6a56

Call-ID: [EMAIL PROTECTED]

CSeq: 101 REGISTER

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

WWW-Authenticate: Digest algorithm=MD5, realm=asterisk,
nonce=75af4945

Content-Length: 0

 

 



Scheduling destruction of SIP dialog
'[EMAIL PROTECTED]' in 32000 ms (Method:
REGISTER)

Sending to x.x.x.x : 5060 (no NAT)

ast-office*CLI 

--- Transmitting (no NAT) to x.x.x.x:5060 ---

SIP/2.0 100 Trying

Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bKd526fdb0;received=x.x.x.x

From: sip:[EMAIL PROTECTED];tag=001906af068d0002cce99518-da3b5d4e

To: sip:[EMAIL PROTECTED]

Call-ID: [EMAIL PROTECTED]

CSeq: 102 REGISTER

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Contact: sip:[EMAIL PROTECTED]

Content-Length: 0

 

 



ast-office*CLI 

--- Transmitting (no NAT) to x.x.x.x:5060 ---

SIP/2.0 200 OK

Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bKd526fdb0;received=x.x.x.x

From: sip:[EMAIL PROTECTED];tag=001906af068d0002cce99518-da3b5d4e

To: sip:[EMAIL PROTECTED];tag=as73be6a56

Call-ID: [EMAIL PROTECTED]

CSeq: 102 REGISTER

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Expires: 3600

Contact: sip:[EMAIL PROTECTED]:5060;transport=udp;expires=3600

Date: Tue, 30 Sep 2008 10:36:03 GMT

Content-Length: 0

 

 



Scheduling destruction of SIP dialog
'[EMAIL PROTECTED]' in 32000 ms (Method:
REGISTER)

 

  CALL  

 

Sending to x.x.x.x : 5060 (no NAT)

Found RTP audio format 0

Found RTP audio format 8

Found RTP audio format 18

Found RTP audio format 116

Found RTP audio format 101

Peer audio RTP is at port x.x.x.x:32384

Found audio description format PCMU for ID 0

Found audio description format PCMA for ID 8

Found audio description format G729 for ID 18

Got unsupported a:fmtp in SDP offer 

Found audio description format iLBC for ID 116

Got unsupported a:fmtp in SDP offer 

Found audio description format telephone-event for ID 101

Got unsupported a:fmtp in SDP offer 

Capabilities: us - 0x8010e (gsm|ulaw|alaw|g729|h263), peer - audio=0x50c
(ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0x10c
(ulaw|alaw|g729)

Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)

Peer audio RTP is at port x.x.x.x:32384

Looking for 7408 in bum (domain 192.168.1.252)

 

--- Reliably Transmitting (no NAT) to x.x.x.x:5060 ---

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 

Re: [asterisk-users] OT: real 2 line phone vs. 1 line and call waiting

[Lyle Giese]

Brian J. Murrell wrote:
 I'm looking into getting a new phone and wondering what the difference
 in functionality is between a single line phone with call waiting and a
 real 2 line phone (either a real SIP phone or an analog 2 line phone and
 a 2 port ATA) is.  Why would I want the real 2 lines vs. just being able
 to take an incoming call via call-waiting?

 Cheers,
 b.

   
1) a two line phone can register with two different * servers or sip 
carriers.

2) It's easy for both incoming and outgoing to separate business from 
personal calls. (ie line1 is personal, line2 is business)

3) It's easy for a two line phone to register to two different accounts 
on * and then subsubscribe to two different MWI's on different VM 
boxes(again goes back to seperating business from personal or your VM 
from your significate other's VM)

That's just off the top of my head.

Lyle

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Re: [asterisk-users] test call generator

[Steve Totaro]

You can't touch this.

Anyways, I am sure I could do it with Hammer, but a tool is just that, use a
screwdriver if you feel a Hammer is too complicated for you.


On Tue, Sep 30, 2008 at 5:25 AM, Sam Tam [EMAIL PROTECTED] wrote:

 I looked at the hammer thing.
 It is quite complicate and quite useless too
 All I want is something that will dial a list of number in schedule per hr
 or per 3 hours
 Collect the PDD, ASR and comparing it with other route and determine which
 is the best.
 If the call does not pass through then alert the admin

 Obviously hammer can't do that
 Sam

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
 Sent: Tuesday, September 30, 2008 4:29 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] test call generator

 If you have some time, interest and desire, I would like to see how
 FreeSwitch compares to the 9 calls per second lost SIP message issue.


 On Sun, Sep 28, 2008 at 2:55 AM, Gnu Devel [EMAIL PROTECTED] wrote:



I'm using Sipp to load test, but it  lost some SIP message when I
increment Call Per Second more than 9.

Regards

Grey Man escribió:

 I've used both the Hammer Call Analyzer software and als to the
 Hammer
 XMS system which is a server that they install in your rack to do
 the
 packet captures and provide you with all sorts of statistics.

 I suspect the Empirix Hammer products would be able to take care
 of
 any load, monitoring or analysis scenarios you have including
 signalling and media.

 The price is going to be the issue. When we looked at the solution
 the
 Call Analyzer software was 5 figures and the XMS solution was 6.

 Regards,

 Greyman.

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 No virus found in this incoming message.
 Checked by AVG - http://www.avg.com
 Version: 8.0.169 / Virus Database: 270.7.3/1694 - Release Date:
 26/09/2008 06:55 p.m.





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 --
 Thanks,
 Steve Totaro
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 1.240.938.1212 (cell)



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1.240.938.1212 (cell)
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Re: [asterisk-users] ATA for large networks

[Jerry Jones]

Google works

enter this along with your search string

site:lists.digium.com your.search.string.here

dont type the 



On Sep 29, 2008, at 2:42 PM, Brian Webster wrote:

What is the best-recommended resource for searching archives of this  
mailing list?


Thanks for your time
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Re: [asterisk-users] ATA for large networks

[Loic Didelot]

I would use Xorcom devices. Its not realy an ATA but you will have less
problems managing an asterisk with a few Xorcoms than many ATA devices.

Also you might have Fax devices and modems in your building and here
Xorcom is definitively a better choice than ATA devices.

Loic.

On Mon, 2008-09-29 at 01:06 -0700, Vieri wrote:
 Hi,
 
 I would like to know if someone can suggest a multi-port ATA worth buying (at 
 least 8 ports).
 
 I have around 380 analog phones to convert to SIP extensions. So I need quite 
 a few ATAs but they need to be enterprise-grade, ie. they need to be 
 reliable and stable.
 
 I bought and have been using 11 Grandstream GXW4008 (8-port FXS ATA) in a 
 production environment and have been experiencing stability and quality 
 issues which are not acceptable in a large company.
 
 I chose Grandstream because:
 
 - it was a cheap way to start
 - I thought their products were stable and reliable because I had already 
 heard their brand name
 
 So since my experience with 11 Grandstream GXW4008 has been overall negative 
 (I need to reboot the devices too often!), I'd like to know if someone could 
 help me decide what brand/model to buy.
 
 I would also need to find these products in Europe (or at least deliverable 
 there).
 
 I've been considering a few products but I don't know if they are reliable:
 
 TopGate TG8048 (48 FXS)
 Soundwin S2400 (24 FXS)
 
 In other words, I'd really appreciate feedback from voip administrators (not 
 from resellers) who have had experience testing their devices and are happy 
 with them.
 
 Thanks,
 
 Vieri
 
 
   
 
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[asterisk-users] Question about Asterisk and Java

[Santiago Panchi]

Hello there.

 I have a problem that I can't solve. I am developing an application
with Java and Asterisk. In addition, I am using Windows Vista, AsteriskWin32
PBX, asterisk-java-0.3.jar and XLite. I startup the DefaultAgiServer without
problems and I have a java application running for the extension
1300(extensions.conf). When I use X-lite and make a call to extension 1300
the application is ok and I can listen to the messages that I put on the
java code. Next, I tried to use the function getData to print the pressed
keys from the softphone. I can listen to the sound that I set for the
function but the answer for the pressed keys is always -1. I can't figure
out the answer to this problem.
Please help me to solve this issue.

Greetings
Santiago
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[asterisk-users] asterisk app store

[Dean Collins]

Just saw this video clip

http://vuclip.com/p?cid=40749677bu=800802t=thumb120x60

 

 

interesting to see that the asterisk app store got announced at astricon
- has anyone seen anything announced on the email list or actual
specifics mentioned?

 

LOL - I hope I got a mention. but seeing I didn't get a free ticket
to Astricon I assume I didn't :-(

 

 

 

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).

 

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Re: [asterisk-users] Question about Asterisk and Java

[Martin Smith]

-1 means Asterisk thinks the command failed. I've seen that if you
hangup on the script, thought it might also happen if the file you
specified doesn't exist. I encourage you to get the latest 1.0 snapshot
from http://asterisk-java.org as we had one parsing bug due to spacing
in the response upon a timeout with no digits pressed. I'd also
encourage you to check out the Asterisk-Java mailing list via
http://asterisk-java.org/development/mail-lists.html.
 
Cheers,

Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221 

 




From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Santiago
Panchi
Sent: Tuesday, September 30, 2008 10:14 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Question about Asterisk and Java


Hello there.

 I have a problem that I can't solve. I am developing an
application with Java and Asterisk. In addition, I am using Windows
Vista, AsteriskWin32 PBX, asterisk-java-0.3.jar and XLite. I startup the
DefaultAgiServer without problems and I have a java application running
for the extension 1300(extensions.conf). When I use X-lite and make a
call to extension 1300 the application is ok and I can listen to the
messages that I put on the java code. Next, I tried to use the function
getData to print the pressed keys from the softphone. I can listen to
the sound that I set for the function but the answer for the pressed
keys is always -1. I can't figure out the answer to this problem.
Please help me to solve this issue.

Greetings
Santiago


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Re: [asterisk-users] asterisk app store

[Steve Totaro]

On Tue, Sep 30, 2008 at 10:22 AM, Dean Collins [EMAIL PROTECTED] wrote:

  Just saw this video clip

 http://vuclip.com/p?cid=40749677bu=800802t=thumb120x60

 interesting to see that the asterisk app store got announced at astricon –
 has anyone seen anything announced on the email list or actual specifics
 mentioned?

 LOL – I hope I got a mention….. but seeing I didn't get a free ticket to
 Astricon I assume I didn't

 Regards,

 Dean Collins
 Cognation Pty Ltd
 [EMAIL PROTECTED]
 +1-212-203-4357 Ph
 +61-2-9016-5642 (Sydney in-dial).


The link you posted is a low quality thumbnail picture of someone I cannot
even recognize with no text or link.

I have no idea what the Asterisk App Store is but the name sounds
commercial, maybe should be on the biz list but again, I have no idea what
you are referring to.  If it is a Digium thing, they usually send emails
directly, off-list.

Finally, to get a free ticket, you need to a speaker and sponsors probably
got some free tickets as well (maybe).

-- 
Thanks,
Steve Totaro
1.888.777.1888
1.240.938.1212 (cell)
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Re: [asterisk-users] Maybe OT - routing calls in PSTN

[Bill Michaelson]

That is my position, and I appreciate the affirmation, as well as the 
offer to determine the carrier. I might email you about that. But having 
no business relationship with the other carrier, it is at best awkward 
for me to initiate contact on this matter, and this should be obvious to 
Vitelity staff. Worse, they are now telling me to contact the user of 
the number to ask them what provider they use. I think this is apalling.


So I'm more concerned with the practicality of relying on Vitelity for 
service in general and in the future. Their tech support has been 
absolutely cavalier to the point of insulting in refusing to deal with 
this basic issue of connectivity. I'm wondering if my experience is unique.

From: Alex Balashov [EMAIL PROTECTED]
It is their responsibility to contact the underlying origination carrier 
to resolve the issue.



  
I have a Vitelity DID which generally works, but calls from a particular 
caller do not reach it.  Vitelity has thus far disavowed any 
responsibility for working through this problem.



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Re: [asterisk-users] asterisk app store

[Dean Collins]

Sorry - my bad, try this link instead
http://www.youtube.com/watch?v=z5yAXBxsCVk 

I didn't realize the other link wasn't the full video.

 

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Totaro
Sent: Tuesday, 30 September 2008 10:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk app store

 

 

On Tue, Sep 30, 2008 at 10:22 AM, Dean Collins [EMAIL PROTECTED]
wrote:

Just saw this video clip

http://vuclip.com/p?cid=40749677bu=800802t=thumb120x60

interesting to see that the asterisk app store got announced at astricon
- has anyone seen anything announced on the email list or actual
specifics mentioned?

LOL - I hope I got a mention. but seeing I didn't get a free ticket
to Astricon I assume I didn't  

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).

 


The link you posted is a low quality thumbnail picture of someone I
cannot even recognize with no text or link.

I have no idea what the Asterisk App Store is but the name sounds
commercial, maybe should be on the biz list but again, I have no idea
what you are referring to.  If it is a Digium thing, they usually send
emails directly, off-list.

Finally, to get a free ticket, you need to a speaker and sponsors
probably got some free tickets as well (maybe).

-- 
Thanks,
Steve Totaro
1.888.777.1888
1.240.938.1212 (cell)



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[asterisk-users] Asterisk Documentation now on voip-info.org Wiki

[Josiah Bryan]

Hey All -

Per a discussion earlier, I've setup a small cron job on one of my 
servers that automatically updates voip-info.org wiki with the latest 
and greatest Asterisk Documentation, straight from svn (specifically, 
the /branches/$version/doc folder for each version.) The files are 
located under 'Asterisk Documentation' on voip-info.org:

Link to the index page:

 http://www.voip-info.org/wiki/view/Asterisk+Documentation


It currently polls the following Asterisk branches from subversion:
 * 1.2
 * 1.4
 * 1.6.0
 * 1.6.1

---
Let me know what you think. If anyone has any questions or comments, 
please do let me know. Oh, and many thanks to James Thompson of 
voip-info.org for his quick response to my questions about an API for 
updating pages. His help was invaluable.

---
Technical Specifics about the Cron Job:
---

The cron job runs daily (about 4am EST) and does an 'svn update' for 
each version's 'doc' folder. If there are any changes, the job uploads 
ONLY the 'text/plain' files in the folder to the wiki (prefixing the 
pages with 'Asterisk Documentation '+$version+' '+$filename, so 
/branches/1.6.1/doc/callfiles.txt becomes 'Asterisk Documentation 1.6.1 
callfiles.txt': 
http://www.voip-info.org/wiki/view/Asterisk+Documentation+1.6.1+callfiles.txt

Note that right now, the files are just passed straight to the wiki and 
quoted in a '~pp~' block (essentially, a pre block) - formatting can 
be applied later if requested - and if presented with a reliable 
formatting algorithm.

Let me know what you all think. Cheers!
-josiah

-- 
Josiah Bryan
IT Manager
Productive Concepts, Inc.
[EMAIL PROTECTED]
(765) 964-6009, ext. 224


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Re: [asterisk-users] Maybe OT - routing calls in PSTN

[Steve Totaro]

On Mon, Sep 29, 2008 at 8:47 PM, Bill Michaelson [EMAIL PROTECTED] wrote:

 I have a Vitelity DID which generally works, but calls from a particular
 caller do not reach it.  Vitelity has thus far disavowed any responsibility
 for working through this problem.  I recognize that some action might be
 required by another provider which is outside Vitelity's control, but it
 seems that they should at least be trying to help resolve the problem by
 helping me determine the responsible party and facilitating contact -
 because it is their DID/service that cannot be reached.

 In the past when I had a similar problem with a Junction DID, the folks at
 Junction resolved it with no hassles and zero intervention on my part.  But
 Vitelity just keeps closing out my trouble tickets while responding in a way
 that indicates that they are not reading my reports carefully.

 How does this compare to others' experiences with Vitelity and other
 providers?  Is there a way that I can determine whom to contact given only
 an originating number?  Any words of wisdom?  Documents I can read for
 educating myself?


I had this issue with VoicePulse a long time ago, they said they didn't
officially support Asterisk and that was obviously the problem (quick easy
way to make me go away).

I leave Asterisk out of most conversations on tech support nowdays.  I get
much further that way.

I said look, humor me, do you have a cell phone?  Dial this number, oh
the call went through?.  Does your desk phone use your system, oh it does,
please humor me and try the same number, oh it didn't go through...?  I
think that eliminates any config issues on my side don't you?  He could not
argue that fact.

It was promptly fixed two days, they had to contact and work with the
other carrier, something about reloading switching tables was the
explanation given to me.  I didn't care, so long as it worked.

I have had no problems with Vitelity but just use them for testing so I
would probably eventually have the same issue.  Do they have a support line
or just a ticket system?

Maybe you could use a take on my above story to help prove your case.

-- 
Thanks,
Steve Totaro
1.888.777.1888
1.240.938.1212 (cell)
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Re: [asterisk-users] Maybe OT - routing calls in PSTN

[Igor Hernandez]

I've had issues with DID service from other providers. My experience has
been hit or miss. Some don't want to deal with any issues, they seem to
think that just because you can run an ITSP without having any lines you
should be exempt from providing any support on the issues that do come
up with the underlying infrastructure.

Others have extremely good tech support. For example, globalpops so far
has been excellent in this department. I've had problems with one of our
DID's and in a matter of around 30 minutes they had contacted the
underlying provider and resolved the issue.

Regards,

-- 
Igor Hernandez
Escape Communications
http://www.escapetel.com

Bill Michaelson wrote:
 That is my position, and I appreciate the affirmation, as well as the
 offer to determine the carrier. I might email you about that. But having
 no business relationship with the other carrier, it is at best awkward
 for me to initiate contact on this matter, and this should be obvious to
 Vitelity staff. Worse, they are now telling me to contact the user of
 the number to ask them what provider they use. I think this is apalling.
 
 So I'm more concerned with the practicality of relying on Vitelity for
 service in general and in the future. Their tech support has been
 absolutely cavalier to the point of insulting in refusing to deal with
 this basic issue of connectivity. I'm wondering if my experience is unique.
 From: Alex Balashov [EMAIL PROTECTED]
 It is their responsibility to contact the underlying origination
 carrier to resolve the issue.


  
 I have a Vitelity DID which generally works, but calls from a
 particular caller do not reach it.  Vitelity has thus far disavowed
 any responsibility for working through this problem.
 
 
 
 
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Re: [asterisk-users] Asterisk Documentation now on voip-info.org Wiki

[Steve Totaro]

On Tue, Sep 30, 2008 at 11:32 AM, Josiah Bryan 
[EMAIL PROTECTED] wrote:

 Hey All -

 Per a discussion earlier, I've setup a small cron job on one of my
 servers that automatically updates voip-info.org wiki with the latest
 and greatest Asterisk Documentation, straight from svn (specifically,
 the /branches/$version/doc folder for each version.) The files are
 located under 'Asterisk Documentation' on voip-info.org:

 Link to the index page:

 http://www.voip-info.org/wiki/view/Asterisk+Documentation


 It currently polls the following Asterisk branches from subversion:
 * 1.2
 * 1.4
 * 1.6.0
 * 1.6.1

 ---
 Let me know what you think. If anyone has any questions or comments,
 please do let me know. Oh, and many thanks to James Thompson of
 voip-info.org for his quick response to my questions about an API for
 updating pages. His help was invaluable.

 ---
 Technical Specifics about the Cron Job:
 ---

 The cron job runs daily (about 4am EST) and does an 'svn update' for
 each version's 'doc' folder. If there are any changes, the job uploads
 ONLY the 'text/plain' files in the folder to the wiki (prefixing the
 pages with 'Asterisk Documentation '+$version+' '+$filename, so
 /branches/1.6.1/doc/callfiles.txt becomes 'Asterisk Documentation 1.6.1
 callfiles.txt':

 http://www.voip-info.org/wiki/view/Asterisk+Documentation+1.6.1+callfiles.txt

 Note that right now, the files are just passed straight to the wiki and
 quoted in a '~pp~' block (essentially, a pre block) - formatting can
 be applied later if requested - and if presented with a reliable
 formatting algorithm.

 Let me know what you all think. Cheers!
 -josiah

 --
 Josiah Bryan
 IT Manager
 Productive Concepts, Inc.
 [EMAIL PROTECTED]
 (765) 964-6009, ext. 224


Coolness.  I will check it out.  Seems like something that should have been
done ages ago.

-- 
Thanks,
Steve Totaro
1.888.777.1888
1.240.938.1212 (cell)
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Re: [asterisk-users] asterisk-users Digest, Vol 50, Issue 89

[Bill Michaelson]

Interesting to see it done. Vitelity claims it is impossible. The number 
is 212-651-5632.



BTW, if you provide the originating number, the underlying carrier can 
be determined, either by the pooling or NANPA block it is assigned to, 
or its LRN if ported.  If you want, you can privately e-mail me the 
number and I'll tell you who the carrier is.




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Re: [asterisk-users] Asterisk Documentation now on voip-info.org Wiki

[Tzafrir Cohen]

On Tue, Sep 30, 2008 at 11:32:41AM -0400, Josiah Bryan wrote:
 Hey All -
 
 Per a discussion earlier, I've setup a small cron job on one of my 
 servers that automatically updates voip-info.org wiki with the latest 
 and greatest Asterisk Documentation, straight from svn (specifically, 
 the /branches/$version/doc folder for each version.) The files are 
 located under 'Asterisk Documentation' on voip-info.org:
 
 Link to the index page:
 
  http://www.voip-info.org/wiki/view/Asterisk+Documentation
 
 
 It currently polls the following Asterisk branches from subversion:
  * 1.2
  * 1.4
  * 1.6.0
  * 1.6.1

Why not link to the SVN instead?

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] Maybe OT - routing calls in PSTN

[Alex Balashov]

Well, they do need to know what carrier the call is being generated from 
in order to troubleshoot routing problems.

Also, it is theoretically common practise for the caller to report the 
number as being unreachable to their carrier, and for carriers to deal 
with these issues between themselves.  Theoretically.

Bill Michaelson wrote:

 That is my position, and I appreciate the affirmation, as well as the 
 offer to determine the carrier. I might email you about that. But having 
 no business relationship with the other carrier, it is at best awkward 
 for me to initiate contact on this matter, and this should be obvious to 
 Vitelity staff. Worse, they are now telling me to contact the user of 
 the number to ask them what provider they use. I think this is apalling.
 
 So I'm more concerned with the practicality of relying on Vitelity for 
 service in general and in the future. Their tech support has been 
 absolutely cavalier to the point of insulting in refusing to deal with 
 this basic issue of connectivity. I'm wondering if my experience is unique.
 From: Alex Balashov [EMAIL PROTECTED]
 It is their responsibility to contact the underlying origination 
 carrier to resolve the issue.


  
 I have a Vitelity DID which generally works, but calls from a 
 particular caller do not reach it.  Vitelity has thus far disavowed 
 any responsibility for working through this problem.
 
 
 
 
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-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] Channel variables materializing ...

[Julian Lyndon-Smith]

Hi Brent,

comments inline:

Brent Davidson wrote:
 Julian Lyndon-Smith wrote:
   
 I am trying to track a strange bug down, and need to ask a really stupid 
 question, just so I can eliminate the possibility ..

 When a SIP channel is hung up, I import a variable called MEETMEROOM 
 from the BRIDGEPEER channel, and if it is set, jump to another part of 
 the dialplan.

 [snip]
 exten = h,1,ImportVar(PARKED=${BRIDGEPEER},MEETMEROOM)
 exten = h,n,GotoIf($[${PARKED} != ]?end)
 exten = h,n,goto(DialStatus,${DIALSTATUS},1)
 exten = h,n(end),NoOp()
 [snip]

 There have been several occasions over the past couple of days where 
 this variable has not executed the goto, and gone to the (end) label 
 when I know for certain that the BRIDGEPEER channel does not have the 
 variable set (I was able to duplicate the error once during a test phase 
 when I was not setting the MEETMEROOM variable at all)

 so, to the stupid question: If at some stage the BRIDGEPEER channel 
 *has* had the MEETMEROOM variable declared, are there any circumstances 
 at all where this variable may be transmitted to the next call that uses 
 this channel.

 There, I asked it. I don't believe that I just did. But there you have 
 it. It's out in the open now ...

 The only other thing that I was thinking of - if the PARKED variable was 
 already set on the SIP channel, would an import of a non-existant 
 variable from the BRIDGEPEER channel overwrite it, or keep it at the 
 previous value ? Hmmm. Time to experiment.

 Julian.

 __
   
 
 This may be a long shot but would it not be better to check to see 
 whether or not the MEETMEROOM variable is defined before assigning it's 
 value to another variable?  With just a cursory glance through the 
   

I am importing it from another channel, so I don't know if it has been 
defined or not. That's the problem :)
 asterisk documentation I have available I don't see any indication of 
 how asterisk variables behave if they are undefined. 

 The other possibility I was considering is maybe BRIDGEPEER is not 
 always being set to the correct channel?
   
Yeah, the thought had crossed my mind as well. Does anyone know of any 
circumstances where this might happen ?
 Good luck,
   

heh. Thanks !

Julian
 -Brent

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Re: [asterisk-users] Problem with pickup extension *8 from features.conf using IAX

[coco]

 Hello
 
Could you please help me understand if this behavior is corect or not?
I did not find something that says that from iax channels i cannot pickup 
ringing ext using the feature defined in features.conf.
Should I open a bug at Digium?
Any of you tryed this feature and worked? so that i could understand if I am 
doing something wrong,
 
So, if anyody used this feature and worked, please tell me so I can understand,
If not, and is a bug, please place your oppinions.
 
Regards,
Cosmin
 
 
 
Hello Cosmin,

I also tried this, and it doesn't work. I think it is a bug but i'm not sure. 
Let us know if you find any solution.

Regards,
Serghei Gutanu



Cosmin Nistor wrote:






 Hello and thank you for replyes.
 
Eric, I looked for it on the mailing list and google and did not find something 
relevant to be 100% sure that this feature is not supported.
 
Some information clare I founded in 
http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups where 
it says that for IAX channels I can use the pickup feature from features.conf.
 
I was looking for an anser to understand if this is supported or not, not to 
lose more time trying to make it work.
 
Shazaum , thank you for your anser, the application Pickup works ok. 
My problem is that this application issued from the dial-plan is
directed pickup, thos means that I have to know the exten that is 
ringing.

I have difficulties because I an using call queues and the channel is not 
anymore only the exten that is ringing, and if I want to pikup a call that is 
comming from a queue, I cannot do this with app Pickup(at least I did not find 
any way to do this--any help from somebody who did is apreciated.)
 
Also, since IAX is developed by asterisk, is strange that for SIP there is 
support, and for IAX, this kind of application is not supported--this is why I 
asked, maybe I am doing something wrong.
 
In this case(if it is not supportted), shoul we/I open a bug repot to Digium? 
 
Botton line, what i am trying to do is to pickup any call that cames in, direct 
call, transfered call, queue call, using IAX, and I am wondering if this is 
possible in any way.
 
Regards,
Cosmin
 
 
I believe chan_iax2 does not support call pickup.  Search the archives.

Shazaum wrote:

 already tested with an exten?
 ex:
 exten = _*8.,1,Pickup(${EXTEN:[EMAIL PROTECTED])
 exten = _*8.,n,Hangup()
 
 2008/9/27 coco [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
 
 Hello list
 
  
 
 I am trying to configure a PBX using Asterisk.
 
 The problem I am havong is the following: I want to use the *8 from
 features.conf to pickup any ringing extension from a group, becouse
 I want to put the users in call queues and I want anybody from the
 company to be able to pick a ringing channel, even if is in a queue.
 
  
 
 Whwn using Sip protocol for the users, everithing is going fine, I
 can pickup any ringing extension from the group using *8.
 
 But the problem appears when I am using IAX protocol. When issuing
 *8 from the IAX phone, asterisk tryes to find the *8 in the dialling
 rules returning:
 
  
 
 *CLI -- Registered IAX2 '40' (AUTHENTICATED) at 10.0.0.30:4569
 http://10.0.0.30:4569
 [Sep 27 12:04:33] NOTICE[19796]: chan_iax2.c:8914 socket_process:
 Rejected connect attempt from 10.0.0.30 http://10.0.0.30, request
 '[EMAIL PROTECTED]' mailto:[EMAIL PROTECTED] does not exist
 
 This I think is wrong, is something like asterisk cannot read from
 features.
 
 With the same setting, when using SIP, i get:
 
  
 
 *CLI   == Using SIP RTP CoS mark 5
 [Sep 27 12:06:23] NOTICE[19802]: chan_sip.c:17092
 handle_request_invite: Nothing to pick up for
 [EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED]
 
 and it works ok.
 
  
 
 I am wondering if any had this problem before and if you can help me
 figure it out(how to make it work--or if is a bug), or find a
 sollution using the app pickup.
 
  
 
 I tryed using asterisk 1.4.13, asterisk 1.4.21.2 http://1.4.21.2,
 asterisk 1.6-rc6 and always the same problem ocurs.
 
  
 
  
 
 Regards,
 
 Cosmin




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Re: [asterisk-users] asterisk app store

[zac wolfe]

Very nice idea -- I hope they follow through and do it soon.  For companies
like mine with limited funds for marketing and who are selling fairly low
cost products, there's very few channels available to get your product seen.
It'd be nice to have a single searchable repository with all the Asterisk
add-ons (open source and otherwise), especially if there was some kind of
standard delivery mechanism.
Zac Wolfe,
Safi Systems LLC
www.safisystems.com


On Tue, Sep 30, 2008 at 8:22 AM, Dean Collins [EMAIL PROTECTED] wrote:

  Sorry – my bad, try this link instead
 http://www.youtube.com/watch?v=z5yAXBxsCVk

 I didn't realize the other link wasn't the full video.





 Regards,

 Dean Collins
 Cognation Pty Ltd
 [EMAIL PROTECTED]
 +1-212-203-4357 Ph
 +61-2-9016-5642 (Sydney in-dial).
   --

 *From:* [EMAIL PROTECTED] [mailto:
 [EMAIL PROTECTED] *On Behalf Of *Steve Totaro
 *Sent:* Tuesday, 30 September 2008 10:58 AM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] asterisk app store





 On Tue, Sep 30, 2008 at 10:22 AM, Dean Collins [EMAIL PROTECTED] wrote:

 Just saw this video clip

 http://vuclip.com/p?cid=40749677bu=800802t=thumb120x60

 interesting to see that the asterisk app store got announced at astricon –
 has anyone seen anything announced on the email list or actual specifics
 mentioned?

 LOL – I hope I got a mention….. but seeing I didn't get a free ticket to
 Astricon I assume I didn't

 Regards,

 Dean Collins
 Cognation Pty Ltd
 [EMAIL PROTECTED]
 +1-212-203-4357 Ph
 +61-2-9016-5642 (Sydney in-dial).




 The link you posted is a low quality thumbnail picture of someone I cannot
 even recognize with no text or link.

 I have no idea what the Asterisk App Store is but the name sounds
 commercial, maybe should be on the biz list but again, I have no idea what
 you are referring to.  If it is a Digium thing, they usually send emails
 directly, off-list.

 Finally, to get a free ticket, you need to a speaker and sponsors probably
 got some free tickets as well (maybe).

 --
 Thanks,
 Steve Totaro
 1.888.777.1888
 1.240.938.1212 (cell)


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Re: [asterisk-users] Asterisk Documentation now on voip-info.org Wiki

[Josiah Bryan]

Tzafrir Cohen wrote:
 On Tue, Sep 30, 2008 at 11:32:41AM -0400, Josiah Bryan wrote:
 Hey All -

 Link to the index page:

  http://www.voip-info.org/wiki/view/Asterisk+Documentation

 Why not link to the SVN instead?

I considered that as well. My thoughts:

1) Ungoogleabelness (if thats a word :-) - since google already ranks 
voip-info.org high on search for asterisk related content, I thought the 
docs should be where the users are, not vis-a-versa.

2) Formatability - the docs are plain text in subversion, whereas 
putting the in the wiki offers the possibility for formatting and 
auto-linking as the algorithm presents itself.

2) UI similarity - linking to the file on svn, for example:

http://svn.digium.com/view/asterisk/branches/1.6.0/doc/callfiles.txt?view=co

Brings just the plain text view, whereas putting it in the wiki offers 
the same UI as the rest of the site.


Note that all these comments are merely my thoughts - feel free to 
comment against them at will. If desired, I can update the index page 
generator to just put links to the svn instead.

Cheers!
-josiah



-- 
Josiah Bryan
IT Manager
Productive Concepts, Inc.
[EMAIL PROTECTED]
(765) 964-6009, ext. 224


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Re: [asterisk-users] test call generator

[zac wolfe]

Sipp looks pretty good! I don't know how I missed this one.  This would've
saved me tons of time a couple months ago.
I plan on using it to load test using 2 Asterisk servers, one to initiate
the SIP calls, the other to receive. Thanks for the tip Alex.

Zac Wolfe
Safi Systems LLC
www.safisystems.com


On Sat, Sep 27, 2008 at 5:58 AM, Alex Balashov [EMAIL PROTECTED]wrote:

 What you are looking for is SIPP:   http://sipp.sourceforge.net/

 It won't intrinsically tell you anything about the data;  it's up to you
 to appropriate the findings.  But it accomplishes the generation of
 traffic (and dummy media!) on a technical level.

 Igor Hernandez wrote:

  Sam Tam wrote:
  Hello everyone
 
 
 
  I am trying to look for a free test call generator that will get me some
  stats like PDD, ASR and call quality etc on each route. As well as do
  test at every interval too
 
 
  If you know something like this please enlighten me.
 
  Sam
 
 
  
 
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 http://lists.digium.com/mailman/listinfo/asterisk-users
 
  Hey Sam,
 
  I've been looking for such a tool also. I can't seem to find a tool that
  does those things.
 
  If nothing comes up in the next couple of weeks I'm going to code
  something up, I wouldn't mind letting you and anyone else who might be
  interested have the source once its done.
 
  Let me know if you find anything thats already out there in the
  meantime, might just save me a few hours of work.
 
  Regards,
 
 


 --
 Alex Balashov
 Evariste Systems
 Web: http://www.evaristesys.com/
 Tel: (+1) (678) 954-0670
 Direct : (+1) (678) 954-0671
 Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] Asterisk Documentation now on voip-info.org Wiki

[Mark Hamilton]

I'm game. It's just perfect the way it is - long overdue!
On my behalf, and behalf of the community (hopefully?), thanks a lot Mr.
Bryan for taking the initiative to get this done.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Josiah Bryan
Sent: September 30, 2008 1:35 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk Documentation now on voip-info.org
Wiki


Tzafrir Cohen wrote:
 On Tue, Sep 30, 2008 at 11:32:41AM -0400, Josiah Bryan wrote:
 Hey All -

 Link to the index page:

  http://www.voip-info.org/wiki/view/Asterisk+Documentation

 Why not link to the SVN instead?

I considered that as well. My thoughts:

1) Ungoogleabelness (if thats a word :-) - since google already ranks 
voip-info.org high on search for asterisk related content, I thought the 
docs should be where the users are, not vis-a-versa.

2) Formatability - the docs are plain text in subversion, whereas 
putting the in the wiki offers the possibility for formatting and 
auto-linking as the algorithm presents itself.

2) UI similarity - linking to the file on svn, for example:

http://svn.digium.com/view/asterisk/branches/1.6.0/doc/callfiles.txt?view=co

Brings just the plain text view, whereas putting it in the wiki offers 
the same UI as the rest of the site.


Note that all these comments are merely my thoughts - feel free to 
comment against them at will. If desired, I can update the index page 
generator to just put links to the svn instead.

Cheers!
-josiah



-- 
Josiah Bryan
IT Manager
Productive Concepts, Inc.
[EMAIL PROTECTED]
(765) 964-6009, ext. 224


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[asterisk-users] Using AMI to View ZAP Channels

[David Budny]

Is there a command similar to sip show inuse for Zap using the AMI? I need to 
be able to see how many channels are in use so I can determine if more calls 
can be sent out using a Zap channel'

Thanks
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Re: [asterisk-users] credit card processing

[Gerald Begumisa]

Hello,

On Sun, Sep 28, 2008 at 1:52 AM, Ruddy Gbaguidi [EMAIL PROTECTED] wrote:
 Hi Guys
 We have a service that can be use by our customer via a website and also
 via telephone.
[...]
 Do you know any company that do this ??

I recently completed implementing such an application - integrated
with www.chasepaymentech.com.  Contact me off-list if you are
interested.

Gerald.

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Re: [asterisk-users] asterisk app store

[Dean Collins]

Here's the details for the conference call where the original proposal
was floated.

http://deancollinsblog.blogspot.com/2008/05/asterisk-3rd-party-ecosystem
.html

 

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of zac wolfe
Sent: Tuesday, 30 September 2008 1:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk app store

 

Very nice idea -- I hope they follow through and do it soon.  For
companies like mine with limited funds for marketing and who are selling
fairly low cost products, there's very few channels available to get
your product seen. It'd be nice to have a single searchable repository
with all the Asterisk add-ons (open source and otherwise), especially if
there was some kind of standard delivery mechanism.

 

Zac Wolfe,

Safi Systems LLC

www.safisystems.com

 

On Tue, Sep 30, 2008 at 8:22 AM, Dean Collins [EMAIL PROTECTED]
wrote:

Sorry - my bad, try this link instead
http://www.youtube.com/watch?v=z5yAXBxsCVk 

I didn't realize the other link wasn't the full video.

 

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Totaro
Sent: Tuesday, 30 September 2008 10:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk app store

 

 

On Tue, Sep 30, 2008 at 10:22 AM, Dean Collins [EMAIL PROTECTED]
wrote:

Just saw this video clip

http://vuclip.com/p?cid=40749677bu=800802t=thumb120x60

interesting to see that the asterisk app store got announced at astricon
- has anyone seen anything announced on the email list or actual
specifics mentioned?

LOL - I hope I got a mention. but seeing I didn't get a free ticket
to Astricon I assume I didn't  

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).

 


The link you posted is a low quality thumbnail picture of someone I
cannot even recognize with no text or link.

I have no idea what the Asterisk App Store is but the name sounds
commercial, maybe should be on the biz list but again, I have no idea
what you are referring to.  If it is a Digium thing, they usually send
emails directly, off-list.

Finally, to get a free ticket, you need to a speaker and sponsors
probably got some free tickets as well (maybe).

-- 
Thanks,
Steve Totaro
1.888.777.1888
1.240.938.1212 (cell)


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Re: [asterisk-users] Asterisk Documentation now on voip-info.org Wiki

[Tzafrir Cohen]

On Tue, Sep 30, 2008 at 01:35:00PM -0400, Josiah Bryan wrote:
 
 Tzafrir Cohen wrote:
  On Tue, Sep 30, 2008 at 11:32:41AM -0400, Josiah Bryan wrote:
  Hey All -
 
  Link to the index page:
 
   http://www.voip-info.org/wiki/view/Asterisk+Documentation
 
  Why not link to the SVN instead?
 
 I considered that as well. My thoughts:
 
 1) Ungoogleabelness (if thats a word :-) - since google already ranks 
 voip-info.org high on search for asterisk related content, I thought the 
 docs should be where the users are, not vis-a-versa.
 
 2) Formatability - the docs are plain text in subversion, whereas 
 putting the in the wiki offers the possibility for formatting and 
 auto-linking as the algorithm presents itself.

Do you intend to add that formatting in your script? They can't be
changed manually.

 
 2) UI similarity - linking to the file on svn, for example:
 
 http://svn.digium.com/view/asterisk/branches/1.6.0/doc/callfiles.txt?view=co
 
 Brings just the plain text view, whereas putting it in the wiki offers 
 the same UI as the rest of the site.

http://svn.digium.com/svn/asterisk/branches/1.6.0/doc/callfiles.txt

Looks better.

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] Asterisk Documentation now on voip-info.org Wiki

[Josiah Bryan]

Tzafrir Cohen wrote:
 On Tue, Sep 30, 2008 at 01:35:00PM -0400, Josiah Bryan wrote:
 Tzafrir Cohen wrote:
 On Tue, Sep 30, 2008 at 11:32:41AM -0400, Josiah Bryan wrote:
 Hey All -

 Link to the index page:

  http://www.voip-info.org/wiki/view/Asterisk+Documentation

 Why not link to the SVN instead?
 I considered that as well. My thoughts:

 1) Ungoogleabelness (if thats a word :-) - since google already ranks 
 voip-info.org high on search for asterisk related content, I thought the 
 docs should be where the users are, not vis-a-versa.

 2) Formatability - the docs are plain text in subversion, whereas 
 putting the in the wiki offers the possibility for formatting and 
 auto-linking as the algorithm presents itself.
 
 Do you intend to add that formatting in your script? They can't be
 changed manually.

The script design supports plugin formatting as it stands. E.g. I can 
insert any formatting algorithm if anyone has any suggestions. Right 
now, the formatter script just does:

#!/usr/bin/perl
use strict;

my $file = $ARGV[0];

print ~pp~\n;
print `cat $file`;
print ~/pp~\n;

Any formatting can be added as desired - this was just a quick way to 
get the content online.


 2) UI similarity - linking to the file on svn, for example:

 http://svn.digium.com/view/asterisk/branches/1.6.0/doc/callfiles.txt?view=co

 Brings just the plain text view, whereas putting it in the wiki offers 
 the same UI as the rest of the site.
 
 http://svn.digium.com/svn/asterisk/branches/1.6.0/doc/callfiles.txt
 
 Looks better.

I agree - if you're looking for the change log. However, I (if I were a 
first-time asterisk user) probably don't care for the change-log-esque 
view, I just want to read the text for myself.

However, I'd be happy to add links to the svn at the bottom of the page 
if that is desired. Thoughts?

Cheers!
-josiah

-- 
Josiah Bryan
IT Manager
Productive Concepts, Inc.
[EMAIL PROTECTED]
(765) 964-6009, ext. 224


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[asterisk-users] How to tell the underlying carrier for your ITSP.

[Alex Balashov]

FYI, for everyone party to the PSTN number discussion:

You can generally tell which underlying carrier your DIDs belong to 
using an information source that aggregates both NANPA (www.nanpa.com) 
10,000 block and Neustar pooling information (www.nationalpooling.com).

Generally, localcallingguide.com is an excellent choice for this, 
although telcodata.us and others that people are fond of work well also. 
  Sometimes the data isn't current, especially because blocks in pooling 
areas or areas with mandatory pooling change hands somewhat frequently.

On LocalCallingGuide.com, if you go to Area Code/Prefix/OCN search and 
put in the NPA-NXX of your DID, you will get the code assignment or 
pooling assignments if the block is pooled, e.g.

   NPA-NXX-Y ... Carrier 1
   NPA-NXX-Z ... Carrier 2
   ...

Pay attention to this pooling information.  A lot of the carriers that 
nationwide DID providers use (XO, Global Crossing, Level3) have pooled 
blocks.  The aggregate 10,000 block information is not going to apply to 
your specific DID in a great deal of MSAs.

Of course, it is very possible that the number is ported, in which case 
figuring out the LRN and OCN/SPID of the carrier is much harder unless 
you are a carrier and have NPAC access.

Contact me privately off-list if you are having problems with a DID and 
I might be able to help you determine the underlying carrier.


-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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[asterisk-users] OT- NIU Framing

[Cory Andrews]

Off topic - is anyone familiar with NIU Framing it is a signaling
method/protocol found sometimes on a DS3.  Wondering if any of the SIP
gateway solutions out there support NIU Framing?

 

Thanks

 

Cory J. Andrews

Director New Market Initiatives

 

Sayers Media Group

VoIP Supply, LLC

454 Sonwil Drive

Buffalo, NY 14225

716-250-3402 OFFICE

716-630-1548 FAX

716-601-4474 MOBILE

[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] 

 

 

Have I exceeded your expectations?  Please share your experience with my
boss,  Benjamin P. Sayers mailto:[EMAIL PROTECTED] , CEO

 

NOTICE: The information contained in this email and any document
attached hereto is intended only for the named recipient(s). It is the
property of the VoIP Supply, LLC and shall not be used, disclosed or
reproduced without the express written consent of VoIP Supply, LLC. If
you are not the intended recipient, nor the employee or agent
responsible for delivering this message in confidence to the intended
recipient(s), you are hereby notified that you have received this
transmittal in error, and any review, dissemination, distribution or
copying of this transmittal or its attachments is strictly prohibited.
If you have received this transmittal and/or attachments in error,
please notify me immediately by reply e-mail or telephone and then
delete this message, including any attachments. Our mailing address is
454 Sonwil Drive, Buffalo, NY 14225 USA. 

 

 

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Re: [asterisk-users] OT- NIU Framing

[Alex Balashov]

I've only heard of M23 and C-bit DS3 framing.

Cory Andrews wrote:

 Off topic – is anyone familiar with “NIU Framing” it is a signaling 
 method/protocol found sometimes on a DS3.  Wondering if any of the SIP 
 gateway solutions out there support NIU Framing?
 
  
 
 Thanks
 
  
 
 *Cory J. Andrews*
 
 Director New Market Initiatives
 
  
 
 *Sayers Media Group*
 
 *VoIP Supply, LLC*
 
 454 Sonwil Drive
 
 Buffalo, NY 14225
 
 716-250-3402 OFFICE
 
 716-630-1548 FAX
 
 716-601-4474 MOBILE
 
 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]_
 
 _ _
 
 _ _
 
 Have I exceeded your expectations?  Please share your experience with my 
 boss,  Benjamin P. Sayers mailto:[EMAIL PROTECTED], CEO
 
  
 
 NOTICE: The information contained in this email and any document 
 attached hereto is intended only for the named recipient(s). It is the 
 property of the VoIP Supply, LLC and shall not be used, disclosed or 
 reproduced without the express written consent of VoIP Supply, LLC. If 
 you are not the intended recipient, nor the employee or agent 
 responsible for delivering this message in confidence to the intended 
 recipient(s), you are hereby notified that you have received this 
 transmittal in error, and any review, dissemination, distribution or 
 copying of this transmittal or its attachments is strictly prohibited. 
 If you have received this transmittal and/or attachments in error, 
 please notify me immediately by reply e-mail or telephone and then 
 delete this message, including any attachments. Our mailing address is 
 454 Sonwil Drive, Buffalo, NY 14225 USA.
 
  
 
  
 
 
 
 
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-- 
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Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] OT- NIU Framing

[Cory Andrews]

Actually, they botched the acronym, it's actually NI-2 or ANSI NI-2.  

Cory J. Andrews
Director New Market Initiatives
 
Sayers Media Group
VoIP Supply, LLC
454 Sonwil Drive
Buffalo, NY 14225
716-250-3402 OFFICE
716-630-1548 FAX
716-601-4474 MOBILE
[EMAIL PROTECTED]


Have I exceeded your expectations?  Please share your experience with my boss,  
Benjamin P. Sayers, CEO

NOTICE: The information contained in this email and any document attached 
hereto is intended only for the named recipient(s). It is the property of the 
VoIP Supply, LLC and shall not be used, disclosed or reproduced without the 
express written consent of VoIP Supply, LLC. If you are not the intended 
recipient, nor the employee or agent responsible for delivering this message in 
confidence to the intended recipient(s), you are hereby notified that you have 
received this transmittal in error, and any review, dissemination, distribution 
or copying of this transmittal or its attachments is strictly prohibited. If 
you have received this transmittal and/or attachments in error, please notify 
me immediately by reply e-mail or telephone and then delete this message, 
including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 
14225 USA. 



-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov
Sent: Tuesday, September 30, 2008 3:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] OT- NIU Framing

I've only heard of M23 and C-bit DS3 framing.

Cory Andrews wrote:

 Off topic - is anyone familiar with NIU Framing it is a signaling 
 method/protocol found sometimes on a DS3.  Wondering if any of the SIP 
 gateway solutions out there support NIU Framing?
 
  
 
 Thanks
 
  
 
 *Cory J. Andrews*
 
 Director New Market Initiatives
 
  
 
 *Sayers Media Group*
 
 *VoIP Supply, LLC*
 
 454 Sonwil Drive
 
 Buffalo, NY 14225
 
 716-250-3402 OFFICE
 
 716-630-1548 FAX
 
 716-601-4474 MOBILE
 
 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]_
 
 _ _
 
 _ _
 
 Have I exceeded your expectations?  Please share your experience with my 
 boss,  Benjamin P. Sayers mailto:[EMAIL PROTECTED], CEO
 
  
 
 NOTICE: The information contained in this email and any document 
 attached hereto is intended only for the named recipient(s). It is the 
 property of the VoIP Supply, LLC and shall not be used, disclosed or 
 reproduced without the express written consent of VoIP Supply, LLC. If 
 you are not the intended recipient, nor the employee or agent 
 responsible for delivering this message in confidence to the intended 
 recipient(s), you are hereby notified that you have received this 
 transmittal in error, and any review, dissemination, distribution or 
 copying of this transmittal or its attachments is strictly prohibited. 
 If you have received this transmittal and/or attachments in error, 
 please notify me immediately by reply e-mail or telephone and then 
 delete this message, including any attachments. Our mailing address is 
 454 Sonwil Drive, Buffalo, NY 14225 USA.
 
  
 
  
 
 
 
 
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-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] Question about Asterisk and Java

[Santiago Panchi]

Thanks for your answer Martin.

The problem was the library. I updated the library to v1.0

Thanks for all
With kind regards

Santiago Panchi

2008/9/30 Martin Smith [EMAIL PROTECTED]

  -1 means Asterisk thinks the command failed. I've seen that if you hangup
 on the script, thought it might also happen if the file you specified
 doesn't exist. I encourage you to get the latest 1.0 snapshot from
 http://asterisk-java.org as we had one parsing bug due to spacing in the
 response upon a timeout with no digits pressed. I'd also encourage you to
 check out the Asterisk-Java mailing list via
 http://asterisk-java.org/development/mail-lists.html.

 Cheers,

 Martin Smith, Systems Developer
 [EMAIL PROTECTED]
 Bureau of Economic and Business Research
 University of Florida
 (352) 392-0171 Ext. 221


  --
 *From:* [EMAIL PROTECTED] [mailto:
 [EMAIL PROTECTED] *On Behalf Of *Santiago Panchi
 *Sent:* Tuesday, September 30, 2008 10:14 AM
 *To:* asterisk-users@lists.digium.com
 *Subject:* [asterisk-users] Question about Asterisk and Java

  Hello there.

  I have a problem that I can't solve. I am developing an
 application with Java and Asterisk. In addition, I am using Windows Vista,
 AsteriskWin32 PBX, asterisk-java-0.3.jar and XLite. I startup the
 DefaultAgiServer without problems and I have a java application running for
 the extension 1300(extensions.conf). When I use X-lite and make a call to
 extension 1300 the application is ok and I can listen to the messages that I
 put on the java code. Next, I tried to use the function getData to print the
 pressed keys from the softphone. I can listen to the sound that I set for
 the function but the answer for the pressed keys is always -1. I can't
 figure out the answer to this problem.
 Please help me to solve this issue.

 Greetings
 Santiago


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[asterisk-users] Transfer a call without announce : no sound

[Nicolas Ross]

When we receive a call from outside (via a sangoma 104d card) and we do a
blind transfer, that is without anouncing to the called party , we have no
sound either way.

Exemple :

I take my cell phone to call my * box, it rings on my aastra 9113i phone, I
answer.
Then I hit the xfer buton, make my second call to another extention (it can
be either a aastra phone, nortel phone trough ciel portico, whatever.
As soon it rings I hangup or hit the xfer buton again.
Then the bridged call between the other extension and the zap channel have
no sound either way.

If I wait for the called party to answer and announce the transfer, all is
fine.

I've had report of sound one way also, but I wasn't able to reproduce.

Here's the log from my console :

   -- SIP/224-09e0f098 answered Zap/1-1
-- Started music on hold, class 'default', on Zap/1-1
-- Executing [EMAIL PROTECTED]:1] Macro(SIP/224-09e1d728,
ael-std-exten|225|SIP/225) in new stack
-- Executing [EMAIL PROTECTED]:1] Set(SIP/224-09e1d728,
ext=225) in new stack
-- Executing [EMAIL PROTECTED]:2] Set(SIP/224-09e1d728,
dev=SIP/225) in new stack
-- Executing [EMAIL PROTECTED]:3] Answer(SIP/224-09e1d728, ) in
new stack
-- Executing [EMAIL PROTECTED]:4] NoOp(SIP/224-09e1d728,
Nicolas Ross 224) in new stack
-- Executing [EMAIL PROTECTED]:5] Wait(SIP/224-09e1d728, 0.5)
in new stack
-- Executing [EMAIL PROTECTED]:6] Dial(SIP/224-09e1d728,
SIP/225|15) in new stack
-- Called 225
-- SIP/225-09e73388 is ringing
-- Stopped music on hold on Zap/1-1
  == Spawn extension (macro-ael-std-exten, s, 6) exited non-zero on
'SIP/224-09e1d728ZOMBIE' in macro 'ael-std-exten'
  == Spawn extension (macro-ael-std-exten, s, 6) exited non-zero on
'SIP/224-09e1d728ZOMBIE'
-- SIP/225-09e73388 answered Zap/1-1

Any ideas ? 


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Re: [asterisk-users] Asterisk Documentation now on voip-info.org Wiki

[Steve Totaro]

On Tue, Sep 30, 2008 at 2:40 PM, Josiah Bryan [EMAIL PROTECTED]
 wrote:

 Tzafrir Cohen wrote:
  On Tue, Sep 30, 2008 at 01:35:00PM -0400, Josiah Bryan wrote:
  Tzafrir Cohen wrote:
  On Tue, Sep 30, 2008 at 11:32:41AM -0400, Josiah Bryan wrote:
  Hey All -
 
  Link to the index page:
 
   http://www.voip-info.org/wiki/view/Asterisk+Documentation
 
  Why not link to the SVN instead?
  I considered that as well. My thoughts:
 
  1) Ungoogleabelness (if thats a word :-) - since google already ranks
  voip-info.org high on search for asterisk related content, I thought
 the
  docs should be where the users are, not vis-a-versa.
 
  2) Formatability - the docs are plain text in subversion, whereas
  putting the in the wiki offers the possibility for formatting and
  auto-linking as the algorithm presents itself.
 
  Do you intend to add that formatting in your script? They can't be
  changed manually.

 The script design supports plugin formatting as it stands. E.g. I can
 insert any formatting algorithm if anyone has any suggestions. Right
 now, the formatter script just does:

 #!/usr/bin/perl
 use strict;

 my $file = $ARGV[0];

 print ~pp~\n;
 print `cat $file`;
 print ~/pp~\n;

 Any formatting can be added as desired - this was just a quick way to
 get the content online.


  2) UI similarity - linking to the file on svn, for example:
 
 
 http://svn.digium.com/view/asterisk/branches/1.6.0/doc/callfiles.txt?view=co
 
  Brings just the plain text view, whereas putting it in the wiki offers
  the same UI as the rest of the site.
 
  http://svn.digium.com/svn/asterisk/branches/1.6.0/doc/callfiles.txt
 
  Looks better.

 I agree - if you're looking for the change log. However, I (if I were a
 first-time asterisk user) probably don't care for the change-log-esque
 view, I just want to read the text for myself.

 However, I'd be happy to add links to the svn at the bottom of the page
 if that is desired. Thoughts?

 Cheers!
 -josiah

 --
 Josiah Bryan
 IT Manager
 Productive Concepts, Inc.
 [EMAIL PROTECTED]
 (765) 964-6009, ext. 224


I think there should be links for changelogs and links for every different
language, so script that up ASAP :-P

I only bother with the changelog to see why something may be broken or if an
upgrade might fix something.  I don't think too many people care about it on
the wiki anyways.

-- 
Thanks,
Steve Totaro
1.888.777.1888
1.240.938.1212 (cell)
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Re: [asterisk-users] OT: real 2 line phone vs. 1 line and call waiting

[Brian J. Murrell]

On Tue, 2008-09-30 at 08:23 -0500, Lyle Giese wrote:

 1) a two line phone can register with two different * servers or sip 
 carriers.

Indeed.  But if I only had the one * server which itself registered to
my carriers...

 2) It's easy for both incoming and outgoing to separate business from 
 personal calls. (ie line1 is personal, line2 is business)

Yeah.  Given this is a home office phone though, that I even route the
house calls to it is just a convenience for when I am in the home
office.  IOW, if I'm in the office, I almost always want to answer it
vs. if I am at a personal/house phone, indeed, I may not want to answer
business calls, but this is not the case...

 3) It's easy for a two line phone to register to two different accounts 
 on * and then subsubscribe to two different MWI's on different VM 
 boxes

Ahhh.  Now this is an interesting possibility.

 (again goes back to seperating business from personal or your VM 
 from your significate other's VM)

Ahhh.  Indeed.

This use case is worth considering.  Although, really, I want to migrate
to VM in IMAP so that I don't even (have to) use the phone to know there
is VM or listen to/delete it.  I would use my e-mail client which is my
preferred interface.

In any case, this one is an interesting benefit.

Not sure I'm convinced enough yet though.  That said, thanks for the
input Lyle, I really appreciate your thoughts on that.

b.



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Re: [asterisk-users] OT: real 2 line phone vs. 1 line and call waiting

[Lyle Giese]

Brian J. Murrell wrote:

On Tue, 2008-09-30 at 08:23 -0500, Lyle Giese wrote:
  
   
1) a two line phone can register with two different * servers or sip 
carriers.



Indeed.  But if I only had the one * server which itself registered to
my carriers...

  
2) It's easy for both incoming and outgoing to separate business from 
personal calls. (ie line1 is personal, line2 is business)



Yeah.  Given this is a home office phone though, that I even route the
house calls to it is just a convenience for when I am in the home
office.  IOW, if I'm in the office, I almost always want to answer it
vs. if I am at a personal/house phone, indeed, I may not want to answer
business calls, but this is not the case...

  
3) It's easy for a two line phone to register to two different accounts 
on * and then subsubscribe to two different MWI's on different VM 
boxes



Ahhh.  Now this is an interesting possibility.

  
(again goes back to seperating business from personal or your VM 
from your significate other's VM)



Ahhh.  Indeed.

This use case is worth considering.  Although, really, I want to migrate
to VM in IMAP so that I don't even (have to) use the phone to know there
is VM or listen to/delete it.  I would use my e-mail client which is my
preferred interface.

  
I have never been convinced that VM via email is a convenence.  You have 
to use the loudspeakers on the PC or headphones, which is not as 
convenient as a handset.  Not to mention the privacy issues/problems 
using loudspeakers for VM.  Do you want your kids/wife overhearing your 
customer that is upset with you?


I find that the email notification is more than enough to know who 
called and many times why without listening to the actual message and 
deciding how urgent it is to listen to the message or deal with it.

In any case, this one is an interesting benefit.

Not sure I'm convinced enough yet though.  That said, thanks for the
input Lyle, I really appreciate your thoughts on that.

b.

  



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[asterisk-users] Asterisk in VM.

[Alex Balashov]

Does anyone have any perspective on how well Asterisk performs and 
scales inside a Xen hypervisor environment?

Obviously, the answer depends largely on what sort of hardware it's 
running on, whether it's in PAE mode, whether it's a newer CPU that has 
some paravirtualisation instruction sets available to assist it, how 
much memory is allocated to each VM, and other architectural 
considerations.

Any perspective would be helpful, however.

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] OT: real 2 line phone vs. 1 line and call waiting

[Brian J. Murrell]

On Tue, 2008-09-30 at 17:29 -0500, Lyle Giese wrote:
 I have never been convinced that VM via email is a convenence.  You
 have to use the loudspeakers on the PC or headphones, which is not as
 convenient as a handset.

Depends on your working environment I guess.

 Not to mention the privacy issues/problems using loudspeakers for VM. 

In my office, that's not usually a problem.

 Do you want your kids/wife overhearing your customer that is upset
 with you?

Heh.  Fortunately I don't get those kinds of calls.

 I find that the email notification is more than enough to know who
 called and many times why without listening to the actual message and
 deciding how urgent it is to listen to the message or deal with it.

The beauty of IMAP VMail storage is that you get the best of both
worlds.  You get the convenience of your VMs accessible by IMAP if you
want or Asterisk can still access them (in the IMAP store) and do most
of the normal VMail functions on them itself.

b.



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[asterisk-users] zap destroy

[Jeff LaCoursiere]

One of my clients today had a POTS line with a bad punch, and no dialtone.
I used zap destroy channel x remotely to keep it from being used to send
outbound calls, which worked fine.  Line repunched, ready again to use,
but how do I undestroy the channel?

In the end I kicked everyone off with zap restart (which for some reason
I had to do twice).  Is there are a more elegant method to deal with this
kind of issue?

Cheers,

j

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[asterisk-users] is DNS SRV enough for failover?

[Nhadie]

hi,

i'm using DNS SRV for failover, i tried to test shutting the server 
down, sip client should still register on the other server but it did 
not.  i'm using x-lite which i don't know if it's doing a srv query. 
does this mean SRV is not enough for failover? if a client has dns 
caching would this cause a problem?

TIA

regards
nhadie

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Re: [asterisk-users] is DNS SRV enough for failover?

[Alex Balashov]

Nhadie wrote:
 hi,
 
 i'm using DNS SRV for failover, i tried to test shutting the server 
 down, sip client should still register on the other server but it did 
 not.  i'm using x-lite which i don't know if it's doing a srv query. 
 does this mean SRV is not enough for failover? if a client has dns 
 caching would this cause a problem?

SRV records are DNS.  DNS is cached.  Ergo, SRV records are cached. 
Ergo, if they are cached excessively - either because the TTL is long, 
or in defiance of the TTL - it can cause a problem.

No, DNS is not a good way to do real-time failover for anything.

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] is DNS SRV enough for failover?

[Andres]

Nhadie wrote:

hi,

i'm using DNS SRV for failover, i tried to test shutting the server 
down, sip client should still register on the other server but it did 
not.  i'm using x-lite which i don't know if it's doing a srv query. 
does this mean SRV is not enough for failover? if a client has dns 
caching would this cause a problem?
  

No it would not.  You should have at least 2 SRV records pointing to 
your 2 servers.  It is ok if your client is caching them if they are not 
supposed to change.  The problem lies in your client.  It should realize 
after a few seconds of trying to register to your main server that it 
cannot so it should try the next one in line according to your SRV Records.

We have deployed thousands of Linksys units configured to query SRV 
records and they work fine in failover scenarios.  I cannot comment on 
X-Lite.

Andres
http://www.neuroredes.com

TIA

regards
nhadie

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[asterisk-users] Cisco Dropping SIP support?

[Michael Graves]

Earlier today I glanced at Junction Networks blog and was surprised to
find a post indicating that Cisco was dropping SIP support in their
79xx series phones. Here's t
link:

http://www.junctionnetworks.com/blog/charlotte/2008/09/19/junction-netwo
rks-lab-cisco-7960-phones

Is this true? What are they thinking? Only SCCP?

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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Re: [asterisk-users] OT: real 2 line phone vs. 1 line and call waiting

[Andrew Joakimsen]

On Tue, Sep 30, 2008 at 9:23 AM, Lyle Giese [EMAIL PROTECTED] wrote:
 1) a two line phone can register with two different * servers or sip
 carriers.

Many phones/ATA with multiple lines only allow 1 server and multiple
registrations!


On Tue, Sep 30, 2008 at 6:29 PM, Lyle Giese [EMAIL PROTECTED] wrote:
 I have never been convinced that VM via email is a convenence.  You have to
 use the loudspeakers on the PC or headphones, which is not as convenient as
 a handset.  Not to mention the privacy issues/problems using loudspeakers
 for VM.  Do you want your kids/wife overhearing your customer that is upset
 with you?

I find it very convenient because I use a Windows Mobile phone with an
Exchange server. So if someone leaves a message while I am out of the
office 1) I am (pretty much) instantly notified 2) I can listen to the
message (after download which takes 2-3 seconds normally) without
having to place a phone call, which avoids using airtime and is just
faster than placing a call, going through the menu, listening to all
other messages, etc. And  I know who the caller is beforehand so I
know if the message needs attention right then and there or if it can
(or should) wait until later. each to his own I suppose.

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[asterisk-users] Polycom 3.1.0RevB

[Andrew Joakimsen]

Could someone please tell me where to download Polycom 3.1.0RevB?
Polycom.com is not possible. Thanks.

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[asterisk-users] No reply to our critical packet

[Andrew Joakimsen]

I am using a Polycom 501 SIP phone behind NAT. Asterisk server is
public with no NAT... everything works on the Asterisk end just fine
EXCEPT that I can never check voice mail

After about 30 seconds the call drops with these messagess:

[Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1950 retrans_pkt: Maximum
retries exceeded on transmission
[EMAIL PROTECTED] for seqno 2 (Critical
Response)
[Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1972 retrans_pkt: Hanging
up call [EMAIL PROTECTED] - no reply to our
critical packet.

It seems to me that the problem is the way Asterisk is handling this
critical packet -- of course it can not be sent to 192.168.1.54, the
phone is at that IP behind a NAT and the Asterisk server is not. I can
make any other phone call from this same phone as long as it is not
voicemail and I can be on the line for hours with no problem.

I am really at a loss here. I have searched a bit and come up with
nothing other than blaming the UA. I know the Polycoms dont have the
best NAT support but besides this it works problem-free. It's odd I
can make a call anywhere else even for hours and not have any issues
at all but 30 seconds into a voicemail call it just drops


app5*CLI sip show peer 17865221569
app5*CLI

  * Name   : 17865221569
  Secret   : Set
  MD5Secret: Not set
  Context  : blended-lcr
  Subscr.Cont. : sla_stations
  Language : en
  AMA flags: Unknown
  Transfer mode: closed
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup:
  Pickupgroup  :
  Mailbox  : 17865221569
  VM Extension : 14193016245
  LastMsgsSent : 0/0
  Call limit   : 2
  Dynamic  : Yes
  Callerid :  CENSORED
  MaxCallBR: 256 kbps
  Expire   : 63
  Insecure : no
  Nat  : Always
  ACL  : No
  T38 pt UDPTL : Yes
  CanReinvite  : No
  PromiscRedir : No
  User=Phone   : Yes
  Video Support: No
  Trust RPID   : No
  Send RPID: No
  Subscriptions: Yes
  Overlap dial : No
  DTMFmode : rfc2833
  LastMsg  : 0
  ToHost   :
  Addr-IP : 74.CENSORED.213 Port 5060
  Defaddr-IP  : 0.0.0.0 Port 5060
  Reg. exten   :
  Def. Username: 17865221569
  SIP Options  : (none)
  Codecs   : 0x104 (ulaw|g729)
  Codec Order  : (g729:20,ulaw:20)
  Auto-Framing:  No
  Status   : OK (130 ms)
  Useragent: PolycomSoundPointIP-SPIP_501-UA/3.0.1.0032
  Reg. Contact : sip:[EMAIL PROTECTED]


app5*CLI core show version
Asterisk 1.4.21.1 built by root @ app5 on a i686 running Linux on
2008-07-09 01:41:43 UTC

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Re: [asterisk-users] G723 on asterisk 1.4.1

[Andrew Joakimsen]

On Sun, Mar 23, 2008 at 11:34 PM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
 It is completely illegal in any country that recognizes patents.

You mean countries that recognize software patents, right?


 Please do NOT discuss ways to use unlicensed codecs on this list or any other 
 forum
 provided by Digium.  This has been discussed multiple times as to why not,
 and I don't feel like rehashing the argument again.

I did not know you were a moderator on this list.

 contributory infringement

What if  I make a page that explains the patent issues and then
provide a link to http://asterisk.hosting.lv/ from that site and only
provide people on this list a link to my site? What if I provide a
link to the Google search for asterisk g723? Where do we draw the
line? If that site is so illegal, why hasn't it been taken down? Why
hasn't the patent holder at the very least provided Google with a DMCA
notice?

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Re: [asterisk-users] ATA for large networks

[Col Ferguson]

We have one hotel using Xorcom devices. It has 1 32 port FXS bank, and 1 24
port FXS + 8 Port FXO.
It works great with all the old analog phones in the motel, over the
existing wiring.
I haven't tried it with a fax though, but modem usage is very hit and miss.
The Xorcom guys are looking into this, as it should work. Apart from that
problem though, I'm very happy with the Xorcom boxes.

To do the 380 extensions though would require 12 of these boxes, so you'd be
using 12 USB connection on a single PC.

Cheers,
Col

- Original Message -
From: Loic Didelot [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Sent: Wednesday, October 01, 2008 12:13 AM
Subject: Re: [asterisk-users] ATA for large networks


 I would use Xorcom devices. Its not realy an ATA but you will have less
 problems managing an asterisk with a few Xorcoms than many ATA devices.

 Also you might have Fax devices and modems in your building and here
 Xorcom is definitively a better choice than ATA devices.

 Loic.

 On Mon, 2008-09-29 at 01:06 -0700, Vieri wrote:
  Hi,
 
  I would like to know if someone can suggest a multi-port ATA worth
buying (at least 8 ports).
 
  I have around 380 analog phones to convert to SIP extensions. So I need
quite a few ATAs but they need to be enterprise-grade, ie. they need to be
reliable and stable.
 
  I bought and have been using 11 Grandstream GXW4008 (8-port FXS ATA) in
a production environment and have been experiencing stability and quality
issues which are not acceptable in a large company.
 
  I chose Grandstream because:
 
  - it was a cheap way to start
  - I thought their products were stable and reliable because I had
already heard their brand name
 
  So since my experience with 11 Grandstream GXW4008 has been overall
negative (I need to reboot the devices too often!), I'd like to know if
someone could help me decide what brand/model to buy.
 
  I would also need to find these products in Europe (or at least
deliverable there).
 
  I've been considering a few products but I don't know if they are
reliable:
 
  TopGate TG8048 (48 FXS)
  Soundwin S2400 (24 FXS)
 
  In other words, I'd really appreciate feedback from voip administrators
(not from resellers) who have had experience testing their devices and are
happy with them.
 
  Thanks,
 
  Vieri
 
 
 
 
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Re: [asterisk-users] is DNS SRV enough for failover?

[Nhadie]

Andres wrote:
 Nhadie wrote:
 
 hi,

 i'm using DNS SRV for failover, i tried to test shutting the server 
 down, sip client should still register on the other server but it did 
 not.  i'm using x-lite which i don't know if it's doing a srv query. 
 does this mean SRV is not enough for failover? if a client has dns 
 caching would this cause a problem?
  

 No it would not.  You should have at least 2 SRV records pointing to 
 your 2 servers.  It is ok if your client is caching them if they are not 
 supposed to change.  The problem lies in your client.  It should realize 
 after a few seconds of trying to register to your main server that it 
 cannot so it should try the next one in line according to your SRV Records.
 
 We have deployed thousands of Linksys units configured to query SRV 
 records and they work fine in failover scenarios.  I cannot comment on 
 X-Lite.

Hi Sir.

This is the result of my query:

~$ host -t SRV _sip._udp.mydomain.com

_sip._udp.mydomain.com has SRV record 0 1 5060 sip-1.mydomain.com.
_sip._udp.mydomain.com has SRV record 0 3 5060 sip-2.mydomain.com.

is that what you meant on having at least 2 SRV record?
does this mean i need a UA capable of querying DNS SRV?

i know it's not a real failover but at least the UA should still try to 
register on the other server if it cannot connect.

thank you

regards,
nhadie









 
 Andres
 http://www.neuroredes.com
 
 TIA

 regards
 nhadie

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