[Asterisk-Users] Grandstream ATA 286 and ilbc

[Anton Krall]

Guys, anybody having problem with ilbc and GS ata 286? I just tried it for
fun (always using alaw) and voices sounded quite bad... crappy voice
prompts, not bad quality, just like weird noises.
 
Anybody had this? whats the latest FW for those units?

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[Asterisk-Users] Spanish TTS

[Anton Krall]

Anybody doing Spanish TTS? what are you using? Festival, Cepstral? 
 
I just tried Cepstral and their male spanish voice is not bad but still
sounds a bit robotic.. anything better? although their licensing price is
quite nice :)
 

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RE: [Asterisk-Users] Web Client with IAX2 and ilbc

[Anton Krall]

I could put something together than can do the auto detect... Can you give
me the urls for those apps you mentioned?

I can post the final html code for the autodetect if anybody wants it (wiki,
etc). 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Matt Riddell
|Sent: Miércoles, 18 de Mayo de 2005 10:23 p.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Web Client with IAX2 and ilbc
|
|Anton Krall wrote:
| Guys.
| 
| Maybe this is asking for a lot :) but is there any web 
|client that can 
| use
| IAX2 and ilbc?
| 
| This is for a call us web idea Any leads?
|
|The problem is that the underlying library comes in different formats. 
|I.E. there is a dll for windows, .so for linux and something 
|else for Mac.
|
|While there have been web-based apps made, they kind of rely 
|on the underlying operating system.
|
|I.E. There is a Java Applet IAX client (but it uses the dll on 
|the back end so is windows only).
|
|There is the ActiveX phone which will require Internet 
|Explorer (and probably Windows)
|
|There is the MozPhone which appears to be cross-platform, but 
|requires Firefox.
|
|Maybe you could build something where the person selects their 
|browser/operating system and that takes you to the appropriate program.
|
|--
|Cheers,
|
|Matt Riddell
|___
|
|http://www.sineapps.com/news.php (Daily Asterisk News - html) 
|http://www.sineapps.com/rssfeed.php (Daily Asterisk News - 
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Re: [Asterisk-Users] VoipSupply.com

[Wilson Pickett]

 support after purchase does not reply to any e-mails. looks like support
 e-mails essentially get's routed to /dev/null 

I had an issue exactly one week ago with a piece of hardware and my
first email to support was answered within an hour with a question
like  (what happens when you...) and an answer to my answer again
within an hour asking me to return the device.

Some things they sell may come only with optional (paid) support,
although you'd think a company would answer then with we do not
support blah...
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[Asterisk-Users] Telecom SIP termination vs. DS3

[Michael Welter]

(Cross posting on purpose)
What is the common wisdom on the list...  find a telco that offers SIP 
termination or wait for Digium's DS3 card?

Who are the telcos that offer SIP termination?
Thanks,
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Re: [Asterisk-Users] Polycom Instant Messaging

[Adrian A]

MSN Messenger does not support SIP, Windows Messenger does.  There's a
difference between the two.

On 5/18/05, Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote:
 Since the Polycom Instant Messaging features uses MSN Messenger, I
 doubt it will work with Asterisk.
 
 C F wrote:
 
  Asterisk can with the sendtext cmd which is available in CVS-HEAD.
 
  On 5/18/05, Chris Coulthurst [EMAIL PROTECTED] wrote:
 
 
 
 
 Can anyone explain the Polycom Text Messaging features built in to the IP
 500/600?   Can Asterisk (or something else) talk to it?  I've seen vague
 references to MSN Messenger, and somehow that's mentally disturbing
 
 
 --
 Always do right. This will gratify some people and astonish the rest.
 Mark Twain
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Re: [Asterisk-Users] Traffic shaping for IAX and SIP calls through Asterisk?

[Luki]

 Is it possible to put some kind of bridge which will do traffic
 shaping/prioritising between my 6 external IP addresses

The Linux kernel provides plenty of queueing disciplines. See the
Advanced Routing  Traffic Control HowTo, chapter 9:

http://www.tldp.org/HOWTO/Adv-Routing-HOWTO/lartc.qdisc.classful.html

--Luki
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Re: [Asterisk-Users] OT: carrying a router, firewall, switch, server, some phones with me on flight to Europe

[Wilson Pickett]

 Last, I might be traveling to Europe (from US)  want to tow along
 hardware  haven't done this before  was wondering what experiences you
 have had  what tips you have. Since this is a bit off topic, feel free
 to reply  not bother the list. (I apologize for wasting everyone's
 bandwidth with this it if is too OT, but I also knew that someone could
 kill this thread in short order, I hope that happens  doesn't create a
 long thread.)

Could be useful info so I'll reply to list. I never come back to
Europe without a bunch of gadgetsn usually both in checked bags and
carry-on. (Assuming you are not travelling by ship?)

The current climate in the US at security checkpoints is ahem,
variable, but I have never had an issue stronger than what is that?.
This last trip my suitcase was inspected, but repacked neatly and the
neither the Digium cards nor the phones suffered.

There is of course one obvious issue, that of powering your equipment
at 220V/50~ and the plug convertors if your are lucky enough to have
power supplies that do 100-250v.

hth
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Re: [Asterisk-Users] Spanish TTS

[Ben Hencke]

Cepstral. They will tune it for $150/hr too. Depending on the
vocabulary of your app, this could still be cheaper than most
commercial TTS engines.
They have 2 new English voices that sound very nice, maybe they will
upgrade their other languages too.
- Ben


On 5/18/05, Anton Krall [EMAIL PROTECTED] wrote:
 Anybody doing Spanish TTS? what are you using? Festival, Cepstral?
 
 I just tried Cepstral and their male spanish voice is not bad but still
 sounds a bit robotic.. anything better? although their licensing price is
 quite nice :)
 
 
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RE: [Asterisk-Users] Polycom Instant Messaging

[Ryan Finnesey]

The LCS 2005 client will also have SIP support.

Ryan
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adrian A
Sent: Thursday, May 19, 2005 2:17 AM
To: Asterisk Users Mailing List - -Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom Instant Messaging

MSN Messenger does not support SIP, Windows Messenger does.  There's a
difference between the two.

On 5/18/05, Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote:
 Since the Polycom Instant Messaging features uses MSN Messenger, I
 doubt it will work with Asterisk.
 
 C F wrote:
 
  Asterisk can with the sendtext cmd which is available in CVS-HEAD.
 
  On 5/18/05, Chris Coulthurst [EMAIL PROTECTED] wrote:
 
 
 
 
 Can anyone explain the Polycom Text Messaging features built in to
the IP
 500/600?   Can Asterisk (or something else) talk to it?  I've seen
vague
 references to MSN Messenger, and somehow that's mentally
disturbing...
 
 
 --
 Always do right. This will gratify some people and astonish the rest.
 Mark Twain
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RE: [Asterisk-Users] SIP Phone Recommendations?

[Paul Hales]

Snom make good gear. Not cheap though.

PaulH 

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Svensson
Sent: Thursday, 19 May 2005 3:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP Phone Recommendations?

On Wed, 18 May 2005, John Mensel wrote:

 Hi all. I'm in the process of putting together a new Asterisk system 
 as a proof-of-concept, and wanted to see which SIP phones all of you 
 had the best luck using with Asterisk.  I've just come off a very 
 trying experience with some Cisco 7960s, and am looking for something 
 else to round out the phones on our network.

Try the Grandstream GXP-2000. With the upcoming firmware it fits our needs 
except for the receptionist. Note that we use headsets instead of speakerphones 
except in conference rooms. If a good two-way speakerphone is needed you should 
look at other phones.

The price is hard to beat. 

Peter


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Re: [Asterisk-Users] Agent Queues and Sending URLs

[Waldo Rubinstein]

I have continued looking for a softphone that can pop up a browser  
window from a URL sent by Asterisk (specifically from the Queue  
application) to no avail.

I'm looking for the following features:
1) SIP based
2) Conference call support
3) Multiple line appearances
4) Can do screen popups (understand the URL passed by Asterisk from  
the Queue application and open a browser)
5) For MS Windows

Does anyone know of such a softphone?
Thanks
Waldo
On May 18, 2005, at 5:25 PM, Waldo Rubinstein wrote:
Thanks. I'm actually looking for a SIP client softphone.
Also, I checked the configuration and there is no space in between  
http:// and www.google.com. It must have gotten inserted when I  
pasted the text.

Any other suggestions?
Waldo
On May 18, 2005, at 2:28 PM, Richard Lyman wrote:

Waldo Rubinstein wrote:

Hi guys,
I'm testing the sending of a URL to an XLite softphone when a  
call is  in queue. See the output of the CLI below:
-- Executing Queue(Zap/69-1, q_sample|tT|http://  
www.google.com/) in new stack
-- Started music on hold, class 'default', on Zap/69-1
-- outgoing agentcall, to agent '1000', on 'Local/  
[EMAIL PROTECTED],1'
-- Called Agent/1000
-- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/1000|20| 
t)  in new stack
-- Called 1000
-- SIP/1000-cb3b is ringing
-- Agent/1000 is ringing
-- Starting simple switch on 'Zap/87-1'
-- SIP/1000-cb3b answered Local/[EMAIL PROTECTED],2
-- Agent/1000 answered Zap/69-1
-- Stopped music on hold on Zap/69-1
  == Spawn extension (agents, 1000, 1) exited non-zero on 'Local/  
[EMAIL PROTECTED],2'
It queues the application correctly. However, when the call is  
sent  to the agent, no URL is displayed. It is a bug in Asterisk  
or is it  that the XLite doesn't support it? Any help will be  
greatly appreciated.
I have heard of others that have been able to do this with  
XLite,  although I haven't actually seen it working or any sample  
configs.
Thanks,
Waldo


last i heard xlite was still working on this ability, try diax g  
or q version (can't remember).  also fix your url string you have  
a space after // and before www.

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RE: [Asterisk-Users] Polycom Instant Messaging

[Chris Coulthurst]

If anyone has had success with IM with these phones in ANY
configuration, I, as well as others Im sure would love to hear about how
its done.  

I envision messages being sent to the phone letting people know about
pending appointments, etc.  I honestly don't care too much about sending
a message back to a user, just receiving one, but the more detail the
better..

Thanks again,

Chris Coulthurst
[EMAIL PROTECTED]
 

|-Original Message-
|From: [EMAIL PROTECTED] [mailto:asterisk-users-
|[EMAIL PROTECTED] On Behalf Of Dean Collins
|Sent: Wednesday, May 18, 2005 6:14 PM
|To: Asterisk Users Mailing List - Non-Commercial Discussion; C F
|Subject: RE: [Asterisk-Users] Polycom Instant Messaging
|
|There is a really good article in this months months Von magazine on
|page 26 about why asterisk will need to adopt sip extensions for
|Microsoft messenger.
|
|I'd post here but you don't accept images. But worth reading.
|
|Cheers,
|Dean
|
|
| -Original Message-
| From: [EMAIL PROTECTED] [mailto:asterisk-users-
| [EMAIL PROTECTED] On Behalf Of Eric Wieling aka ManxPower
| Sent: Wednesday, 18 May 2005 8:41 PM
| To: C F; Asterisk Users Mailing List - Non-Commercial Discussion
| Subject: Re: [Asterisk-Users] Polycom Instant Messaging
|
| Since the Polycom Instant Messaging features uses MSN Messenger, I
| doubt it will work with Asterisk.
|
| C F wrote:
|
|  Asterisk can with the sendtext cmd which is available in CVS-HEAD.
| 
|  On 5/18/05, Chris Coulthurst [EMAIL PROTECTED] wrote:
| 
| 
| 
| 
| Can anyone explain the Polycom Text Messaging features built in to
|the
| IP
| 500/600?   Can Asterisk (or something else) talk to it?  I've seen
|vague
| references to MSN Messenger, and somehow that's mentally
|disturbing...
|
|
| --
| Always do right. This will gratify some people and astonish the rest.
| Mark Twain
| ___
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|
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Re: [Asterisk-Users] zttest

[Jean-Hugues ROBERT]

Hi,
I just migrated my colinux to kernel 2.6.10. And I get really
weird results (voice is just highly distorted slow noise). Here
is zttest...
Opened pseudo zap interface, measuring accuracy...
-799.853516% -799.951172% -800.329590% -799.755859% -799.951172% 
-799.804688% -799.987793%
--- Results after 7 passes ---
Best: 0.00 -- Worst: -800.329590

./ztspeed
Count: 251671
I am running  CVS-HEAD-05/17/05-16:23:07, with no TDM hardware at all
Any idea of what could be wrong ?
Yours,
  JeanHuguesRobert
At 23:26 15/05/2005 -0400, you wrote:
I was browsing the applications developed in zaptel and came across
zttest.
After I run it, I get the following:
Opened pseudo zap interface, measuring accuracy...
99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 100.00%
99.987793%
99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793%
99.987793% 99.975586%
99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793%
99.987793% 99.987793%
99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793%
99.987793% 99.987793%
99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793%
99.987793% 99.987793%
99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793%
99.987793% 99.987793%
100.00% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793%
99.987793% 99.987793%
99.987793% 99.987793%
--- Results after 57 passes ---
Best: 100.00 -- Worst: 99.975586 -- Average: 99.987793
What does this mean? Should I have expected to get 100% across the
board?
This is from a TE410P running on Debian 2.6.11-1-686-smp on a dual
Xeon 2.4GHz server.
Thanks,
Waldo
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-
Web:  http://hdl.handle.net/1030.37/1.1
Phone: +33 (0) 4 92 27 74 17
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RE: [Asterisk-Users] Pickup other ringing phone

[Mark Brown]

Thanks allot everyone; will check it out a bit later.
Helps when you have people who know their stuff on the other end of the
keyboard ;)

Mark

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ariel
Batista
Sent: 19 May 2005 00:59
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Pickup other ringing phone

You need to go into the extensions setup and put the pickupgroup and
callgroup to the same on both.  That way when you hear the other
extension
ring you just dial *8 send and you can pickup the ringing phone call.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Brown
Sent: Wednesday, May 18, 2005 6:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Pickup other ringing phone

Hi everyone,
Is there a simple way of answering a different ringing extension from a
sip phone using AAH?
I have absolutely zero technical know-how when it comes to modifying
conf files etc. Still working on figuring it all out. ;)

That brings me to my second question... where the hell does one find an
extensive manual of sorts that explains all conf files and what the
strings all mean etc?

Cheers All
Mark
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RE: [Asterisk-Users] Static on TDM Zaptel FXO

[Dave Cotton]

On Wed, 2005-05-18 at 15:11 -0600, Rich Adamson wrote:

 
 FWIW, I'm 7db from my central office and am using rxgain=5.0 and
 txgain=1.0 for now.
 

Here in France I'm connected to ADSL using Free and they have a page
available that you can get some info on your line, I'm 1355 metres from
the central with 18dB. I'll play with rx/txgain.

-- 
Dave Cotton [EMAIL PROTECTED]

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[Asterisk-Users] dail out with SIP through a second server

[Arjan Kroon]

Hello,



Im trying to get the following situation.

Someone calls an application on one of our asterisk
server.

In this application the caller will call a SIP
client. (with the command Dial)

The Sip client is connected with another asterisk
server. (see below)

Caller  asterisk01 (incoming server)  asterisk00 (outbound server)  SIP client (X-lite)



Do anybody now how the command is to call a SIP
client on another asterisk server?

It is possible to send the configuration file, just
say which configuration files (sip.conf, etc)



First I tried to a simpler situation, see below

Caller  asterisk00 (inbound/outbound server)  SIP client (X-lite)

This situation worked perfect.



Thanks in advance



Arjan Kroon 

email: [EMAIL PROTECTED] 










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Re: [Asterisk-Users] Follow Me solution

[Vladyslav]

Better take a look at Dial cmd. and on it's possibility to run Macros.
On Thu, 2005-05-19 at 00:19, Ben Johnson wrote:
 I read an article in the wiki on a 
 (http://www.voip-info.org/wiki-Asterisk+Tips+follow+me) that allows Asterisk 
 to forward a call to a cell phone if someone does not answer there office 
 phone.  The example waits for the cell phone user to press the # button 
 before bridging the two calls.  In the example there is the c switch that 
 tells asterisk to wait for the #.  Is there a similar dial statement that 
 would allow me to do this with a IAX2 connection??
 
 Thanks
 Ben
 
 
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Re: [Asterisk-Users] Web Client with IAX2 and ilbc

[Matt Riddell]

Anton Krall wrote:
I could put something together than can do the auto detect... Can you give
me the urls for those apps you mentioned?
I can post the final html code for the autodetect if anybody wants it (wiki,
etc). 

|I.E. There is a Java Applet IAX client (but it uses the dll on 
|the back end so is windows only).
Was
http://www.hem.za.org/jiaxclient/
(seems to be down)
|There is the ActiveX phone which will require Internet 
|Explorer (and probably Windows)
http://www.geocities.com/babarnazmi/middlepage.htm
|There is the MozPhone which appears to be cross-platform, but 
|requires Firefox.
http://www.voip-info.org/wiki-MozPhone
--
Cheers,
Matt Riddell
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[Asterisk-Users] ser+asterisk problem

[Kamran Ahmad]

hello

I am using ser with asterisk

asterisk on 5070 (on back end)
ser on 5060 (on front end)

i am getting all requests at asterisk.

i tried by changing asterisk port
bindport=5090
but still getting all requests from sjphone at
asterisk.

can any one tell what is the reason

regrads
Kamran



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Re: [Asterisk-Users] ser+asterisk problem

[Peter Bowyer]

On 19/05/05, Kamran Ahmad [EMAIL PROTECTED] wrote:
 hello
 
 I am using ser with asterisk
 
 asterisk on 5070 (on back end)
 ser on 5060 (on front end)
 
 i am getting all requests at asterisk.
 
 i tried by changing asterisk port
 bindport=5090
 but still getting all requests from sjphone at
 asterisk.
 
 can any one tell what is the reason

Did you restart Asterisk - that's a complete restart, not just a 'reload'

Peter

-- 
Peter Bowyer
Email: [EMAIL PROTECTED]
Tel: +44 1296 768003
VoIP: sip:[EMAIL PROTECTED]
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[Asterisk-Users] Forbidden - wrong password on authentication for NOTIFY

[c waddy]

Hi,

I am trying to get to the bottom of a warning i am recieving through
the console.

May 18 13:26:29 WARNING[8281]: chan_sip.c:6837 handle_response:
Forbidden - wrong password on authentication for NOTIFY

Calls are still working. I cannot work out what is causing it.

Asterisk - Ingate - Asterisk.

I have googled and cannot find anything on the above.

Thanks.
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[Asterisk-Users] GOTO statement in Realtime-Extensions not working like expected

[niels]

Hi .. When I use the GoTo statement in realtime to goto a priority only
... E.g. Goto(3) then there's no problem

But, If I try to jump to another context ... E.g.
Goto(othercontext,${EXTEN},3) then it doesn't work

If I process the same statement in extensions.conf things go well

Are there things broken regarding GoTo in combination with Realtime
Extensions ?

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[Asterisk-Users] Newbie X100P question

[Fabrice Delambre]

Hello,

I just bought a X100P from digitnetworks.
It is supposed to be a FXO card, but there are 2 rj-11 plug on the card.
One is labelled phone and the other pstn. When i plug the pstn on
the wall and the phone on my analog phone, everything (incoming and
outgoing calls) works like before (without asterisk).
AFAIU, i should have an FXS card in my box to be able to use my analog
phone, so why does it work this way ?

Second question, what is the cheapest card to use one analog phone only
(TDM400 is too expensive). I read there's a S100U which seems to be a
single FXS card, but I can't find a webshop selling it.

Thank you.

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RE: [Asterisk-Users] Polycom Instant Messaging

[Dean Collins]

The idea behind this is that web services will be able to 'message you'
through a number of devices, your handset being one of them.

For example outlook may want to send you a sip message via the exchange
server, via asterisk to your handset that your 2pm meeting is 10 minutes
away.

(I've got some ideas for cooler apps but this is an easy one to explain)

So Asterisk needs to move to be able to accept these messages.

Cheers,
Dean


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Chris Coulthurst
 Sent: Thursday, 19 May 2005 2:45 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] Polycom Instant Messaging
 
 If anyone has had success with IM with these phones in ANY
 configuration, I, as well as others Im sure would love to hear about
how
 its done.
 
 I envision messages being sent to the phone letting people know about
 pending appointments, etc.  I honestly don't care too much about
sending
 a message back to a user, just receiving one, but the more detail the
 better..
 
 Thanks again,
 
 Chris Coulthurst
 [EMAIL PROTECTED]
 
 
 |-Original Message-
 |From: [EMAIL PROTECTED] [mailto:asterisk-users-
 |[EMAIL PROTECTED] On Behalf Of Dean Collins
 |Sent: Wednesday, May 18, 2005 6:14 PM
 |To: Asterisk Users Mailing List - Non-Commercial Discussion; C F
 |Subject: RE: [Asterisk-Users] Polycom Instant Messaging
 |
 |There is a really good article in this months months Von magazine on
 |page 26 about why asterisk will need to adopt sip extensions for
 |Microsoft messenger.
 |
 |I'd post here but you don't accept images. But worth reading.
 |
 |Cheers,
 |Dean
 |
 |
 | -Original Message-
 | From: [EMAIL PROTECTED]
[mailto:asterisk-users-
 | [EMAIL PROTECTED] On Behalf Of Eric Wieling aka ManxPower
 | Sent: Wednesday, 18 May 2005 8:41 PM
 | To: C F; Asterisk Users Mailing List - Non-Commercial Discussion
 | Subject: Re: [Asterisk-Users] Polycom Instant Messaging
 |
 | Since the Polycom Instant Messaging features uses MSN Messenger, I
 | doubt it will work with Asterisk.
 |
 | C F wrote:
 |
 |  Asterisk can with the sendtext cmd which is available in
CVS-HEAD.
 | 
 |  On 5/18/05, Chris Coulthurst [EMAIL PROTECTED] wrote:
 | 
 | 
 | 
 | 
 | Can anyone explain the Polycom Text Messaging features built in
to
 |the
 | IP
 | 500/600?   Can Asterisk (or something else) talk to it?  I've
seen
 |vague
 | references to MSN Messenger, and somehow that's mentally
 |disturbing...
 |
 |
 | --
 | Always do right. This will gratify some people and astonish the
rest.
 | Mark Twain
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[Asterisk-Users] Asterisk real time extensions problem...

[Bharat M. Sarvan]

Hello everybody,


I have setup asterisk real time extensions and its working pretty well. But the
problem is when I am jumping between the contexts using the Goto statement in
the database. I am getting a error 



= Parsing '/etc/asterisk/sip_notify.conf': Found

 -- SIP Seeding peers from Astdb: 'ezzibpo4' at
[EMAIL PROTECTED]:5061 for 60

 -- Executing Goto(SIP/ezzibpo4-4636,
incoming-next,6069,1)

May 19 05:00:04 NOTICE[6420]: pbx.c:1688 pbx_extension_helper:
Cannot find extension '6069' in context 'incom'

May 19 05:00:04 WARNING[6420]: pbx.c:6256 ast_parseable_goto:
Priority 'incoming-next,



The structure of the extensions db is as given below



++---+---+--+-+--+

| id | context |
exten | priority |
app |
appdata
|

++---+---+--+-+--+

| 1 | incoming |
6069 | 1 |
Goto |
incoming-next,6069,1 |

| 2 | incoming |
6069 | 2 |
Hangup
|
|

| 3 | incoming-next | 6069
| 1 | DigitTimeout
|
10
|

| 4 | incoming-next | 6069
| 2 | ResponseTimeout |
30
|

| 5 | incoming-next | 6069
| 3 |
Background |
welcome
|





 The context incom
in the above error is the context defined for placing outgoing call in the
sip.conf file. I dont understand why is it looking for extension 6069 in
the incom context.



 The Goto statement in
the context incoming is getting executed without any probs, but the control is
not getting transferred to the context incoming-next upon
execution of the Goto statement. 

 

 Could anybody suggest me as
to where might the problem be and any way to get rid of this problem. Please do
reply.









Regards,

Bharat M. Sarvan








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[Asterisk-Users] Manager Port

[Chee Foong]

Hello all,

I am using flash operator panel, when i stop iptables everthing is fine, but
once iptables is started, the operator panel doesn't work anymore. Anyone
know how to set up the iptable in order for to op panel to work? I am using
tcp port 5038 for asterisk manager, and I have try open both tcp and udp
port 5038 in iptables but without success.

thanks

CCF


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Re: [Asterisk-Users] A good SIP receptionist phone

[Sven Fischer (support)]

On Monday 02 May 2005 20:10, Pedro wrote:
 What I did once was create an announcement that got played to the
 receptionist announcing who the call was for based on the number that
 was called.  This allowed the receptionist to know which greeting to
 recite.

Cool idea !


 On 5/2/05, Michael Welter [EMAIL PROTECTED] wrote:
  Chris Mason (Lists) wrote:
   The user name is the extension and the password is always the same. Not
   hard to configure.
 
  With the SNOM 220, you have five buttons/lamps that can be used as
  line appearances--these buttons can each register to a different SIP
  URL.
 
  Each sidecar has 20 buttons/lamps, and you may have up to three
  sidecars.  Using the hint priority in Asterisk, the buttons serve as
  extension busy lamps.  You can also use these buttons to transfer calls.
 
  I have an executive suites customer where each tenant is a separate
  business.  For an incoming call, the attendant needs to know which DID
  number is being called so she can answer with the proper greeting.
 
  I would like the sidecar buttons to be able to register to a SIP URL, so
  an incoming call would blink the tenants button, but that is not
  possible--I can only use the five buttons on the phone for that purpose,
  and there are more than five tenants.
 
  A suggestion was to alter the Called ID Name to the DID number.  This
  would work for the attendant, but the tenant would like to see the
  original Caller ID Name.
 
  I would rather not have to put a PC at the attendants position, but that
  is the way this is shaping up.  Does anyone have any suggestions?
 
  Thanks,
 
 
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---
See our FAQs at: http://www.snom.com/faq0.html?L=1
Whitepapers at:  http://www.snom.com/white_papers.html
---
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Sven Fischer fax +49 30 39833111tel +49 30 39833444
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RE: [Asterisk-Users] Static on TDM Zaptel FXO

[Rich Adamson]

  
  FWIW, I'm 7db from my central office and am using rxgain=5.0 and
  txgain=1.0 for now.
  
 
 Here in France I'm connected to ADSL using Free and they have a page
 available that you can get some info on your line, I'm 1355 metres from
 the central with 18dB. I'll play with rx/txgain.

Those numbers don't sound right. In the US, many of the telcos have 
the same type of static information available (at least within their
internal records), but the numbers are intended to be used by non-
technical telco types to determine whether a customer's location can
reasonably expect to support dsl.

If the 1355 meters is correct, the cable loss should be more like
about 4db (give or take a little based upon exactly what gauge of
copper is actually used to serve your location).

My guess is the 18db number is only there to suggest some sort of
upper 'limit' for adsl.

If that guess is correct, then I'd expect settings somewhere close
to rxgain=3 and txgain=0 might be a reasonable starting point for
zapata.conf parameters.

Rich


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RE: [Asterisk-Users] Static on TDM Zaptel FXO

[Dave Cotton]

On Thu, 2005-05-19 at 06:15 -0600, Rich Adamson wrote:

 Those numbers don't sound right. In the US, many of the telcos have 
 the same type of static information available (at least within their
 internal records), but the numbers are intended to be used by non-
 technical telco types to determine whether a customer's location can
 reasonably expect to support dsl.
 
 If the 1355 meters is correct, the cable loss should be more like
 about 4db (give or take a little based upon exactly what gauge of
 copper is actually used to serve your location).
 
 My guess is the 18db number is only there to suggest some sort of
 upper 'limit' for adsl.
 
 If that guess is correct, then I'd expect settings somewhere close
 to rxgain=3 and txgain=0 might be a reasonable starting point for
 zapata.conf parameters.
 

Interestingly I've found another site that I can get the data for any
number in France, the data is supposed to come from France Telecom's own
database. Even if you give an ISDN number it still reports the figures
but, of course, says the line is not suitable for ADSL.

-- 
Dave Cotton [EMAIL PROTECTED]

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Re: [Asterisk-Users] Newbie X100P question

[Rich Adamson]

 I just bought a X100P from digitnetworks.
 It is supposed to be a FXO card, but there are 2 rj-11 plug on the card.
 One is labelled phone and the other pstn. When i plug the pstn on
 the wall and the phone on my analog phone, everything (incoming and
 outgoing calls) works like before (without asterisk).
 AFAIU, i should have an FXS card in my box to be able to use my analog
 phone, so why does it work this way ?

The two rj-11 jacks are wired in parallel. There isn't any support
for fxs on either jack.

 Second question, what is the cheapest card to use one analog phone only
 (TDM400 is too expensive). I read there's a S100U which seems to be a
 single FXS card, but I can't find a webshop selling it.

My understanding is the S100U is old and discontinued; don't spend
any money on it.

The least expensive but reliable fxs approach is probably using an
external ata device. Something like the cisco ata186, sipura spa1000,
or the sipura spa3000 (which has both an fxo and fxs port on the
same box).

You should be aware the majority of the x100p cards (and compatibles)
use a chipset on the card that was designed to meet US telco impedence
standards. Trying to use those cards with other country standards
will likely end up with echo that negatively impacts all conversatons.


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[Asterisk-Users] asterisk-oh323 building problems

[FaberK]

Hello Guys,
first of all, I'm very new with asterisk.
I'm trying to set it up. I've already compiled and installed Asterisk-1.0.7
Now I'm trying with asterisk-oh323
I've already installed pwlib, oh323 and I've already set the variables.
Now, when I try to make asterisk-oh323 I receive this error messagge:
for x in wrapper asterisk-driver; do make -C $x all || exit 1 ; done
make[1]: Entering directory `/root/voip/asterisk/asterisk-oh323/wrapper'
g++ -Wall -mcpu=i586 -DP_LINUX -D_REENTRANT -DP_HAS_SEMAPHORES -DP_SSL
-DP_PTHREADS -DPBYTE_ORDER=PLITTLE_ENDIAN -DPHAS_TEMPLATES -O3
-DNDEBUG -I/usr/include -I/usr/include/crypto
-I/usr/lib/pwlib/include/ptlib/unix -I/usr/lib/pwlib/include
-I/usr/lib/openh323/include -I../asterisk-driver -g -c wrapper.cxx -o
wrapper.o
wrapper.cxx: In constructor
   `WrapH323Connection::WrapH323Connection(WrapH323EndPoint, unsigned int,
   int, int, short unsigned int)':
wrapper.cxx:563: `SetMaxAudioDelayJitter' undeclared (first use this function)
wrapper.cxx:563: (Each undeclared identifier is reported only once for each
   function it appears in.)
wrapper.cxx: In function `call_ret_val_t h323_clear_call(const char*)':
wrapper.cxx:1230: warning: unused variable `ClearCallThread*clearCallThread'
make[1]: *** [wrapper.o] Error 1
make[1]: Leaving directory `/root/voip/asterisk/asterisk-oh323/wrapper'
make: *** [subdirs_all] Error 1


What's wrong?

Thanks

-- 
.:FaberK:.
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[Asterisk-Users] asterisk-oh323 build problems

[FaberK]

Hello Guys,
first of all, I'm very new with asterisk.
I'm trying to set it up. I've already compiled and installed Asterisk-1.0.7
Now I'm trying with asterisk-oh323
I've already installed pwlib, oh323 and I've already set the variables.
Now, when I try to make asterisk-oh323 I receive this error messagge:
for x in wrapper asterisk-driver; do make -C $x all || exit 1 ; done
make[1]: Entering directory `/root/voip/asterisk/asterisk-oh323/wrapper'
g++ -Wall -mcpu=i586 -DP_LINUX -D_REENTRANT -DP_HAS_SEMAPHORES -DP_SSL
-DP_PTHREADS -DPBYTE_ORDER=PLITTLE_ENDIAN -DPHAS_TEMPLATES -O3
-DNDEBUG -I/usr/include -I/usr/include/crypto
-I/usr/lib/pwlib/include/ptlib/unix -I/usr/lib/pwlib/include
-I/usr/lib/openh323/include -I../asterisk-driver -g -c wrapper.cxx -o
wrapper.o
wrapper.cxx: In constructor
   `WrapH323Connection::WrapH323Connection(WrapH323EndPoint, unsigned int,
   int, int, short unsigned int)':
wrapper.cxx:563: `SetMaxAudioDelayJitter' undeclared (first use this function)
wrapper.cxx:563: (Each undeclared identifier is reported only once for each
   function it appears in.)
wrapper.cxx: In function `call_ret_val_t h323_clear_call(const char*)':
wrapper.cxx:1230: warning: unused variable `ClearCallThread*clearCallThread'
make[1]: *** [wrapper.o] Error 1
make[1]: Leaving directory `/root/voip/asterisk/asterisk-oh323/wrapper'
make: *** [subdirs_all] Error 1


What's wrong?

Thanks

-- 
.:FaberK:.
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Re: [Asterisk-Users] Newbie X100P question

[Adrian Urquhart]

On Thu, 19 May 2005, Rich Adamson wrote:
[snip]
You should be aware the majority of the x100p cards (and compatibles)
use a chipset on the card that was designed to meet US telco impedence
standards. Trying to use those cards with other country standards
will likely end up with echo that negatively impacts all conversatons.

I can confirm this - I've just bought one of these cards cheap via eBay 
(I'm in the UK - 10.00 UKP) and although it works echo is a major 
problem with it. It's usable but I intend to replace it a.s.a.p.

-Adrian
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RE: [Asterisk-Users] Static on TDM Zaptel FXO

[Rich Adamson]

  Those numbers don't sound right. In the US, many of the telcos have 
  the same type of static information available (at least within their
  internal records), but the numbers are intended to be used by non-
  technical telco types to determine whether a customer's location can
  reasonably expect to support dsl.
  
  If the 1355 meters is correct, the cable loss should be more like
  about 4db (give or take a little based upon exactly what gauge of
  copper is actually used to serve your location).
  
  My guess is the 18db number is only there to suggest some sort of
  upper 'limit' for adsl.
  
  If that guess is correct, then I'd expect settings somewhere close
  to rxgain=3 and txgain=0 might be a reasonable starting point for
  zapata.conf parameters.
  
 
 Interestingly I've found another site that I can get the data for any
 number in France, the data is supposed to come from France Telecom's own
 database. Even if you give an ISDN number it still reports the figures
 but, of course, says the line is not suitable for ADSL.

In very very general terms, adsl is limited to sites that are within
about 18,000 feet of the central office. (The real limit is expressed
in terms of loss, but the limit was restated in terms of something the
average non-technical telco employees can understand -- distance.)
Adsl would never function on a cable pair that had 18db of loss, so
something's wrong with that reported number, or, it was meant to
communicate something different then how we're reading it.

If asterisk (zaptel) would behave like commercial pbx's (and associated
hardware), a real pbx engineer would:
 - measure the transmission loss from the customer's site to the
   central office milliwatt generator (eg, -7db loss)
 - then set the rxgain and txgain parameters to something slightly
   less then the measured loss (eg, rxgain=5, txgain=5). Its not
   uncommon to see those settings around 2db below the measured
   loss, but that varys by each pbx/telco engineering staff.

However, the echo canceller in asterisk is not very good, and that
forces us to use gain values substantially lower (causing complaints 
about low volume in many cases).

Part of the echo canceller problems seem to be related to highly
variable general-purpose PC hardware (amoung other things). That
includes pci/interrupt latency issues, OS overhead, etc. As a 
result, gain settings that work for one asterisk system may not
even be close for another system, and generally will be very
different from those used in commercial pbxs. (That is exactly
why T1/E1 interfaces to asterisk are preferred over analog
interfaces.)

Those asterisk systems further away from the central office typically
have more issues with echo and audio levels then do those systems 
closer to the central office (distance measured in terms of pstn 
cable loss). That's also one of the reasons why colocated asterisk
boxes don't have as many echo  audio level issues.



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RE: [Asterisk-Users] OT: carrying a router, firewall, switch, server, some phones with me on flight to Europe

[Shaoul Jacobson - TELLINK]

Hi,

 There is of course one obvious issue, that of powering your equipment
 at 220V/50~ and the plug convertors if your are lucky enough to have
 power supplies that do 100-250v.

and the plug format is different 
(UK, germany+NL, France+Belgium, Italy, ...)
there are some 'universal' plug changers

good trip

Shaoul Jacobson
Senior VoIP Consultant
Tellink
Tel :   +32 3 201 96 36
Fax :   +32 3 227 09 81
e-mail  [EMAIL PROTECTED]


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[Asterisk-Users] asterisk

[Allan Regenbaum]

I have been trying for days to get an outbound connection to broadvoice 
with no luck ..details below ... I have scoured all postings and seem to 
get similar responses but none of these seem to help... any help is 
appreciated ..
my [EMAIL PROTECTED] box is sitting as 192.168.1.106 behind a linksys router 
that feeds to comcast as the provider.

trying to make outbound calls from a analog phone extension on a digium 
baord to broadvoice ..
system works fine analog phone  to analog trunk , but cant get calls out 
from analog phone or softphone to broadvoice .

asterisk log throws
  -- Executing Dial(Zap/1-1, SIP/[EMAIL PROTECTED]|30) in new
 stack
-- Called [EMAIL PROTECTED]
-- Got SIP response 604 Does not exist anywhere back from 147.135.0.128
  == No one is available to answer at this time
-- Executing Congestion(Zap/1-1, ) in new stack
  == Spawn extension (from-internal, 17705229625, 2) exited non-zero on 
'Zap/1-1

sip .conf is
[general]
port = 5060   ; Port to bind to (SIP is 5060)
bindaddr = 192.168.1.106; Address to bind to (all addresses on machine)
disallow=all
allow=gsm
allow=ulaw
allow=alaw
;context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
context = from-broadvoice
externip=69.??.??.??
localnet=192.168.1.0/255.255.255.0

 sip_additional.conf shows
register=561???:91?:@sip.broadvoice.com/201
*** i have tried various permutations of this
[bv]
username=5618282155
user=phone
type=peer
secret=myPassword
nat=yes
insecure=very
host=sip.broadvoice.com
fromuser=561??
fromdomain=sip.broadvoice.com
dtmfmode=inband
dtmf=inband
context=from-broadvoice
canreinvite=no
authname=561??
[sip.broadvoice.com]
username=561
user=561
type=user
secret=91???
nat=yes
insecure=very
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
dtmfmode=inband
dtmf=inband
context=from-broadvoice
canreinvite=no

also , per postings on the boards ..i pasted this to extensions.conf 
..seems that amp had not created an entry for this

exten = _1NXXNXX, 1, dial(SIP/[EMAIL PROTECTED],30)
exten = _1NXXNXX, 2, congestion()
exten = _1NXXNXX, 102, busy()
==
outbound routing ...
i have prefix 1 directing to the BV trunk
all (other than general section in sip.conf and the extensions.conf) were 
setup using amp ..seems amp does not place the entries in extension.conf ...

===
trunks in amp is a follows
sip trunk...
outbound caller is is broadvoice
max channels is blank
no dial rules
no dial prefix
outgoing settings
trunk name is bv
peer details are
authname=561???
canreinvite=no
context=from-broadvoice
dtmf=inband
dtmfmode=inband
fromdomain=sip.broadvoice.com
fromuser=561???
host=sip.broadvoice.com
insecure=very
nat=yes
secret=91??
type=peer
user=phone
username=561???
incoming settings
user context sip.broadvoice.com
user details
canreinvite=no
context=from-broadvoice
dtmf=inband
dtmfmode=inband
fromdomain=sip.broadvoice.com
host=sip.broadvoice.com
insecure=very
nat=yes
secret=91?
type=user
user=561?
username=561
register string ...
561??:91?:@sip.broadvoice.com/201
=
fyi ...
this is an [EMAIL PROTECTED] setup 
my bv number is shown as 561??
my bv fancy password is shown as 91??
i have a outbound rule that says numbers with prefix 1 ..go to sip/bv trunk
any help is MUCH appreciated
 

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[Asterisk-Users] retail unit for cards

[Iqbal]

Hi
Does anyone know of a retail outlet in the UK where you maybe able to 
purchase cards for asterisk.

Iqbal
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[Asterisk-Users] Deleting Monitor Files After 2 Months

[Steve Totaro]

Does anyone knowthe best wayto automate 
the deletion of monitor files after they age two months?

Thanks,
Steve 
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Re: [Asterisk-Users] Deleting Monitor Files After 2 Months

[Gavin Hamill]

On Thursday 19 May 2005 13:51, Steve Totaro wrote:
 Does anyone know the best way to automate the deletion of monitor files
 after they age two months?

How about ... 

$ find /path/to/files -ctime +60 -exec rm {}\; 

Cheers,
Gavin.
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[Asterisk-Users] Random Blip

[Mike Spradling]

Im
using Asterisk 1.0.7 and making calls from the Eyebeam SIP softphone through
asterisk, an IAX2 connection to voicepulse and out to the PSTN 



Eyebeam
- Asterisk 1.0.7 - (Sonicwall) - VoicePulse via IAX2 - PSTN



On
any call I make after a few minutes on the phone, I get a sort of blip
noise followed by a click. The call continues and the other person just
hears a click. In an hour call it probably happens about 4 or 5 times but
not at predictable intervals.



Any
ideas where to start to figure this out? Ill be happy to pick
through logs if I know which one. 



Mike






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[Asterisk-Users] MusicOnHold probelms

[chawki hammoud]

This is my second attempt trying to get help and I am
hoping someone can. When the musiconhold extension is
matched, Asterisk attempts to execute musiconhold and
stops right away, this is what I gets:

 Executing MusicOnHold(OSS/dsp, ) in new stack
-- Started music on hold, class 'default', on
OSS/dsp
-- Stopped music on hold on OSS/dsp

Is there a file that musiconhold try to play and can't
find. Please help withy any suggestions.




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Stay in touch with email, IM, photo sharing and more. Check it out! 
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Re: [Asterisk-Users] connecting a sipura sip device to asterisk beforedialing any digits

[Jon Gabrielson]

Thanks, I must have looked at the list of available
commands a dozen times, I knew that there almost 
had to be one, but that one kept hiding from me.


Thanks,


Jon.

On Thursday 19 May 2005 02:12 am, Dave Cotton wrote:
 On Wed, 2005-05-18 at 16:02 -0500, Jon Gabrielson wrote:
  Thanks, that works great.
  It transfers directly to ext 100.
  Now how do I tell asterisk to give ext 100 a dialtone?
  I can do Dial(Zap/1), but that only gives an external
  dialtone, is there a way to get asterisk to give an
  internal dialtone?

 Maybe using the logic of DISA would help.
 Just an idea.
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Re: [Asterisk-Users] retail unit for cards

[David John Walsh]

assuming you mean digium zap style cards, yes there are several.

I don't want to directly quote you any as I have a relationship with a
number of them, however googling for digium wildcard brings up
several

David

On 5/19/05, Iqbal [EMAIL PROTECTED] wrote:
 Hi
 
 Does anyone know of a retail outlet in the UK where you maybe able to
 purchase cards for asterisk.
 
 Iqbal
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RE: [Asterisk-Users] OT: carrying a router, firewall, switch, ser ver, some phones with me on flight to Europe

[Huddleston, Robert]

Well here's a suggestion - a little crazy - but works... Most equipment is
taking the 120vac and converting it into DC voltage. So why not just feed it
DC voltage directly???
We had a situation where our field techs needed to test dsl circuits and
voip ata from the demarcation point outside a house or business. A UPS might
have worked - but the down conversion of 12v dc battery in ups up to 120vac
to power the plugs on the ata and modem - just to convert back down to 12
and 5.. Make sense... Common electronics theory tells you that there is
waste in step-up/step-down === heat...
So maybe that's an idea... I took a UPS battery and a small project case
from common electronics retail store... Then bought me a very small voltage
regulator and soldered it in the case I was able to split off 12v and 5v
from the ups battery and run for days...
Sounds like weird science - but it works!!

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shaoul
Jacobson - TELLINK
Sent: Thursday, May 19, 2005 8:26 AM
To: Wilson Pickett; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] OT: carrying a router, firewall, switch,
server, some phones with me on flight to Europe

Hi,

 There is of course one obvious issue, that of powering your equipment
 at 220V/50~ and the plug convertors if your are lucky enough to have
 power supplies that do 100-250v.

and the plug format is different 
(UK, germany+NL, France+Belgium, Italy, ...)
there are some 'universal' plug changers

good trip

Shaoul Jacobson
Senior VoIP Consultant
Tellink
Tel :   +32 3 201 96 36
Fax :   +32 3 227 09 81
e-mail  [EMAIL PROTECTED]


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Re: [Asterisk-Users] Most stable HEAD

[NVC List Manager]

On Wednesday 18 May 2005 22:43, NVC List Manager wrote:
 Hi,

 I'd like to get a census of what you consider the most stable HEAD.

 Thanks!

(Of course with this I mean as of what date.)

-- 

NVC List Manager
(For external lists)
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Re: [Asterisk-Users] Deleting Monitor Files After 2 Months

[Mike Dent]

On 5/19/05, Steve Totaro [EMAIL PROTECTED] wrote:
  
 Does anyone know the best way to automate the deletion of monitor files
 after they age two months? 
   
 Thanks, 
 Steve  
 ___

Something like:

find /files/to/check/ -mtime 60 -exec rm {} \;

put this in a crontab entry maybe and run each day.


Mike
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Re: [Asterisk-Users] Issues with Polycom 1.5.2 ProFTPd

[[EMAIL PROTECTED]]

FYI, for anybody running Proftpd with Polycom's, here are a couple of 
things that I found that seem to help.

1.  Ident seems to be on by default (at least on Trustix it is). 
Turning it off if the phones are behind a firewall decreases the login 
time substantially.  Here is what I have:

Global
  DenyFilter \*.*/
IdentLookups off
/Global
2.  Increasing the logging on Proftpd can give you a good idea of 
whether there are any errors pulling down files.  I use the following in 
proftpd.confg:

ExtendedLog /var/log/activity-ftp.log read,write
I can see all files that the phones try to download and any errors.  We 
had a permission error that was causing LOTS of retries.  Once I enabled 
the logging, I was able to see what the phones were doing and then fix 
it.  Our logs dropped to virtually nothing.

Peder
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[Asterisk-Users] Public vs. Private Network

[David Sampson]

Hello 



I am looking at connecting 7  10 locations together
using Asterisk and possibly some VoIP gateway appliances. I need to
insure best voice quality as these trunks will be used primarily for customer
calls. I am considering implementing a full T1 frame relay circuit to
each location which can be done for a reasonable cost. DSL and Cable are
currently at each location and setup for automatic failover. Should I remove
one of my public connections and replace it with a private circuit for best
quality?

Thank you,


Dave








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Re: [Asterisk-Users] Random Blip

[William Suffill]

What codec are you using to asterisk and what codec to VPC? Also does
this occur if you test the service with another ITSP
(nufone/voipjet/teliax)
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[Asterisk-Users] Do Both! :) Re: Telecom SIP termination vs. DS3

[M O]

Message: 16
Date: Thu, 19 May 2005 00:16:34 -0600
Michael,

Do both!

As for Sip Termination:
---

Contact Kristi Eggers @ Txlink.net for month to month
Originating/Termination VoIP Toll Free or Local USA 
DID #s.  Yes they do both Sip and IAX.  You must have
seperate accounts for either Sip or IAX and fund your
account with a minimum of $100.  This is what I did.

Once I get through testing out my Asterisk/Areski
Calling Card box with my newly acquired DID #s from
TxLink.net, and if testing is successful for remainder
of the month of May, I intend on purchasing, within
the month of June, 1 Digium DS3 card to start.  

Actually, I am trying to budget on adding 1 Digium DS3
card either every 30 - 60 days throughout the
remainder of this year.  

Adding Calling Card sales rep's = placing a
Elephant/HOG in front of a Highbandwidth pipe!!!

WindyCitySDR's situation is such that every 1 Calling 
Card Sales rep. I add = 28 T-1s.  68 concurrent
callers is 1Mbps CONISISTANT BANDWIDTH.  And you dont
want to know what a startup pays per 1Mbps! ALOT OF
$!!

A Digium DS3 card saves the lives of Calling Card
infrastructure providers such as myself. (THANK
GOD!!!)

So, in my humble opinion, doing both the sip
termination NOW, to get that $$$ together for the
Digium DS3 PCI Channelized Voice card is EXCELLENT!
:):) 

I might, at best, add a SER server to feed my
Asterisk,
but beyond that, the above is E X A C T L Y what I am
doing! :)

Sincerely,

SoftwareRadioGuy


From: Michael Welter [EMAIL PROTECTED]
Subject: [Asterisk-Users] Telecom SIP termination vs.
DS3
To: Asterisk Users Mailing List - Non-Commercial
Discussion
asterisk-users@lists.digium.com,  Commercial and
Business-Oriented
Asterisk Discussion asterisk-biz@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1;
format=flowed

(Cross posting on purpose)

What is the common wisdom on the list...  find a
telco that offers SIP termination or wait for
Digium's DS3 card?

Who are the telcos that offer SIP termination?

Thanks,

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RE: [Asterisk-Users] OT: carrying a router, firewall, switch, ser ver, some phones with me on flight to Europe

[David Brodbeck]

 -Original Message-
 From: Giles Coochey [mailto:[EMAIL PROTECTED]

 * While most PC PSUs these days are 100-240V, and they seem to have no
 problems operating both in Europe and the US. UPSs are different,
 however, they are almost universally either 110V or 240V only, and
 there's not even a switch to switch between the two voltages. APC will
 sell you either a US or a EU version, and usually only if they're
 shipping to the destination.
 
 * Just a small UPS will probably do your baggage allowance in as well

Yes.  The company I work for occasionally ships configured PCs (being used
as industrial controllers) to European countries.  The PCs themselves are no
problem, but if a UPS is required we always have them buy it locally.
They're hard to get for European power standards in the U.S. (I've tried),
and they're heavy and expensive to ship.

Don't put anything in your checked bags you can't afford to lose.  One of my
friends is a travelling technician and regularly checks a bag of tools.  It
almost always arrives with something missing.  He's lost three Leatherman
tools.  The TSA won't allow you to lock bags anymore, and the TSA inspectors
and/or baggage handlers apparently have sticky fingers.  If it's expensive
to replace and too big to put in a carry on, consider shipping it to your
destination instead of putting it in your luggage.
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Re: [Asterisk-Users] asterisk

[John Millican]

On Thursday May 19 2005 8:38 am, Allan Regenbaum wrote:
 I have been trying for days to get an outbound connection to broadvoice
 with no luck ..details below ... I have scoured all postings and seem to
 get similar responses but none of these seem to help... any help is
 appreciated ..
 my [EMAIL PROTECTED] box is sitting as 192.168.1.106 behind a linksys router
 that feeds to comcast as the provider.

 trying to make outbound calls from a analog phone extension on a digium
 baord to broadvoice ..
 system works fine analog phone  to analog trunk , but cant get calls out
 from analog phone or softphone to broadvoice .

 asterisk log throws

-- Executing Dial(Zap/1-1, SIP/[EMAIL PROTECTED]|30) in
 new stack
  -- Called [EMAIL PROTECTED]
  -- Got SIP response 604 Does not exist anywhere back from
 147.135.0.128 == No one is available to answer at this time
  -- Executing Congestion(Zap/1-1, ) in new stack
== Spawn extension (from-internal, 17705229625, 2) exited non-zero on
 'Zap/1-1


 sip .conf is

 [general]
 port = 5060   ; Port to bind to (SIP is 5060)
 bindaddr = 192.168.1.106; Address to bind to (all addresses on machine)
 disallow=all
 allow=gsm
 allow=ulaw
 allow=alaw
 ;context = from-sip-external ; Send unknown SIP callers to this context
 callerid = Unknown
 context = from-broadvoice
 externip=69.??.??.??
 localnet=192.168.1.0/255.255.255.0

 

   sip_additional.conf shows

 register=561???:91?:@sip.broadvoice.com/201

 *** i have tried various permutations of this

 [bv]
 username=5618282155
 user=phone
 type=peer
 secret=myPassword
 nat=yes
 insecure=very
 host=sip.broadvoice.com
 fromuser=561??
 fromdomain=sip.broadvoice.com
 dtmfmode=inband
 dtmf=inband
 context=from-broadvoice
 canreinvite=no
 authname=561??

 [sip.broadvoice.com]
 username=561
 user=561
 type=user
 secret=91???
 nat=yes
 insecure=very
 host=sip.broadvoice.com
 fromdomain=sip.broadvoice.com
 dtmfmode=inband
 dtmf=inband
 context=from-broadvoice
 canreinvite=no

 
 also , per postings on the boards ..i pasted this to extensions.conf
 ..seems that amp had not created an entry for this

 exten = _1NXXNXX, 1, dial(SIP/[EMAIL PROTECTED],30)
 exten = _1NXXNXX, 2, congestion()
 exten = _1NXXNXX, 102, busy()

 ==
 outbound routing ...
 i have prefix 1 directing to the BV trunk

 all (other than general section in sip.conf and the extensions.conf) were
 setup using amp ..seems amp does not place the entries in extension.conf
 ...

 ===
 trunks in amp is a follows

 sip trunk...
 outbound caller is is broadvoice
 max channels is blank
 no dial rules
 no dial prefix

 outgoing settings
 trunk name is bv
 peer details are

 authname=561???
 canreinvite=no
 context=from-broadvoice
 dtmf=inband
 dtmfmode=inband
 fromdomain=sip.broadvoice.com
 fromuser=561???
 host=sip.broadvoice.com
 insecure=very
 nat=yes
 secret=91??
 type=peer
 user=phone
 username=561???

 incoming settings
 user context sip.broadvoice.com

 user details
 canreinvite=no
 context=from-broadvoice
 dtmf=inband
 dtmfmode=inband
 fromdomain=sip.broadvoice.com
 host=sip.broadvoice.com
 insecure=very
 nat=yes
 secret=91?
 type=user
 user=561?
 username=561

 register string ...
 561??:91?:@sip.broadvoice.com/201
 =
 fyi ...

 this is an [EMAIL PROTECTED] setup 
 my bv number is shown as 561??
 my bv fancy password is shown as 91??
 i have a outbound rule that says numbers with prefix 1 ..go to sip/bv trunk
try setting a /etc/hosts entry for one of their proxy servers( I use 
147.135.12.128, 147.135.0.128 is not good) if you ping all their proxies and 
set the hosts entry to the fastest one this will help.  
ALSO you should know that there are MAJOR problems with broadvoice.  I have 
had an account with them for 3 months or so and at first all worked great, 
then the last month or so it has been very bad!   As of this morning i am 
getting no sound in either direction.  my asterisk box is getting and 
answering the call, playing the voice prompts that it should but I can not 
here them and it does not receive any DTMF.
John M
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Re: [Asterisk-Users] Deleting Monitor Files After 2 Months

[Matthew Boehm]

Gavin Hamill wrote:
 On Thursday 19 May 2005 13:51, Steve Totaro wrote:
 Does anyone know the best way to automate the deletion of monitor
 files after they age two months?

 How about ...

 $ find /path/to/files -ctime +60 -exec rm {}\;

 Cheers,
 Gavin.

Nice Gavin. I would further turn that into a shell script and pop it into
cron to run nightly.

-Matthew

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[Asterisk-Users] chan_capi patch eicon

[Altus Snyman]

Good day all
Im trying a eicon 4bri card
On fedora core 1
I installed the rpm,lsmod says the driver is working
I then installed asterisk 1.0.7
I then download chan_capi 0.3.5
But now it says I should patch it for asterisk
So I got the patch..fixed it
And did a make
and it gives a lot of  syntax errors
Please Help
Thanks
Altus

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Re: [Asterisk-Users] SIP Phone Recommendations?

[Michael Graves]

On Wed, 18 May 2005 22:29:40 -0500, Kristian Kielhofner wrote:

Ariel,

   It's probably not a good idea to reccomend the IP 500/300 anymore. 
They are being phased out by Polycom because they (and the IP 300) only 
have 2mb of flash, and Polycom is looking to standardize on 4mb for 
their firmware (which the IP 600 has had since day one).

   If you are going to buy a Polycom now, get an IP 600, or, wait for the 
301's or 501's.  Don't say I didn't warn you!

Good advice!. BTW, I LOVE my IP600's. 

I also kinda like the Zultys 4x4/4x5.The hardware and software is good
but their support arrangement is terrible. They provide no end user
support at all. Period. They rely upon their dealers to provide all
support, but then they're ok with signing up dealers that know nothing
about the products.

Michael

--
Michael Graves   [EMAIL PROTECTED]
Sr. Product Specialist  www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]

o713-861-4005
o800-905-6412
c713-201-1262



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[Asterisk-Users] tdm400p fxo not working

[Matt Scott]

Dear all.

I have a tdm400p with an FXO module in slot 4 and 
an FXS module in slot 1.
I have not configured the FXS port in an attempt to 
keep things simple.
The problem is that when I call the POTS number 
(assigned by phone company) asterisk is seeing the call but then not doing 
anything with it.

The verbose output from asterisk is as 
follows:
--
*CLI  == Starting post polarity CID 
detection on channel 4 -- Starting simple switch on 
'Zap/4-1'May 19 15:10:29 NOTICE[30934]: chan_zap.c:5542 ss_thread: Got event 
17 (Polarity Reversal)...May 19 15:10:31 WARNING[30934]: chan_zap.c:5582 
ss_thread: CID timed out waiting for ring. Exiting simple 
switch -- Hungup 'Zap/4-1'
---
From the caller end it just rings 
constantly.
I have the following configurations:

zaptel.conf
fxsks=4loadzone=ukdefaultzone=uk

zapata.conf
; Zapata telephony interface; Configuration 
file;[channels]language=ukgroup=1context=from-pstnsignalling=fxs_kschannel 
= 4

extensions.conf
[from-pstn]exten = 
s,1,Dial(SIP/1001,20)exten = s,2,Hangup

The SIP elements of my system are working well, I 
just need to get this incoming call on a POTS line working.
I have tried to keep things as simple as 
possible.

Does anyone know why my call is not being handed to 
my sip phone?
What is CID timed out waiting for ring? Is this 
something to do with caller ID?
I have tried it with a 'wait' command in the 
extensions.conf as well but no joy.

Kind regards
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Re: [Asterisk-Users] MusicOnHold probelms

[Christian Wengel]

Hi!
Could you post your musiconhold.conf and modules.conf, please?
MfG Christian
chawki hammoud wrote:
This is my second attempt trying to get help and I am
hoping someone can. When the musiconhold extension is
matched, Asterisk attempts to execute musiconhold and
stops right away, this is what I gets:
Executing MusicOnHold(OSS/dsp, ) in new stack
   -- Started music on hold, class 'default', on
OSS/dsp
   -- Stopped music on hold on OSS/dsp
Is there a file that musiconhold try to play and can't
find. Please help withy any suggestions.

		
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Re: [Asterisk-Users] Forwarding To Cell Phones with Asterrisk PBX

[Moody]

Colin, 

I'm not sure this helps the problem, if you want to try DIALs the
caller is still left hanging during each 30 or 40 second period.

As for timing the call, be careful of voicemail on busy as I think
you'll find that most cell phone voicemail will also answer if the
line is busy/off/not-in-service.

I have been using the 'press 1 to accept' macro and find it works well
but doesn't address the time issue for multiple destinations. My best
solution to that is a fixed Please hold while we connect you message
(and not using 'r' in the DIAL cmd).

I have also tried to use '' in the DIAL parameter without success as
it does not wait for the macro to complete but rather simply an answer
so Asterisk drops all other outgoing attempts before the macro is
complete.

Am I missing a trick to dial multiple destination over multiple
channels (SIP/IAX) at once and not merge the channels until after the
DIAL macro is complete?

Jonathon



On 5/16/05, Colin Anderson [EMAIL PROTECTED] wrote:
   
 exten = 12345,1,Dial(SIP/12345,40) 'Dial extension 12345 for 40 seconds. If
 no one picks up then... 
 exten = 12345,2,Dial(ZAP/g0/5551212,25) 'Forward the call out to the user's
 cell. Once they pick up, a native bridge of ZAP channels occur and Asterisk
 is out 'of the media stream 
 exten = 12345,3,(anything else that happens later, like go to voicemail,
 etc) 
   
 It's important to time how long it takes for the remote user's cellphone to
 pick up for voicemail. If the user's voicemail on the cell kicks in after,
 say 4 rings, time your second Dial() command to be just short of that,
 otherwise the remote caller will get the cell phone's voicemail, which is
 probably not the desired behavior. In my case, I set it for 25 seconds, as
 our cells' voicemail kicks in after 30 seconds. If there's no call pickup on
 the cell, call processing continues to the next priority, which is voicemail
 or IVR depending on what number they called. 

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RE: [Asterisk-Users] Spanish TTS

[Anton Krall]

May be worth asking them.. 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Ben Hencke
|Sent: Jueves, 19 de Mayo de 2005 01:25 a.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Spanish TTS
|
|Cepstral. They will tune it for $150/hr too. Depending on the 
|vocabulary of your app, this could still be cheaper than most 
|commercial TTS engines.
|They have 2 new English voices that sound very nice, maybe 
|they will upgrade their other languages too.
|- Ben
|
|
|On 5/18/05, Anton Krall [EMAIL PROTECTED] wrote:
| Anybody doing Spanish TTS? what are you using? Festival, Cepstral?
| 
| I just tried Cepstral and their male spanish voice is not bad but 
| still sounds a bit robotic.. anything better? although their 
|licensing 
| price is quite nice :)
| 
| 
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Re: [Asterisk-Users] chan_capi patch eicon

[Armin Schindler]

On Thu, 19 May 2005, Altus Snyman wrote:
 Good day all
 Im trying a eicon 4bri card
 On fedora core 1
 I installed the rpm,lsmod says the driver is working
 I then installed asterisk 1.0.7
 I then download chan_capi 0.3.5
 But now it says I should patch it for asterisk
 So I got the patch..fixed it

Patch? What patch?

Armin

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[Asterisk-Users] New IAXy from Digium

[Robert Webb]

I was just browsing Digium's web site and noticed they are 
taking orders for the new IAXy. Has anyone purchased and 
tested one of these yet?? I have thought about buying one 
for testing, but want to make sure it isn't going to be a 
flop like the last one.

Robert
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Re: [Asterisk-Users] Do Both! :) Re: Telecom SIP termination vs. DS3

[BJ Weschke]

 Did I miss pricing/availability announcements from Digium on that DS3
card somewhere? I wasn't aware they were going to be GA in less than 3
weeks from now.

On 5/19/05, M O [EMAIL PROTECTED] wrote:
 Message: 16
 Date: Thu, 19 May 2005 00:16:34 -0600
 Michael,
 
 Do both!
 
 As for Sip Termination:
 ---
 
 Contact Kristi Eggers @ Txlink.net for month to month
 Originating/Termination VoIP Toll Free or Local USA
 DID #s.  Yes they do both Sip and IAX.  You must have
 seperate accounts for either Sip or IAX and fund your
 account with a minimum of $100.  This is what I did.
 
 Once I get through testing out my Asterisk/Areski
 Calling Card box with my newly acquired DID #s from
 TxLink.net, and if testing is successful for remainder
 of the month of May, I intend on purchasing, within
 the month of June, 1 Digium DS3 card to start.
 
 Actually, I am trying to budget on adding 1 Digium DS3
 card either every 30 - 60 days throughout the
 remainder of this year.
 
 Adding Calling Card sales rep's = placing a
 Elephant/HOG in front of a Highbandwidth pipe!!!
 
 WindyCitySDR's situation is such that every 1 Calling
 Card Sales rep. I add = 28 T-1s.  68 concurrent
 callers is 1Mbps CONISISTANT BANDWIDTH.  And you dont
 want to know what a startup pays per 1Mbps! ALOT OF
 $!!
 
 A Digium DS3 card saves the lives of Calling Card
 infrastructure providers such as myself. (THANK
 GOD!!!)
 
 So, in my humble opinion, doing both the sip
 termination NOW, to get that $$$ together for the
 Digium DS3 PCI Channelized Voice card is EXCELLENT!
 :):)
 
 I might, at best, add a SER server to feed my
 Asterisk,
 but beyond that, the above is E X A C T L Y what I am
 doing! :)
 
 Sincerely,
 
 SoftwareRadioGuy
 
 
 From: Michael Welter [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Telecom SIP termination vs.
 DS3
 To: Asterisk Users Mailing List - Non-Commercial
 Discussion
asterisk-users@lists.digium.com,  Commercial and
 Business-Oriented
Asterisk Discussion asterisk-biz@lists.digium.com
 Message-ID: [EMAIL PROTECTED]
 Content-Type: text/plain; charset=ISO-8859-1;
 format=flowed
 
 (Cross posting on purpose)
 
 What is the common wisdom on the list...  find a
 telco that offers SIP termination or wait for
 Digium's DS3 card?
 
 Who are the telcos that offer SIP termination?
 
 Thanks,
 
 __
 Do You Yahoo!?
 Tired of spam?  Yahoo! Mail has the best spam protection around
 http://mail.yahoo.com
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RE: [Asterisk-Users] SIP Phone Recommendations?

[Charlie Watts]

Kristian Kielhofner wrote:
   It's probably not a good idea to reccomend the IP 500/300
anymore.
 They are being phased out by Polycom because they (and the IP 300)
 only have 2mb of flash, and Polycom is looking to standardize on 4mb
 for their firmware (which the IP 600 has had since day one).  
 
   If you are going to buy a Polycom now, get an IP 600, or, wait
for
 the 301's or 501's.  Don't say I didn't warn you! 

It looks like the only features the 300 and 500 currently don't support,
but the 301, 501, and 600 *do* support, is HTTPS/FTPS encrypted
provisioning. Am I wrong about this?

Obviously this may change in the future, but the 300 and 500 haven't
suddenly become any *less* capable than they were last week. They may
just not get *more* capable in the future.
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Re: [Asterisk-Users] asterisk-oh323 build problems

[Michael Manousos]

What versions of OpenH323/Pwlib/asterisk-oh323 are you trying
to install?
Michael.
FaberK wrote:
Hello Guys,
first of all, I'm very new with asterisk.
I'm trying to set it up. I've already compiled and installed Asterisk-1.0.7
Now I'm trying with asterisk-oh323
I've already installed pwlib, oh323 and I've already set the variables.
Now, when I try to make asterisk-oh323 I receive this error messagge:
for x in wrapper asterisk-driver; do make -C $x all || exit 1 ; done
make[1]: Entering directory `/root/voip/asterisk/asterisk-oh323/wrapper'
g++ -Wall -mcpu=i586 -DP_LINUX -D_REENTRANT -DP_HAS_SEMAPHORES -DP_SSL
-DP_PTHREADS -DPBYTE_ORDER=PLITTLE_ENDIAN -DPHAS_TEMPLATES -O3
-DNDEBUG -I/usr/include -I/usr/include/crypto
-I/usr/lib/pwlib/include/ptlib/unix -I/usr/lib/pwlib/include
-I/usr/lib/openh323/include -I../asterisk-driver -g -c wrapper.cxx -o
wrapper.o
wrapper.cxx: In constructor
   `WrapH323Connection::WrapH323Connection(WrapH323EndPoint, unsigned int,
   int, int, short unsigned int)':
wrapper.cxx:563: `SetMaxAudioDelayJitter' undeclared (first use this function)
wrapper.cxx:563: (Each undeclared identifier is reported only once for each
   function it appears in.)
wrapper.cxx: In function `call_ret_val_t h323_clear_call(const char*)':
wrapper.cxx:1230: warning: unused variable `ClearCallThread*clearCallThread'
make[1]: *** [wrapper.o] Error 1
make[1]: Leaving directory `/root/voip/asterisk/asterisk-oh323/wrapper'
make: *** [subdirs_all] Error 1
What's wrong?
Thanks
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Re: [Asterisk-Users] Public vs. Private Network

[Andrew Latham]

yes

On 5/19/05, David Sampson [EMAIL PROTECTED] wrote:
  
  
 
 Hello  
 
   
 
 I am looking at connecting 7  10 locations together using Asterisk and
 possibly some VoIP gateway appliances.  I need to insure best voice quality
 as these trunks will be used primarily for customer calls.  I am considering
 implementing a full T1 frame relay circuit to each location which can be
 done for a reasonable cost.  DSL and Cable are currently at each location
 and setup for automatic failover.  Should I remove one of my public
 connections and replace it with a private circuit for best quality?
  
  Thank you, 
 
 
  Dave 
 
   
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-- 
sig
Andrew Latham - AKA: LATHAMA (lay-th-ham-eh)
WWW: http://lathama.com
Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED]
If any of the above are down we have bigger problems than my email!
/sig
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[Asterisk-Users] Selling: E100P interface card

[ms]

Hi,

I'm selling a E100P card. 
(32 channel ISDN PCI interface card. Works great with asterisk).

http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=5776112128

Michael

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[Asterisk-Users] User cannot dial

[Chris Mason (Lists)]

I have a user connecting from behind a firewall. The location is remote and
I have no access to the firewall to so any port forwarding.
She is using SJPHONE as the client. I can dial the extension and she can
answer, we can converse. However, she cannot dial out. Any ideas what causes
this?

Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 

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Re: [Asterisk-Users] Agent Queues and Sending URLs

[Richard Lyman]

Waldo Rubinstein wrote:
Thanks. I'm actually looking for a SIP client softphone.
Also, I checked the configuration and there is no space in between  
http:// and www.google.com. It must have gotten inserted when I  pasted 
the text.

Any other suggestions?
Waldo
On May 18, 2005, at 2:28 PM, Richard Lyman wrote:
Waldo Rubinstein wrote:
Hi guys,
I'm testing the sending of a URL to an XLite softphone when a call  
is  in queue. See the output of the CLI below:
-- Executing Queue(Zap/69-1, q_sample|tT|http://  
www.google.com/) in new stack
-- Started music on hold, class 'default', on Zap/69-1
-- outgoing agentcall, to agent '1000', on 'Local/  
[EMAIL PROTECTED],1'
-- Called Agent/1000
-- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/1000|20| t)  
in new stack
-- Called 1000
-- SIP/1000-cb3b is ringing
-- Agent/1000 is ringing
-- Starting simple switch on 'Zap/87-1'
-- SIP/1000-cb3b answered Local/[EMAIL PROTECTED],2
-- Agent/1000 answered Zap/69-1
-- Stopped music on hold on Zap/69-1
  == Spawn extension (agents, 1000, 1) exited non-zero on 'Local/  
[EMAIL PROTECTED],2'
It queues the application correctly. However, when the call is  sent  
to the agent, no URL is displayed. It is a bug in Asterisk  or is it  
that the XLite doesn't support it? Any help will be  greatly 
appreciated.
I have heard of others that have been able to do this with XLite,   
although I haven't actually seen it working or any sample configs.
Thanks,
Waldo

last i heard xlite was still working on this ability, try diax g or  q 
version (can't remember).  also fix your url string you have a  space 
after // and before www.


sorry, can't help you with sip. (i don't use it, too many 
headaches G)

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[Asterisk-Users] Re: Grandstream ATA 286 and ilbc

[Kevin McCauley]

Anton Krall akrall-lists at intruder.com.mx writes:

 
 Guys, anybody having problem with ilbc and GS ata 286? I just tried it for
 fun (always using alaw) and voices sounded quite bad... crappy voice
 prompts, not bad quality, just like weird noises.
 
 Anybody had this? whats the latest FW for those units?
 
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Anton,

I use iLBC exclusively on the 286/486 and it interoperates with other devices on
my network fine.  In fact I use iLBC because some of the people I talk to only
have dialup and it works the best for that.  

I will mention though, that I have stayed on FW version 1.0.5.16 since I have
had troubles with newer versions.

-Kevin

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Re: [Asterisk-Users] MusicOnHold probelms

[admin]

Do you have mpg123 installed?
Is there a .mp3 file available to play in your /var/lib/asterisk/mohmp3 
directory?

-daryl

-Original Message-
From: chawki hammoud [EMAIL PROTECTED]
To: Asterisk-Users@lists.digium.com
Cc: 
Date: Thu, 19 May 2005 06:03:55 -0700 (PDT)
Subject: [Asterisk-Users] MusicOnHold probelms

 This is my second attempt trying to get help and I am
 hoping someone can. When the musiconhold extension is
 matched, Asterisk attempts to execute musiconhold and
 stops right away, this is what I gets:
 
  Executing MusicOnHold(OSS/dsp, ) in new stack
 -- Started music on hold, class 'default', on
 OSS/dsp
 -- Stopped music on hold on OSS/dsp
 
 Is there a file that musiconhold try to play and can't
 find. Please help withy any suggestions.
 
 
 
   
 Discover Yahoo! 
 Stay in touch with email, IM, photo sharing and more. Check it out! 
 http://discover.yahoo.com/stayintouch.html
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RE: [Asterisk-Users] SIP Phone Recommendations?

[Ariel Batista]

Just want to let everyone know that even if there changing it out to the new
501 it's still on of the best. Remember that people are still buying the
Cisco 7960G which is being phased out as well.

The IP-500 works and works very well. I know that there price will be going
down soon once there are some supplies of the IP-501.  But if you need a
phone now it is a very good one for the price.  


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves
Sent: Thursday, May 19, 2005 10:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP Phone Recommendations?

On Wed, 18 May 2005 22:29:40 -0500, Kristian Kielhofner wrote:

Ariel,

   It's probably not a good idea to reccomend the IP 500/300 anymore. 
They are being phased out by Polycom because they (and the IP 300) only 
have 2mb of flash, and Polycom is looking to standardize on 4mb for 
their firmware (which the IP 600 has had since day one).

   If you are going to buy a Polycom now, get an IP 600, or, wait for
the 
301's or 501's.  Don't say I didn't warn you!

Good advice!. BTW, I LOVE my IP600's. 

I also kinda like the Zultys 4x4/4x5.The hardware and software is good
but their support arrangement is terrible. They provide no end user
support at all. Period. They rely upon their dealers to provide all
support, but then they're ok with signing up dealers that know nothing
about the products.

Michael

--
Michael Graves   [EMAIL PROTECTED]
Sr. Product Specialist  www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]

o713-861-4005
o800-905-6412
c713-201-1262



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Re: [Asterisk-Users] Deleting Monitor Files After 2 Months

[Steve Totaro]

- Original Message - 
From: Matthew Boehm [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Thursday, May 19, 2005 10:04 AM
Subject: Re: [Asterisk-Users] Deleting Monitor Files After 2 Months


Gavin Hamill wrote:
On Thursday 19 May 2005 13:51, Steve Totaro wrote:
Does anyone know the best way to automate the deletion of monitor
files after they age two months?
How about ...
$ find /path/to/files -ctime +60 -exec rm {}\;
Cheers,
Gavin.
Nice Gavin. I would further turn that into a shell script and pop it into
cron to run nightly.
-Matthew
Thanks! 

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[Asterisk-Users] Two TDM04 with Poweredge

[Tom Hayden]

Has anyone on this list succesfully managed to get two (or more) TDM04
(with four FXO each) working on a Dell PowerEdge server? If so, which
model? Was it a hassle?  I'm doing a seven-line installation and a
callbank seems like overkill, I just don't want to get suck with a
PowerEdge that gets into an IRQ mess.

Thanks in Advance,

Tom Hayden
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Re: [Asterisk-Users] Public vs. Private Network

[Eric Wieling aka ManxPower]

I am looking at connecting 7  10 locations together using Asterisk and
possibly some VoIP gateway appliances.  I need to insure best voice quality
as these trunks will be used primarily for customer calls.  I am considering
implementing a full T1 frame relay circuit to each location which can be
done for a reasonable cost.  DSL and Cable are currently at each location
and setup for automatic failover.  Should I remove one of my public
connections and replace it with a private circuit for best quality?
To run VoIP over Frame Relay you need your Port Speed to be the same 
as your CIR.  Cisco has extensive docs about this, but I'm too lazy to 
look them up right now.
--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
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Re: [Asterisk-Users] asterisk-oh323 build problems

[VoIP Newbie]

Read README file first. You will get a clue.

On 5/19/05, FaberK [EMAIL PROTECTED] wrote:
 Hello Guys,
 first of all, I'm very new with asterisk.
 I'm trying to set it up. I've already compiled and installed Asterisk-1.0.7
 Now I'm trying with asterisk-oh323
 I've already installed pwlib, oh323 and I've already set the variables.
 Now, when I try to make asterisk-oh323 I receive this error messagge:
 for x in wrapper asterisk-driver; do make -C $x all || exit 1 ; done
 make[1]: Entering directory `/root/voip/asterisk/asterisk-oh323/wrapper'
 g++ -Wall -mcpu=i586 -DP_LINUX -D_REENTRANT -DP_HAS_SEMAPHORES -DP_SSL
 -DP_PTHREADS -DPBYTE_ORDER=PLITTLE_ENDIAN -DPHAS_TEMPLATES -O3
 -DNDEBUG -I/usr/include -I/usr/include/crypto
 -I/usr/lib/pwlib/include/ptlib/unix -I/usr/lib/pwlib/include
 -I/usr/lib/openh323/include -I../asterisk-driver -g -c wrapper.cxx -o
 wrapper.o
 wrapper.cxx: In constructor
   `WrapH323Connection::WrapH323Connection(WrapH323EndPoint, unsigned int,
   int, int, short unsigned int)':
 wrapper.cxx:563: `SetMaxAudioDelayJitter' undeclared (first use this function)
 wrapper.cxx:563: (Each undeclared identifier is reported only once for each
   function it appears in.)
 wrapper.cxx: In function `call_ret_val_t h323_clear_call(const char*)':
 wrapper.cxx:1230: warning: unused variable `ClearCallThread*clearCallThread'
 make[1]: *** [wrapper.o] Error 1
 make[1]: Leaving directory `/root/voip/asterisk/asterisk-oh323/wrapper'
 make: *** [subdirs_all] Error 1
 
 
 What's wrong?
 
 Thanks
 
 --
 .:FaberK:.
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Re: [Asterisk-Users] Outbound dialing issue with FXO

[Johnathan Corgan]

Mike Clark wrote:
However, outbound calls are hit or miss. Sometimes they work fine and 
other times we get a you must first dial a 1 or 0 message back from 
telco when dialing out standard POTS lines.
Did you get this working yet?
-Johnathan
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RE: [Asterisk-Users] Two TDM04 with Poweredge

[David Brodbeck]

 -Original Message-
 From: Tom Hayden [mailto:[EMAIL PROTECTED]

 Has anyone on this list succesfully managed to get two (or more) TDM04
 (with four FXO each) working on a Dell PowerEdge server? If so, which
 model? Was it a hassle?

I've got a PowerEdge 800 tower server with two of them.  Only five FXO
modules right now, though.

It mostly works.  When I insert the driver I get an NMI, but that appears to
be harmless.  I have to unload and reload the drivers once a week or so,
otherwise the FXO modules tend to eventually stop responding.  I haven't had
any audio quality or interrupt problems, though.  

The system gets the job done, but I can't wholeheartedly recommend these
cards.  If I had to do it all over again, I'd consider some other method.
I'm not sure if anything else would be practical, though.  A T1 card plus
channel bank is kind of cost prohibitive for such a small installation.
I've heard good things about the Sipura gateways, but I'm interfacing to a
PBX and need the ability to flash the line for transfers, and I think
Flash() is Zap-specific.
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RE: [Asterisk-Users] Re: Grandstream ATA 286 and ilbc

[Anton Krall]

That's what I was starting to think.. Since I've always used ulaw or alaw...
Seems that firmware 1.0.5.23 has ilbc broken. 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Kevin McCauley
|Sent: Jueves, 19 de Mayo de 2005 10:15 a.m.
|To: asterisk-users@lists.digium.com
|Subject: [Asterisk-Users] Re: Grandstream ATA 286 and ilbc
|
|Anton Krall akrall-lists at intruder.com.mx writes:
|
| 
| Guys, anybody having problem with ilbc and GS ata 286? I 
|just tried it 
| for fun (always using alaw) and voices sounded quite bad... crappy 
| voice prompts, not bad quality, just like weird noises.
| 
| Anybody had this? whats the latest FW for those units?
| 
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| 
| 
|
|
|Anton,
|
|I use iLBC exclusively on the 286/486 and it interoperates 
|with other devices on my network fine.  In fact I use iLBC 
|because some of the people I talk to only have dialup and it 
|works the best for that.  
|
|I will mention though, that I have stayed on FW version 
|1.0.5.16 since I have had troubles with newer versions.
|
|-Kevin
|
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[Asterisk-Users] (no subject)

[M O]

BJ,

BJ Weschke [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Do Both! :) Re: Telecom
SIP termination vs. DS3
To: Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1

Did I miss pricing/availability announcements from
Digium on that DS3 card somewhere? 

No idea.  You can contact them if you dont know what
you missed :) 

I wasn't aware they were going to be GA in less than
3
weeks from now.

From my standpoint, I am just so anxious and 
confident that the Digium DS3 Channelized Voice PCI
Card, whenever I get my order of DID #'s and test my
configuration of Asterisk, that I am willing to
prepay, 
or have available to Digium, whatever $$$ they want 
for the card.

I am EVENTUALLY going to need it anyways, so I dont
mind prepaying wheather or not it is available today!
My knowledge of their product offering is no different

than yours.  But I fully intend on purchasing it :)!

We are starting off with a 100Mbps burstable bandwith,
though exspensive to start, after 30 days of usage, my
bandwidth costs will look like $25K.  Going off the
top of head for a Sangoma DS3 Card @ $6000 per card,
If I got 2 of them for $12,000 total, I eliminate,
almost, that $25,000 per month bandwidth cost to me.

So if Digiums DS3 Channelized Voice PCI card costs,
around what Sangomas costs, $6,000, (JUST AS A EXAMPLE
FOR THIS POST), $12,000 for 2 Digium DS3's in 1 month,
I will save almost $10,000 AUTOMATICALLY and ever
month thereafter! :)

Come on Txlink DID #'s.

Come on Digium with the DS3 Channelized Voice PCI
card.

Then all Digium would have left to do is create a
board
or work with someone on getting Radio Waves into your
computer.  :)

Sincerely,

SoftwareRadioGuy



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[Asterisk-Users] Re: Always Ringing

[VoIP Newbie]

Can anyone give me a big hand here??

On 5/16/05, VoIP Newbie [EMAIL PROTECTED] wrote:
 Hi all,
 
 I am using chan_h323 from Asterisk CVS to interconnect with GNUGK
 v2.2.2. Then I made call from a H323 EP, thru GNUGK, to SIP EP on
 Asterisk. However, I only heard ringing when the call was answered on
 SIP side. Below is the debug from chan_h323. Any help is welcome.
 Thanks.
 
 *CLI   == New H.323 Connection created.
-- Setting up Call
--  Call token:  [ip$22.7.20.32:30012/16050]
--  Calling party name:  [6907]
--  Calling party number:  [6907]
--  Called party name:  [0069777]
--  Called party number:  [0069777]
--Received SETUP message
=-= In OnAnswerCall for call 16050
- Progress Indicator: 0
- Inserting PI of 0 into ALERTING message
-- Started logical channel: sending G.729
-- channelsOpen = 1
External RTP Session Starting
RTP channel id 1 parameters:
-- remoteIpAddress: 22.7.20.32
-- remotePort: 51048
-- ExternalIpAddress: 0.0.0.0
-- ExternalPort: 17816
-- Started logical channel: receiving G.729
-- channelsOpen = 2
External RTP Session Starting
RTP channel id 1 parameters:
ExternalRTPChannel Destroyed
ExternalRTPChannel Destroyed
-- Executing Dial(H323/ip$22.7.20.32:30012/16050, SIP/69777)
 in new stack
-- Called 69777
-- SIP/69777-c6ce is ringing
Sending alerting
 
-- SIP/69777-c6ce answered H323/ip$22.7.20.32:30012/16050
Answering call ip$22.7.20.32:30012/16050
-- Transmitting RFC2833 on payload 96
-- Received Facility message...
=-= In OnConnectionEstablished for call 16050
-- Connection Established with 6907 [22.7.20.32]
-- Received Facility message...
-- Started logical channel: receiving G.729
-- channelsOpen = 3
External RTP Session Starting
RTP channel id 1 parameters:
-- Received Facility message...
-- Received RELEASE COMPLETE message...
-- ClearCall: Request to clear call with token
 ip$22.7.20.32:30012/16050, cause EndedByRemoteUser
-- Sending RELEASE COMPLETE
channelsOpen = 2
channelsOpen = 1
channelsOpen = 0
ExternalRTPChannel Destroyed
ExternalRTPChannel Destroyed
ExternalRTPChannel Destroyed
-- ClearCall: Request to clear call with token
 ip$22.7.20.32:30012/16050, cause EndedByTransportFail
  == Spawn extension (default, 0069777, 1) exited non-zero on
 'H323/ip$22.7.20.32:30012/16050'
 -- 6907 [22.7.20.32] has cleared the call
== H.323 Connection deleted.

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[Asterisk-Users] Re: SIP and FastStart

[VoIP Newbie]

Can anyone give me a big here?



On 5/13/05, VoIP Newbie [EMAIL PROTECTED] wrote:
 I am using Asterisk-oh323 v0.7.1 with GNUGK. Please advise what must
 be done  to make FastStart work with SIP phones. Thanks.
 
 On 5/12/05, VoIP Newbie [EMAIL PROTECTED] wrote:
  Hi all,
 
  When I enabled faststart in oh323.conf, calls from H323 endpoint to
  SIP phones could not complete. The originating phone kept ringing when
  calls were answered by SIP phones.
 
  fastStart=yes
  h245Tunnelling =yes
  h245inSetup=yes
 
  Please can you advise.
 
  Many Thanks.
 

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[Asterisk-Users] ACD Methods

[Marshall, Ed]

Can anyone point me in the right direction of info regarding ACD methods
available in Asterisk.

As far as I can see there are time based ring strategies available but I
cannot find any info regarding skills based routing or queue priorities.

Also do the current time based ring strategies work globally.  What I mean
by this is if an agent is a member of more than one queue then would the ACD
algorithm take this into account before deciding to allocate another call ?

Any help would be much appreciated.

Regards
Ed
 


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[Asterisk-Users] Can't make outgoing calls

[Nick Heinemans]

Hello,

When I try to make an outgoing call from my X-lite softphone connected to
Asterisk, I keep getting the following error message:
May 19 18:42:58 WARNING[3086]: Forbidden - wrong password on authentication
for INVITE to '31307110340 sip:[EMAIL PROTECTED];tag=as13ba1ff7'

I'm running AAH 1.0 on a server which is directly hooked up to my ADSL line.
It's second NIC is connected to my LAN on which the PC with X-lite is also
connected. I've configured the Asterisk server as a NAT router and I opened
UDP ports 5060 and 1-2 from the outside.

Any idea what might be wrong?

Regards, Nick

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Re: [Asterisk-Users] Asterisk real time extensions problem...

[Gentian Bajraktari]

HI,

The problem is that you are using: 
incoming-next,60069,1
Use: incoming-next|60069|1 
instead

RG,
Gentian



  - Original Message - 
  From: 
  Bharat M. Sarvan 
  To: 'Asterisk Users Mailing List - 
  Non-Commercial Discussion' 
  Cc: asterisk-dev@lists.digium.com 
  
  Sent: Sunday, April 17, 2005 11:52 
  AM
  Subject: [Asterisk-Users] Asterisk real 
  time extensions problem...
  
  
  Hello 
  everybody,
   
  I have setup asterisk real time extensions and its working pretty well. But 
  the problem is when I am jumping between the contexts using the Goto statement 
  in the database. I am getting a error 
  
  = Parsing 
  '/etc/asterisk/sip_notify.conf': Found
   -- 
  SIP Seeding peers from Astdb: 'ezzibpo4' at [EMAIL PROTECTED]:5061 
  for 60
   -- 
  Executing Goto("SIP/ezzibpo4-4636", 
  "incoming-next,6069,1")
  May 19 05:00:04 
  NOTICE[6420]: pbx.c:1688 pbx_extension_helper: Cannot find extension '6069' in 
  context 'incom'
  May 19 05:00:04 
  WARNING[6420]: pbx.c:6256 ast_parseable_goto: Priority 
  'incoming-next,
  
  The structure of the extensions db 
  is as given below
  
  ++---+---+--+-+--+
  | id | 
  context | exten | 
  priority | 
  app | 
  appdata 
  |
  ++---+---+--+-+--+
  | 1 | 
  incoming | 6069 
  | 1 | 
  Goto | 
  incoming-next,6069,1 |
  | 2 | 
  incoming | 6069 
  | 2 | 
  Hangup 
  | 
  |
  | 3 | incoming-next | 
  6069 | 1 | 
  DigitTimeout | 
  10 
  |
  | 4 | incoming-next | 
  6069 | 2 | ResponseTimeout | 
  30 
  |
  | 5 | incoming-next | 
  6069 | 3 | 
  Background | 
  welcome 
  |
  
  
   The 
  context “incom” in the above error is the context defined for placing outgoing 
  call in the sip.conf file. I don’t understand why is it looking for extension 
  6069 in the “incom” context.
  
   The 
  “Goto” statement in the context incoming is getting executed without any 
  probs, but the control is not getting transferred to the context 
  “incoming-next” upon execution of the Goto statement. 
  
   
  
   
  Could anybody suggest me as to where might the problem be and any way to get 
  rid of this problem. Please do reply….
  
  
  
  
  Regards,
  Bharat 
  M. Sarvan
  
  
  

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[Asterisk-Users] AS5300 - Meridian Configuration

[Aaron Daniel]

We're trying to set up a connection between an AS5300 and a meridian  
CSU/DSU so our asterisk system can interconnect with our current  
legacy system, and for some reason the T1 connection will not come up  
whatsoever.  I've gone through all the configurations I can think of,  
even basically copied our current cisco settings directly to the  
AS5300 so they would be nearly identical, and nothing.  Any help  
would be appreciated.

AS5300 config:
Current configuration : 2094 bytes
!
! Last configuration change at 11:45:56 CDT Thu May 19 2005
! NVRAM config last updated at 11:46:14 CDT Thu May 19 2005
!
version 12.3
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname cdial2
!
boot-start-marker
boot-end-marker
!
enable secret 5 ***
enable password ***
!
spe 1/0 1/7
firmware location system:/ucode/mica_port_firmware
spe 2/0 2/7
firmware location system:/ucode/mica_port_firmware
!
!
resource-pool disable
clock timezone CST -6
clock summer-time CDT recurring
!
no aaa new-model
ip subnet-zero
no ip routing
ip finger
ip domain name shsu.edu
ip name-server 158.135.1.20
ip name-server 158.135.1.200
!
!
isdn switch-type primary-5ess
isdn voice-call-failure 0
!
!
!
!
!
!
!
!
!
!
!
!
controller T1 0
shutdown
framing sf
linecode ami
!
controller T1 1
shutdown
framing sf
linecode ami
!
controller T1 2
framing esf
clock source line primary
linecode b8zs
pri-group timeslots 1-24
!
controller T1 3
shutdown
framing sf
linecode ami
!
!
interface Ethernet0
no ip address
no ip route-cache
shutdown
!
interface Serial2:23
no ip address
ip mroute-cache
dialer-group 1
isdn switch-type primary-5ess
isdn protocol-emulate network
isdn incoming-voice modem
isdn disconnect-cause 1
fair-queue 64 16 3
no cdp enable
ip rsvp bandwidth
ip rtp reserve 1 1
!
interface FastEthernet0
ip address 158.135.1.61 255.255.0.0
no ip route-cache
no ip mroute-cache
duplex full
speed 100
no mop enabled
!
ip classless
no ip http server
!
!
!
!
!
!
!
dial-peer voice 6 voip
incoming called-number 6
destination-pattern 6
session protocol sipv2
session target sip-server
!
dial-peer voice 4 pots
application session
direct-inward-dial
!
sip-ua
retry invite 3
retry response 3
retry bye 3
retry cancel 3
timers trying 1000
sip-server dns:sipproxy1.shsu.edu
!
!
line con 0
line 1 96
line aux 0
line vty 0 4
password ***
login
!
scheduler interval 1000
ntp clock-period 17180204
ntp update-calendar
ntp server 158.135.1.2
!
end
meridian config:
ADAN DCH 13
   CTYP MSDL
   GRP  0
   DNUM 6
   PORT 3
   DES  ASVOIP
   USR  PRI
   DCHL 25
   OTBF 32
   PARM RS422  DTE
   DRAT 64KC
   CLOK EXT
   IFC ESS5
   SIDE USR
   CNEG 1
   RLS  ID   36
   RCAP ND2
   T200 3
   T203 10
   N200 3
   N201 260
   K7
Thanks,
Aaron Daniel
Senior Voice Analyst
Sam Houston State University
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Re: [Asterisk-Users] Public vs. Private Network

[William Suffill]

Point to Point connectivity if they are close enough. Only use
DSL/Cable if you have to since results may vary depnding on
location/route/utilization/ISP.


On 5/19/05, Andrew Latham [EMAIL PROTECTED] wrote:
 yes
 
 On 5/19/05, David Sampson [EMAIL PROTECTED] wrote:
 
 
 
  Hello 
 
 
 
  I am looking at connecting 7  10 locations together using Asterisk and
  possibly some VoIP gateway appliances.  I need to insure best voice quality
  as these trunks will be used primarily for customer calls.  I am considering
  implementing a full T1 frame relay circuit to each location which can be
  done for a reasonable cost.  DSL and Cable are currently at each location
  and setup for automatic failover.  Should I remove one of my public
  connections and replace it with a private circuit for best quality?
 
   Thank you,
 
 
   Dave
 
 
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 --
 sig
 Andrew Latham - AKA: LATHAMA (lay-th-ham-eh)
 WWW: http://lathama.com
 Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED]
 If any of the above are down we have bigger problems than my email!
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Re: [Asterisk-Users] Phone keypad input not working during menu's

[Don]

Wilson Pickett wrote:
What codec are your phones using and which do you have in sip.conf in
general and phone entries?
 

Hi Wilson,
Thanks for the reply.  I didn't know anything about codecs but I've 
tried to look up what I can.  The Polycom documentation (SIP admin 
guide) says the hone supports G.711u-law, G.711a-law, G.729AB, SID and 
RFC2833.  The phone configuration files say that the preference is for 
u-law, a-law and AB in that order.  My sip.conf file says:

disallow=all
allow=ulaw
allow=alaw
I would guess that means I'm ok (i.e. ulaw is good on both sides) but 
this is a new area of * for me.  What do you think?

Don
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[Asterisk-Users] Connecting an External Extension

[Mark Brown]

Hi Everyone,

What is the best way to setup a SIP
Extension that is located outside of the AAH Server network?

The external SIP Phone is assigned a
public IP address and I would like to connect to an AAH server located behind a
NAT Router.

I have already mapped ports etc on the NAT
Router.

I am able to connect to the AAH Server and
the external extension can be rung from other internal extensions but falls
over once the call is answered.

Regards

Mark








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[Asterisk-Users] Cisco Call Manager Asterisk for Voicemail

[[EMAIL PROTECTED]]

Has anybody successfully (or I guess unsuccessfully for that matter) 
implemented Cisco Call Manager and used an * box for voicemail?  I 
checked the wiki and google and I see some references to Call Manager 
Express and *, but CME is completely different than CM.  If anybody has 
done this or has any insight, it would be appeciated.  We are trying to 
migrate ~ 300 users off of Cisco Unity and onto * for voicemail so that 
we have more flexibility and a lot lower maintenance costs.  Thanks.

Peder
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Re: [Asterisk-Users] Run Script when originator hangs up the phone

[Michiel van Baak]

On 07:47, Thu 19 May 05, Peter Svensson wrote:
 On Wed, 18 May 2005, Erik Sundberg wrote:
 
  Wonder if there was away to run a script/marco when the person who
  originates the call hangs up.
  
  I have use the g option in the dial application to continue running
  applications in the dial plan,  but that only works if the person who is
  called hangs up first..
 
 Use the 'h' extension. That is run when the current channel (the caller 
 normally) hangs up.
 
 Peter
 

And if you want an agi script run with the 'h' extension,
execute it with: exten=h,1,DeadAgi(myagi|myparams) instead
of just Agi()

-- 
Michiel van Baak
http://lunteren.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D

Two of the most famous products of Berkeley are LSD and BSD. I don't think 
that this is a coincidence.

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Re: [Asterisk-Users] User cannot dial

[David John Walsh]

When you say she can't dial out, what error message is she reciving?
(if your using the windows version, turn off the skin, then you get an
info button, click on that and you get another box below the user side
- it gives some debug but not a lot)

does you asterisk box see any packets from her?  

As she is behind a firewall, and you can ring her, it means that your
asterisk box has seen her register requests and has communicated with
her, so its unlikely to be the SJphone, unless there are some wayward
settings on it

also what settings does she have on her asterisk profile?

does it work with x-lite

if (on asterisk cli) you do sip debug ip (her ip address and port) ;
or sip debug peer peername and try to make a call - do you see
anything comming in?

David

On 5/19/05, Chris Mason (Lists) [EMAIL PROTECTED] wrote:
 I have a user connecting from behind a firewall. The location is remote and
 I have no access to the firewall to so any port forwarding.
 She is using SJPHONE as the client. I can dial the extension and she can
 answer, we can converse. However, she cannot dial out. Any ideas what causes
 this?
 
 Chris Mason
 NetConcepts
 (264) 497-5670 Fax: (264) 497-8463
 Int:  (305) 704-7249 Fax: (815)301-9759
 
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RE: [Asterisk-Users] Public vs. Private Network

[David Sampson]

Does anyone else have info regarding the port speed matching the CIR?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
Wieling aka ManxPower
Sent: Thursday, May 19, 2005 11:55 AM
To: Andrew Latham; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] Public vs. Private Network


I am looking at connecting 7 - 10 locations together using Asterisk
and
possibly some VoIP gateway appliances.  I need to insure best voice
quality
as these trunks will be used primarily for customer calls.  I am
considering
implementing a full T1 frame relay circuit to each location which can
be
done for a reasonable cost.  DSL and Cable are currently at each
location
and setup for automatic failover.  Should I remove one of my public
connections and replace it with a private circuit for best quality?

To run VoIP over Frame Relay you need your Port Speed to be the same 
as your CIR.  Cisco has extensive docs about this, but I'm too lazy to 
look them up right now.
-- 
Always do right. This will gratify some people and astonish the rest.
Mark Twain
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RE: [Asterisk-Users] Two TDM04 with Poweredge

[Ariel Batista]

I have 2 of them working on a SC420 server and also another one the SC400
and older one that has 4 TDM boards on it. Both systems have been working
fine.  

I did not have to do anything special on them to get them working.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Hayden
Sent: Thursday, May 19, 2005 11:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Two TDM04 with Poweredge

Has anyone on this list succesfully managed to get two (or more) TDM04
(with four FXO each) working on a Dell PowerEdge server? If so, which
model? Was it a hassle?  I'm doing a seven-line installation and a
callbank seems like overkill, I just don't want to get suck with a
PowerEdge that gets into an IRQ mess.

Thanks in Advance,

Tom Hayden
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[Asterisk-Users] Expression in Extension

[Hugh L. Johnson]

Does ^ work as a NOT in an expression for extensions?
Are the following equivalent?

exten = _58[^389],1,dial(${${EXTEN}},${RINGLONG},tr)

exten = _58[0124567],1,dial(${${EXTEN}},${RINGLONG},tr)

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[Asterisk-Users] RHEL 3

[Huddleston, Robert]

Anyone tried to build * + h323 to rhel3...
I have to problems in the process...

a) Zaptel would not build - a whole bunch of errors about kernel...
b) make progdocs failed with reference to dot - check your installation.

Do I need the zaptel ?? I will not be using any interface cards..
I'd like to make progdocs - any suggestions there?

Latest cvs on everything..
Thanks
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[Asterisk-Users] Re: Grandstream ATA 286 and ilbc (Anton Krall)

[Freddi Hansen]

  That's what I was starting to think.. Since I've always used ulaw or alaw...
Seems that firmware 1.0.5.23 has ilbc broken. 

|-Original Message-

Hi,
it works for me with that firmware but you must set the ilbc
framerate to 30.
(worked with framerate=20 until the 1.0.5.23 release)
Freddi




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Re: [Asterisk-Users] FXO Gateways

[Adrian A]

I am in Canada and telephone company is Telus. I did also try with a
Vonage ATA plugged into the Mediatrix but Caller ID still does not go
through.  The caller's number is nowhere to be found in the SIP
message sent from Mediatrix to Asterisk.

On 5/18/05, Calin Serbanescu [EMAIL PROTECTED] wrote:
 which country are you in and what is your provider ?
 
 On Wed, 2005-05-18 at 12:25 -0700, Adrian A wrote:
  Does anyone have any experience with the Audiocodes MP-108 FXO
  gateway?  I'm looking to get one for incoming PSTN lines.
 
  In particular, does it pass caller ID information to Asterisk?
 
  I currently have a Mediatrix 1204 but Caller ID does not work, even
  though the specs say it does.  All it sends are the names of the ports
  set up internally on the gateway (ie. pstnline1 etc) when a call
  comes in.
 
  Thanks.
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[Asterisk-Users] LOOKING TO HIRE

[Jeffrey Richey]

We have
positions in Ho Chi Minh City, Vietnam and Temecula, California.
Please only reply to [EMAIL PROTECTED] no phone calls.

Our
Company comprises a diverse set of individuals who work hard and play hard. We
look for motivated, dedicated candidates who have demonstrated an insatiable
quest for knowledge, opportunity, responsibility and entrepreneurship. Our
goals are ambitious, but ample rewards exist for those who embrace the challenge.


If you
are up for the challenge of helping to shape the future of an industry-leading
VoIP services firm, check out our list of available positions.

Vietnam Office Address:
Our Company
 
Saigon Trade
 Center Building
37 Ton Duc Thang Street
District 1
Ho Chi Minh City, Vietnam
Email: [EMAIL PROTECTED] 

You are
welcome to e-mail your CV or resume [EMAIL PROTECTED] 

Available Positions in Ho
  Chi Minh City, Vietnam:


Senior
Programmer
Job Description:
This position requires significant technical expertise in the design
and implementation of of object oriented programming and web applications.
Qualified applicants will have a minimum of 8 years experience in applications
development, have a thorough understanding of industry standard software
development procedures and practices, and have successfully developed and
implemented medium to large scale projects. Strong familiarity with the
Python programming language as well as the Zope application server and Plone
content management framework is required as the applications will be developed
using these tools. Knowledge of HTML/CSS also beneficial. Experience working in
a Unix/Linux environment is required. Basic Unix/Linux system administration
skills and knowledge of MySQL database and SQL is preferred. Good spoken and
written English language skills.

As a Senior Programmer, work with programmers to develop the application base
from specifications provided by management.

Review and make technical recommendations on code developed by programmers.

Mentor lower level programmer in knowledge transfer during design, build, test
and implementation phase of the project.

Provide System documentation for each phase of the project.


Minimal
Requirements:
Language Requirements: Perl, PHP, MySQL; Python is a bonus
Operating Systems Requirements: Linux Redhat or FREEBSD
Solid knowledge of Unix based systems, TCP/IP Protocols, CVS
Strong Knowledge of: www.zope.org and www.plone.org 


Programmer
Job Description:
The programmer will be responsible for implementing code in the
Python language in the Zope web application server as well as standalone Python
applications according to the specifications provided by management. 2+
years of Python programming experience, knowledge of SQL/MySQL, comfortable
working in a Linux/Unix environment. Prior experience developing database
driven web applications is a plus. Ability to speak and read/write english.

Minimal
Requirements:
Language Requirements: Perl, PHP, MySQL; Python is a bonus
Operating Systems Requirements: Linux Redhat or FREEBSD
Solid knowledge of Unix based systems, TCP/IP Protocols, CVS
Strong Knowledge of: www.zope.org and www.plone.org 


Web hosting
support engineer
Job Description:
Responsible for providing technical support to clients, basic system
administration tasks, maintaining security, and assisting the sales team with
pre and post-sales support to clients in Vietnam while working with an
english speaking team. 

Minimal
Requirements:
Applicants should be very familiar with the Linux operating
system, Apache web server, be familiar with basic security concepts, and have
some experience with at least one programming language such as PHP, Perl, or
Python. Spoken and written english language skills. 


Web developer
Job Description:
The web developer will be responsible for implementing the HTML/CSS
to achieve a professional look and feel for Our Company websites. The websites
should adhere to W3C standards and be easily
accessable to all web browsers. Note that we are not necessarily interested in
flash artists or photoshop gurus. Photoshop (or even better, Gimp!) and graphic
design skills will be required but HTML/CSS should really be the focus.
Experience developing in Zope/Plone a plus. Ability to speak/write english.

Minimal
Requirements:
Thorough knowledge of CSS and XHTML. Good eye for artistic
design and user interface design.


Project Manager
Job Description:
The Software Project Manager is responsible for leading a
project team involved in the requirement specification, technical design,
coding, integration, quality assurance test, and deployment of software
projects. The Project Manager manages software projects at the managerial and
project task level in accordance with Our Company's software development
methodology. The Project Manager may be assigned to lead a major project, a
series of concurrent projects and / or ongoing maintenance of systems. The
Project Manager provides leadership to the project team in order to establish
an accurate 

[Asterisk-Users] LOOKING TO HIRE

[Jeffrey Richey]

We have
positions in Ho Chi Minh City, Vietnam and Temecula, California.
Please only reply to [EMAIL PROTECTED] no phone calls.

Our
Company comprises a diverse set of individuals who work hard and play hard. We look
for motivated, dedicated candidates who have demonstrated an insatiable quest
for knowledge, opportunity, responsibility and entrepreneurship. Our goals are
ambitious, but ample rewards exist for those who embrace the challenge. 

If you
are up for the challenge of helping to shape the future of an industry-leading
VoIP services firm, check out our list of available positions.

Vietnam Office Address:
Our Company
 
Saigon Trade Center
  Building
37 Ton Duc Thang Street
District 1
Ho Chi Minh City, Vietnam
Email: [EMAIL PROTECTED] 

You are
welcome to e-mail your CV or resume [EMAIL PROTECTED] 

Available Positions in Ho
  Chi Minh City, Vietnam:


Senior
Programmer
Job Description:
This position requires significant technical expertise in the design
and implementation of of object oriented programming and web applications.
Qualified applicants will have a minimum of 8 years experience in applications
development, have a thorough understanding of industry standard software
development procedures and practices, and have successfully developed and
implemented medium to large scale projects. Strong familiarity with the
Python programming language as well as the Zope application server and Plone
content management framework is required as the applications will be developed
using these tools. Knowledge of HTML/CSS also beneficial. Experience working in
a Unix/Linux environment is required. Basic Unix/Linux system administration
skills and knowledge of MySQL database and SQL is preferred. Good spoken and
written English language skills.

As a Senior Programmer, work with programmers to develop the application base
from specifications provided by management.

Review and make technical recommendations on code developed by programmers.

Mentor lower level programmer in knowledge transfer during design, build, test
and implementation phase of the project.

Provide System documentation for each phase of the project.


Minimal
Requirements:
Language Requirements: Perl, PHP, MySQL; Python is a bonus
Operating Systems Requirements: Linux Redhat or FREEBSD
Solid knowledge of Unix based systems, TCP/IP Protocols, CVS
Strong Knowledge of: www.zope.org and www.plone.org 


Programmer
Job Description:
The programmer will be responsible for implementing code in the
Python language in the Zope web application server as well as standalone Python
applications according to the specifications provided by management. 2+
years of Python programming experience, knowledge of SQL/MySQL, comfortable
working in a Linux/Unix environment. Prior experience developing database
driven web applications is a plus. Ability to speak and read/write english.

Minimal
Requirements:
Language Requirements: Perl, PHP, MySQL; Python is a bonus
Operating Systems Requirements: Linux Redhat or FREEBSD
Solid knowledge of Unix based systems, TCP/IP Protocols, CVS
Strong Knowledge of: www.zope.org and www.plone.org 


Web hosting
support engineer
Job Description:
Responsible for providing technical support to clients, basic system
administration tasks, maintaining security, and assisting the sales team with
pre and post-sales support to clients in Vietnam while working with an english
speaking team. 

Minimal
Requirements:
Applicants should be very familiar with the Linux operating
system, Apache web server, be familiar with basic security concepts, and have
some experience with at least one programming language such as PHP, Perl, or
Python. Spoken and written english language skills. 


Web developer
Job Description:
The web developer will be responsible for implementing the HTML/CSS
to achieve a professional look and feel for Our Company websites. The websites
should adhere to W3C standards and be easily
accessable to all web browsers. Note that we are not necessarily interested in
flash artists or photoshop gurus. Photoshop (or even better, Gimp!) and graphic
design skills will be required but HTML/CSS should really be the focus.
Experience developing in Zope/Plone a plus. Ability to speak/write english.

Minimal
Requirements:
Thorough knowledge of CSS and XHTML. Good eye for artistic
design and user interface design.


Project Manager
Job Description:
The Software Project Manager is responsible for leading a
project team involved in the requirement specification, technical design,
coding, integration, quality assurance test, and deployment of software
projects. The Project Manager manages software projects at the managerial and
project task level in accordance with Our Company's software development
methodology. The Project Manager may be assigned to lead a major project, a
series of concurrent projects and / or ongoing maintenance of systems. The Project
Manager provides leadership to the project team in order to establish an