date:20050425

I fallback on my PRI at $.039 per minute (LD) and up.. Argghgh!

Consider this:
http://www.voip-info.org/tiki-index.php?page=Asterisk+least+cost+routing+usi
ng+broadvoice


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Monday, April 25, 2005 5:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Gerard Marcel
Subject: Re: [Asterisk-Users] voip problems

> How do you guys deal with voip problems?  do you have multiple backups
> such as land lines, and different voip providers?

Given the number of BV posts over the last several months, it would
certainly appear stability is not in their vocabulary. (I'm a past
user, but not any more.)

Guess I'd have to question why someone would commit a business to
using an "unsupported" service that is somewhat unstable, has limited
codec support, and that relies on the asterisk-user list for support.

Someone else already mentioned the need for backup arrangements.

What do I use? livevoip.com for 800 number, teliax.com for did's,
and, pstn analog's for backup and some outgoing calls. I can guarantee
not all three will be down at the same time (except for the local
fiber seeking backhoe). :)


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[Asterisk-Users] Only want softphone account from Vonage:









    Vonage allows you to only purchase a softphone
account AFTER having signed up for a regular device-attached number. Today when
I cancelled my Vonage account (switched to Broadvoice) I noticed I was still
able to go in and add softphone accounts after they removed the Linksys PAP
attached line and all the softphones (6). Might be handy way for someone
wanting Vonage who doesn’t need an FXS device to get only the service
they want.






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RE: [Asterisk-Users] Broadvoice Down?

Was it just one of their proxies, or all three that was having
problems? I use "mia" which is Miami? I didn't hear any complaints from my
grunts and I use BV for all my outbound intl and inter-state calls for an
office of about 80 or so. 

Also, I tried to register at all three proxies but got a 404 back
after the first registration succeeded. I might see if I can work up a
scheme to do a CHANAVAIL on all three before the call goes out. 

DCA sipBroadvoice3/x  147.135.0.128 N 255.255.255.255  5060 OK (87 ms)
LAX sipBroadvoice2/x  147.135.8.128 N 255.255.255.255  5060 OK (86 ms)
MIA sipBroadvoice1/x  147.135.4.128 N 255.255.255.255  5060 OK (156 ms)
This is what I am seeing in Dallas.


From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of JD Austin
Sent: Monday, April 25, 2005 4:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Broadvoice Down?





On Mon, 2005-04-25 at 12:48, JD Austin wrote:
  
I guess 2 hours is 'soon' to them.  I'm still down.
Is there a reliable voip provider out there that works with Asterisk?
I can't have downtime like this.. it just makes my company look bad.

JD


I just came up.. 4 hours down time.
There are a few changes on broadvoices' web site: 
http://www.broadvoice.com/support_install_asterisk.html
I made the changes, but they didn't immedately work so I don't think that
was it.

What does 'pedantic=no' mean in sip.conf?


JD

JD Austin wrote: 

I tried calling Broadvoice support.. on hold for 1/2 hour then it
hung up on me with a reorder (fast busy), so I tried again.
Just got through to a rep- they said it's a 'carrier issue' that
their 'partner carrier' was having issues and that it would be up
soon.
Makes me wonder if I should be signing up with their 'partner
carrier' instead.

JD

Jerry Geis wrote: 
  
I am having the same broadvoice issue at the moment.

jerry
Is anyone else having difficulty with their Broadvoice service? When I 
dial my number right now it rings either fast busy or tells me it cannot 
complete the call.

I can make outgoing calls from my system through broadvoice however. 
Seems their inbound trunks hit capacity?

Am I alone in this?
-Max




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-- 
JD Austin
Twin Geckos Technology Services LLC
email: [EMAIL PROTECTED]
http://www.twingeckos.com
phone/fax: 480.344.2640 

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[Asterisk-Users] Digium Quad Span Cards

2005-04-25 Thread Callum McGillivray

Hi all,
Just a quick question for everyone.
I understand that there is problems using more than 1 digium quad card 
in a single machine.  Can anyone explain to me what the problems are?

Also, I was wondering what the minimum spec would be for a * machine 
with a single quad card that was running a total of 120 concurrent calls.

Can someone give me some real world examples of what they might have 
used in the field ?

Thanks,
Callum
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RE: [Norton AntiSpam] [Asterisk-Users] Error on the Mysql, realtimedatabase HELP so close so far; .

2005-04-25 Thread Paul Dracevich

Ok..errr (now I am really stumped)

Please help.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ronald
Wiplinger
Sent: Tuesday, April 26, 2005 3:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Norton AntiSpam] [Asterisk-Users] Error on the Mysql,
realtimedatabase HELP so close so far; .

Paul Dracevich wrote:

> *CLI> Apr 26 16:08:04 ERROR[1882]: res_config_mysql.c:588 
> mysql_reconnect: MySQL RealTime: Failed to connect database server 
> asterisk on . Check debug for more info.
>
>  
>
> Regards
>
> Paul Dracevich
>
> Wireless Technology Consultant
>
> Wayby Group
>
>  
>
> Mobile +64 29 638 9675
>
> Phone +64 9 623 2143
>
> Fax +64 9 623 1380
>
> email [EMAIL PROTECTED]
>
> website www.vnet.cc 
>

>
>  
>
> "the freedom to communicate is the right of every individual in the 
> 21st century" "Intellectual Property protection is the key to the 
> Knowledge Economy" This email was sent to you via YOUtopia ... it's 
> all about YOU.
>
>  
>
> The information contained in this email and any attachments is 
> confidential and may be legally privileged. If you are not the 
> intended recipient then you must not use, disseminate, distribute or 
> copy any information contained in this email or any attachments. If 
> you have received this email in error, please contact us immediately 
> and delete this email. Thank you.
>
>  
>
>---
-
>
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Do you think that the message is in relation to the signature too
short



bye

Ronald

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RE: [Asterisk-Users] Polycom ip500 (Not-Registered)

2005-04-25 Thread Eric Alexander

Make sure the address is the same as the userid, IE 
reg.X.auth.userId="username" and reg.X.address="username" 


-Original Message-
From: [EMAIL PROTECTED] on behalf of Sean Kennedy
Sent: Mon 4/25/2005 9:11 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Polycom ip500 (Not-Registered)
 
Dan Morin wrote:

> I just got a few Polycom IP500s and I've been following the info in 
> the wiki trying to configure them. From what I can tell, they seem to 
> be setup correctly (well.they don't work so obviously not.) however, 
> when they try to register with Asterisk, the following error shows up 
> in the Logs:
>
> Apr 25 15:22:05 NOTICE[1718]: Registration from '' failed for 
> '192.168.0.222'
> Apr 25 15:22:20 DEBUG[1718]: Auto destroying call 
> '[EMAIL PROTECTED]'
>
> Where 192.168.0.222 is the IP of the phone. The two single quotes seem 
> to indicate that no credentials are being passed to * (?). If anyone 
> has any experience with these, please let me know.
>
> I can post the configs if that would help. Thanks in advance.
>
> Dan
>
Please do. Specifically, sip.conf ( or whatever your sip configuration 
file for the phones is ), the individual phone settings and your 
sip.conf file from asterisk ( relevant parts only ).

Sean
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RE: [Asterisk-Users] Asterisk integration with Alcatel 4400

I was able to fully integrate my Lucent/Avaya Definity G3. Basically it has
a TIE line PRI between the PBX and the *. I had to do some legacy pbx tricks
on the Definity to make it send the calls across, but it seems to work
pretty well. I would assume the Alcatel could do the same.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Monday, April 25, 2005 4:33 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk integration with Alcatel 4400

I have just finished to look at the interesting documentation linked from
voip-info.org entitled:

PBX Interoperability: Alcatel 4400 R3.2 PBX with Cisco CallManager with
Analog FXS and FXO interfaces as an
MGCP Gateway

here it is explained how to use FXS and FXO interfaces to do integration.

Using * with TDM400P instead of a Cisco 3640 should be possible without big
problems, but...

...this is applicable in those cases where there are a small mumber of
lines to be routed
between the Alcatel PBX and Asterisk, and this will be the solution we will
try to set up for a pilot project.

If the requirement is to route an entire E1 trunk through *, having in any
case the E1 line coming out of the PBX,
is it possible to use some Digium hw (ie TE410P) to do the job?
In other words, which is the best way to integrate an * PBX into an
existing legacy environment?
I know it depends on the PBX, its features and its ability to be (further)
upgraded both in hw and sw...

A short term goal would be to have a number of branch offices (equipped
with IP phones) connected to a central Asterisk box through an IP WAN (ADSL
or MPLS), and have the asterisk box connected to the central PBX so that
any call directed to the root number of the company can be routed by the
PBX to * and then to the remote IP phones.

Then * could conquer also the typical domain of the legacy PBX, and then
...anyone know what else...


Francesco Pellegrini


++
|  Frame Srl |
|  Via Antonio Cantore 62/10 |
|  16149 Genova  |
|  Tel.   +39 010 8680570|
|  Fax.  +39 010 6591413 |
|  Cell.  +348 2237798   |
++





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Re: [Norton AntiSpam] [Asterisk-Users] Error on the Mysql, realtime database HELP so close so far; .

2005-04-25 Thread Ronald Wiplinger

Paul Dracevich wrote:
*CLI> Apr 26 16:08:04 ERROR[1882]: res_config_mysql.c:588 
mysql_reconnect: MySQL RealTime: Failed to connect database server 
asterisk on . Check debug for more info.

 

Regards
Paul Dracevich
Wireless Technology Consultant
Wayby Group
 

Mobile +64 29 638 9675
Phone +64 9 623 2143
Fax +64 9 623 1380
email [EMAIL PROTECTED]
website www.vnet.cc 


 

"the freedom to communicate is the right of every individual in the 
21st century" "Intellectual Property protection is the key to the 
Knowledge Economy" This email was sent to you via YOUtopia ... it's 
all about YOU.

 

The information contained in this email and any attachments is 
confidential and may be legally privileged. If you are not the 
intended recipient then you must not use, disseminate, distribute or 
copy any information contained in this email or any attachments. If 
you have received this email in error, please contact us immediately 
and delete this email. Thank you.

 


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Do you think that the message is in relation to the signature too short

bye
Ronald
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[Asterisk-Users] Error on the Mysql, realtime database HELP so close so far; .

2005-04-25 Thread Paul Dracevich









*CLI> Apr 26 16:08:04 ERROR[1882]: res_config_mysql.c:588 mysql_reconnect:
MySQL RealTime: Failed to
connect database server asterisk on . Check debug for more info.

 

Regards

Paul Dracevich

Wireless Technology Consultant

Wayby Group

 

Mobile +64 29 638 9675

Phone +64 9 623 2143

Fax +64 9 623 1380

email [EMAIL PROTECTED]

website www.vnet.cc

 

"the freedom to communicate is the right of every
individual in the 21st century" "Intellectual Property protection is
the key to the Knowledge Economy" This email was sent to you via YOUtopia
... it's all about YOU.

 

The information contained in this email and any attachments
is confidential and may be legally privileged. If you are not the intended
recipient then you must not use, disseminate, distribute or copy any
information contained in this email or any attachments. If you have received
this email in error, please contact us immediately and delete this email. Thank
you.

 






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Re: [Asterisk-Users] Large Asterisk Setup (~500 Concurrent Calls + Scalability)

2005-04-25 Thread Callum McGillivray

Matt Roth wrote:
I just wanted to make everyone aware that I cross-posted my original 
message to the Biz list.  You may want to check out the responses 
there, too.

It looks like the entire Asterisk slave server pool in my diagram 
(http://home.comcast.net/~mroth01/LargeAsteriskSetup.gif) can be 
replaced by a VoIP gateway.  So far that seems like a better solution 
from every perspective.
Matthew Roth
http://www.voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Debian 

Hey Matt,
What kind of VOIP gateway are you looking at implementing, tips would be 
appreciated.

Cheers,
Callum
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Re: [Asterisk-Users] Polycom ip500 (Not-Registered)

2005-04-25 Thread Sean Kennedy

Dan Morin wrote:
I just got a few Polycom IP500s and I’ve been following the info in 
the wiki trying to configure them. From what I can tell, they seem to 
be setup correctly (well…they don’t work so obviously not…) however, 
when they try to register with Asterisk, the following error shows up 
in the Logs:

Apr 25 15:22:05 NOTICE[1718]: Registration from '' failed for 
'192.168.0.222'
Apr 25 15:22:20 DEBUG[1718]: Auto destroying call 
'[EMAIL PROTECTED]'

Where 192.168.0.222 is the IP of the phone. The two single quotes seem 
to indicate that no credentials are being passed to * (?). If anyone 
has any experience with these, please let me know.

I can post the configs if that would help. Thanks in advance.
Dan
Please do. Specifically, sip.conf ( or whatever your sip configuration 
file for the phones is ), the individual phone settings and your 
sip.conf file from asterisk ( relevant parts only ).

Sean
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RE: [Asterisk-Users] Polycom IP4000 Conference Phone

2005-04-25 Thread Paul Hales




You need to have a very new 
firmware...


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley 
SilerSent: Tuesday, 26 April 2005 6:33 AMTo: Asterisk 
Users Mailing List - Non-Commercial DiscussionSubject: 
[Asterisk-Users] Polycom IP4000 Conference Phone

Can someone verify that this phone uses the same 
configs and sip.ld and other files as the IP 500 ? 
I jus tgot one and I cannot get it provisioned 
yet. 
Thanks, Wiley CAUTION: This email message and accompanying data 
may contain information that is confidential. If you are not the intended 
recipient, you are notified that any use, dissemination, distribution or copying 
of this message or data is prohibited. If you have received this email message 
in error, please notify us immediately and erase all copies of this message and 
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CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you.

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RE: [Asterisk-Users] Siemens SX66 wi-fi handset released

2005-04-25 Thread Dean Collins









How are you finding the speed/quality connecting
to your asterisk server when you are logging in from the average wifi hotspot?

 

Cheers,

Dean

 

 











From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin Ragsdale
Sent: Monday, April 25, 2005 10:33
PM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users]
Siemens SX66 wi-fi handset released



 

I've got one, and installed Xten's
softphone software.  It worked fine with a wired headset, but my bluetooth
headset won't work with it - a limitation of the OS, not Xten - so it's not
really useful for me until that (if it ever does) gets fixed.

 

Kevin

 







From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins
Sent: Monday, April 25, 2005 6:46
PM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Siemens
SX66 wi-fi handset released

http://www.pcmag.com/article2/0,1759,1787787,00.asp

 

I thought this new wifi handset may interest a few of you on
the list, anyone here actually seen or even better used on of these?

 

Cheers,

Dean

 








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RE: [Asterisk-Users] Siemens SX66 wi-fi handset released

2005-04-25 Thread Kevin Ragsdale




I've got one, and installed Xten's softphone 
software.  It worked fine with a wired headset, but my bluetooth headset 
won't work with it - a limitation of the OS, not Xten - so it's not really 
useful for me until that (if it ever does) gets fixed.
 
Kevin


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Dean 
CollinsSent: Monday, April 25, 2005 6:46 PMTo: Asterisk 
Users Mailing List - Non-Commercial DiscussionSubject: 
[Asterisk-Users] Siemens SX66 wi-fi handset released


http://www.pcmag.com/article2/0,1759,1787787,00.asp
 
I thought this new wifi handset may 
interest a few of you on the list, anyone here actually seen or even better used 
on of these?
 
Cheers,
Dean
 
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Re: [Asterisk-Users] Grandstream ATA 286 problems

2005-04-25 Thread seong

I having the same problem when getting the granstream budgetone 101 and
ATA 286/486 to use codec g.723. It just hanging there

And i notice other problem that grandstream devices always appeared as
offline in *, so i have to reboot these devices manually. I wonder are
they sending re-register msg to *? Almost everyday i encouter this
problem and which at least one will down.

Anton Krall wrote:

>Anobody had any problem with GS ata 286? The past few days Ive been having
>some problem with it, while making a call or during a call, I suddely hear a
>low noise like a car engine starting and then the ata dies, as if it got
>stuck or frozen.
>
>Anybody had these problems?
>
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>  
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RE: [Asterisk-Users] Call Recording via monitor

2005-04-25 Thread Chris Mason

> Unfortunately the otherwise excellent Areski stat tool doesn't seem
> to include the unique ID function and thus I can't pull a file back
> directly from that tool
> 
> Anyone fancy some development activity?

If it works out I will sponsor some development in this area, we need this
interface. If anyone else can chip in please let me know.

Chris Mason
US Number: (646)722-0001 US Fax (815)301-9759
Skype: netconcepts
 


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RE: [Asterisk-Users] Playback dosen't play Playtone(Congestion) does ?

Are you seeing anything in your /var/log/asterisk/messages file or even on
the console with verbosity at 3 or more? I'm guessing you have a path or
permissions problem, but you should see either in the logs or the console.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael D
Schelin
Sent: Monday, April 25, 2005 8:29 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Playback dosen't play Playtone(Congestion) does ?

Hi All, What would keep Asterisk from playing out audio files (Playback 
command) but I can play the busy tone . playtone(Congestion)  ??  I have 
verified this with ethereal and see the audio only going one way.  
Because I can hear the audio with the play tone I know there is 
something preventing the playback from working.

Thanks
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RE: [Asterisk-Users] Citel Handset Gateways

2005-04-25 Thread William Boehlke

We've installed them. They cost roughly $2600 for 24 ports through
distribution, and they do support the feature sets of the two phone types
we've installed (one being Meridian).  

William Boehlke
Signate
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeff Heath
Sent: Monday, April 25, 2005 6:35 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Citel Handset Gateways

Does anyone have any direct experience with these?

What do they cost per port?

Do they support most of the features of the original phone (i.e. if I have a
Meridian phone, do all the buttons like conference, flash, hold, etc. work
the same) ?

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[Asterisk-Users] Citel Handset Gateways

2005-04-25 Thread Jeff Heath

Does anyone have any direct experience with these?

What do they cost per port?

Do they support most of the features of the original phone (i.e. if I
have a Meridian phone, do all the buttons like conference, flash, hold,
etc. work the same) ?

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[Asterisk-Users] Playback dosen't play Playtone(Congestion) does ?

2005-04-25 Thread Michael D Schelin

Hi All, What would keep Asterisk from playing out audio files (Playback 
command) but I can play the busy tone . playtone(Congestion)  ??  I have 
verified this with ethereal and see the audio only going one way.  
Because I can hear the audio with the play tone I know there is 
something preventing the playback from working.

Thanks
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RE: [Asterisk-Users] Phone Recommendation.

2005-04-25 Thread Anton Krall

How? You mean if you use [EMAIL PROTECTED] right? 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Wiley Siler
|Sent: Lunes, 25 de Abril de 2005 02:50 p.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: RE: [Asterisk-Users] Phone Recommendation.
|
|Call waiting can be disabled in Asterisk via *71 regardless of 
|the phone used.
|
|Cheers,
|Wiley
|  
|
|
|
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of Sean A.
|Newton
|Sent: Monday, April 25, 2005 11:56 AM
|To: asterisk-users@lists.digium.com
|Subject: [Asterisk-Users] Phone Recommendation.
|
|
|I'm looking for recommendations for a office phone that has 
|the ability to disable call-waiting.
|
|Needs to be similar in features to a Polycom IP300. 
|
|Thanks,
|
|--Sean
|
|-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
| sean a. newton  [EMAIL PROTECTED]
| louisville, ky, usa http://wewt.net 
|
| Another day, another convertible and another hotel 
| full of cops.-- Hunter S. Thompson
|-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
|
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|

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Re: [Asterisk-Users] Broadvoice Down?

2005-04-25 Thread Mailing List

If I dial into my number I get nothing but dead air and then it hangs up. the really odd thing is the call makes it to my sip phone 
but it's just dead air if I answer. The bad thing for me is my outbound does not work as everything times out. Only call I can 
reliably place is to their support number but no matter which option I pick I get sent right to a busy signal. This is all so 
lovely...

- Original Message - 
From: "Max Clark" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
Sent: Monday, April 25, 2005 1:08 PM
Subject: [Asterisk-Users] Broadvoice Down?

Is anyone else having difficulty with their Broadvoice service? When I dial my number right now it rings either fast busy or tells 
me it cannot complete the call.

I can make outgoing calls from my system through broadvoice however. Seems 
their inbound trunks hit capacity?
Am I alone in this?
-Max
--
  Max Clark
  max [at] clarksys.com
  http://www.clarksys.com
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[Asterisk-Users] Asterisk ADSI

2005-04-25 Thread John Bittner

Hi,
Anyone know if you can use a ADSI phone with Asterisk behind
an Linksys ATA. 
I know packet 8 uses these phones with an ATA. I tried
testing it but when you try to program the phone
you get a error ADSI Unavailable on CPE.
Do I have to program the phone with a zaptel card first
before using it with an ATA?

I looked all over the net for some info but not much on
ADSI. 

Any help is appreciated.

Thanks

John Bittner
Simlab.net



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RE: [Asterisk-Users] Cisco's description of echo

2005-04-25 Thread Alexander Lopez










Don’t; forget the Milliwatt
application in Asterisk

 

 

-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Keith O'Brien
Sent: Monday, April 25, 2005 4:28
PM
To: [EMAIL PROTECTED];
Asterisk-Users@lists.digium.com
Subject: RE: [Asterisk-Users]
Cisco's description of echo

 



If you are running a Cisco VoIP gateway you can send a
0dBm 1000Khz test tone into or out of a voice port with:





 





"test voice
port  inject-tone network 1000hz"





 





to measure the tone do a:





 





"sh call active voice brief"





 





>>Another common problem that causes echo in
networks is not setting your
>>loss plan correctly.    You need to be sure that you
aren't coming in too
>>hot at any of your analog interfaces.   In general you should
see a signal
>>between -20dbm and -12dbm when someone is talking on the
line.   If it is
>>significantly hotter then you run the chance of having a larger
reflected
>>signal resulting in echo.   I typically try padding down
analog levels by
>>3dB at a time to see if echo is reduced.   
>How do you measure the amplitude of a pstn line? As an audio engineer in a
>previous life, I would love to be able to send standard level tones down a
>pstn line and measure the amplitude at my end, then adjust the input gain
>accurately. Is there a way to do this?








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[Asterisk-Users] stanaphone now terminating fax

2005-04-25 Thread Dean Collins









I just got an email from Stanaphone saying that any fax
calls terminating on your Stanaphone number will now be converted to PDF and
then forwarded to your email address J

 

Pretty cool considering they don’t charge extra for
this feature.

 

Cheers,

Dean

 






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RE: [Asterisk-Users] voip problems

2005-04-25 Thread William Boehlke




We advise customers to use VoIP only for internal calls and where outbound
is expensive but sound quality is not mission critical. Easy to configure
the switch to fail over to the pstn or another provider during the day the
provider is down every month. I look forward to the day the service
providers get to five eights. 

William Boehlke
Signate

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of JD Austin
Sent: Monday, April 25, 2005 3:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] voip problems



Kerry Garrison wrote:

>The quick and dirty method is to have the main lines come in over PSTN 
>lines, this can then be call-forwarded to a VOIP provider. This makes 
>it simple to change VOIP providers on-the-fly as well as turn off 
>forwarding and let the PSTN line come into the PBX. Setting up failover 
>on outbound calls through multiple providers is relativly simple but 
>keeping the main business number pointing to whatever service you are 
>using is the tricky part.
>-Kerry
> 
>
>  
>
Doesn't that limit you to a single inbound call at a time?

>-Original Message-
>From: [EMAIL PROTECTED]
>[mailto:[EMAIL PROTECTED] On Behalf Of Gerard 
>Marcel
>Sent: Monday, April 25, 2005 1:04 PM
>To: asterisk-users@lists.digium.com
>Subject: [Asterisk-Users] voip problems
>
>How do you guys deal with voip problems?  do you have multiple backups 
>such as land lines, and different voip providers?
>
>Regards,
>
>GM
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>

--
JD Austin
Twin Geckos Technology Services LLC
email: [EMAIL PROTECTED]
http://www.twingeckos.com
phone/fax: 480.344.2640 

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RE: [Asterisk-Users] Alternatives to SpanDSP??

2005-04-25 Thread Adam Goryachev

On Mon, 2005-04-25 at 17:07 -0400, Jeremy Melanson wrote:
> More like, I already have enough Digium cards, and I don't want purchase
> a bunch of fax/modems and more Digium cards than I alrady have.
> I have a PRI line that I'd like to support high-volume faxing on. I've
> gotten SpanDSP to work with * over the PRI, but I need a more
> reliability.
> That, and I guess I'm probably just being cheap too :-)

OK, now you are finally making sense (to me :)

Why not just (yeah, just) debug the issues you see, and try to have them
fixed. From what I have seen on the list, most people seem to be able to
use spandsp for faxing without a problem.

It would be interesting to hear from someone regularly receive more than
say 100 faxes per day from 'random' fax machines using spandsp? or
better, some other large number per day :)

Regards,
Adam

-- 
 -- 
Adam Goryachev
Website Managers
Ph:  +61 2 8304 [EMAIL PROTECTED]
Fax: +61 2 9345 4396www.websitemanagers.com.au

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[Asterisk-Users] Polycom ip500 (Not-Registered)

2005-04-25 Thread Dan Morin









I just got a few Polycom IP500s and I’ve been
following the info in the wiki trying to configure them.  From what I can
tell, they seem to be setup correctly (well…they don’t work so
obviously not…) however, when they try to register with Asterisk, the following
error shows up in the Logs:

 

Apr 25 15:22:05 NOTICE[1718]: Registration from ''
failed for '192.168.0.222'
Apr 25 15:22:20 DEBUG[1718]: Auto destroying call
'[EMAIL PROTECTED]'

 

Where 192.168.0.222 is the IP of the phone.  The two
single quotes seem to indicate that no credentials are being passed to *
(?).  If anyone has any experience with these, please let me know.

 

I can post the configs if that would help.  Thanks in
advance.

Dan

 






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[Asterisk-Users] Siemens SX66 wi-fi handset released

2005-04-25 Thread Dean Collins









http://www.pcmag.com/article2/0,1759,1787787,00.asp

 

I thought this new wifi handset may interest a few of you on
the list, anyone here actually seen or even better used on of these?

 

Cheers,

Dean

 






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Re: [Asterisk-Users] T1 E&M wink issues - bad int'l dial-outandoccasional dropped calls

2005-04-25 Thread bill black

OK.  The latest debug from Qwest is that our 'Nature of Address' field 
is not being set properly for international calls.  He said that this is 
something specified by our switch but could not give details and it is 
not simply keying 011.  With E&M wink, what else is there besides for 
DTMF?  How do we specifiy a Nature of Address field with E&M wink?

Thanks,  Bill
jltaylor wrote:
OK, so you are either D4/ami or B8ZS/esf, and you're doing E&M wink.
And, you have a T1 cable from the Cisco to the Asterisk.
And, the Cisco is set for channelized T1, 8 channels using RBI.
In your zapata.conf you have chosen: signalling=em_w.
Can you get someone at Qwest to watch while your dial?
If they know what trunk you are on then they can capture it.
Have you turned SIP or IAX2 debug on and examined any messages while
dialing?
So do you have to setup the dial string: 1NXXNXX ?
Have you asked Qwest if they support any other type of signaling (Feature
Group?).
I had a problem with ISDN BRI going to a Terminal Adapter and splitting to
to analog lines.
Lines would "hang" off hook.
Calls would drop.
I changed to two analog business lines and the problems stopped.
I think that we were not getting proper CPC.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of bill black
Sent: Sunday, April 24, 2005 3:00 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] T1 E&M wink issues - bad int'l
dial-outandoccasional dropped calls
Customer has integrated access arrangement with 16 channels of data/8
for voice that is split via customer cisco equipment.  No local dialing,
LD and incoming 800 service only via the t1.  Qwest provides both the
local loop and LD/800 service but it is provided via re-seller PNG.
We have verified clock integrity via cisco logs that show no frame slips
(cisco uses CO as a reference and we use the cisco as a reference.)
No reference to Feature Group A (or D for that matter) is on our paperwork.
It is nearly a rural location so I'm guessing we are connected via an
End Office but can check this.
Bill
jltaylor wrote:
 

What kind of service did you subscribe to (what do they call it on your
bill)?
Retail business trunks?
Feature Group A?
Can you dial "local" numbers or is this all long distance?
Is Qwest the LEC or a long distance provider for this service?
Are you connected to an End Office or a Tandem?
These all may give me a hit as to what is going on.
James
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of bill black
Sent: Sunday, April 24, 2005 10:45 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] T1 E&M wink issues - bad int'l dial-out
andoccasional dropped calls
Anyone have any ideas here?
We are using 8 channels of E&M Wink with a T100P for outgoing LD and
incoming tollfree numbers and are apparently connected to a Nortel
DMS-250 at the CO.   We are receiving ANI & DNIS just fine and can
dial-out domestically with DTMF but have two issues that are still
unresolved:
1) We cannot dial-out internationally with an 011 prefix (or any other
prefix that we can think of).  Qwest claims (1) they never get
international calls and (2) domestic calls are routed to their LD
service as 1NXXNXX instead of 1NXXNXX.   Is some form of
prefix/suffix needed for DTMF dialing over an E&M wink channel?  (e.g.
something like the *ANI*DNIS* for incoming.) 011 clearly doesn't work as
a prefix and Qwest's response has invariably been 'there is something
wrong with your PBX' :(   Curiously if we follow an
011+international-number with a * we get a recording that we have not
entered sufficient digits to complete the call whereas without the * we
just get a congestion beep from the far end.
2) Once or twice a day the customer is getting calls dropped.  The log
shows the following:
Apr 21 13:15:51 VERBOSE[22664]: -- Hungup 'Zap/7-1'
Apr 21 13:15:52 VERBOSE[22664]: -- Starting simple switch on
'Zap/7-1'  Apr 21 13:15:53 WARNING[22664]: getdtmf on channel 7:
Operation now in progress
Apr 21 13:15:53 VERBOSE[22664]: -- Hungup 'Zap/7-1'
It appears that we see the line go back on-hook, hangup but then see it go
off-hook again and treat it as another incoming call that never gets a DTMF
input when in fact the call has just been dropped.  We've verified that we
are not sharing interrupts, we are on run level 3 etc. zttest shows (so
   

far)
 

a minimum of 99.987%.  Can anyone think of what might be causing this or
what we could ask Qwest regarding possible diagnostics?
3) Finally, what level of dropped calls is generally considered acceptable?
Like the dead-pixel issue with LCDs this is pretty subjective but is there
an industry number that is typical? (We are presently at ~1% due to this
issue.)
Thanks to all for any shared wisdom.  Bill


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[Asterisk-Users] Transfers tend to fail after upgrade to 1.0.7

2005-04-25 Thread Pablo Alsina

Hi

We have a fairly simple Asterisk setup: E1 card, 8 FXO lines connected
to old PBX, and some SIP phones, used by a callcenter with queues.
Almost all calls are incoming (through E1 line), answered by some
callcenter operator (using SIP phones, call assigned by queue app),
and in some cases, are transferred to some other extension on the old
PBX or other SIP.

We had problems with Music on Hold (on the queue) and with transfers
on version 1.0.3. We now upgraded to 1.0.7 and the MoH problem is
gone, but we still have some transfer problems.

What happens is that sometimes when one callcenter op (SIP client)
does a transfer to another SIP or an extension that is mapped to a FXO
line (old PBX), we get a half-call: the caller hears the called
station, but the called station (the one the call is transferred to)
does not here the caller.

As we need attended transfer, the calls are made from the SIP phone
(Xten), using the transfer button (not blind transfers).

Don't really know how to debug this. Is there a log I can see that can
help me pinpoint the problem?. On that log, what should we be looking
for? I'm used to debug this kind of problems in general, but are not
familiar with SIP protocol nor Asterisk debugging.

We tried to change SIP phones, but its the same. Note that it happens
with calls that have one end on the E1 and the other to FXO, both
local to Asterisk ("joined" by a SIP phone), so it does not seems to
be a codec problem.

Thanks for any advice,
Pablo

PD: I sent this mail some days ago, but didn't see it on the list. My
apologies if it is a dupe.
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RE: [Asterisk-Users] Broadvoice Down?

2005-04-25 Thread trixter http://www.0xdecafbad.com

Whatever is going on is a lot longer than today only.  Last night caller
id was 'odd' (outbound calls to a different provider showed as really
weird numbers).  Broadvoice normally terminates calls after 4 hours they
did not last night.  Etc.

I wonder if this is because they are installing CALEA support, which is
now required in the United States for VoIP providers that connect to the
PSTN (CALEA is the set of laws that forces telcos to have wiretap
equipment, failure to install such equipment is a $10,000 per day fine,
granted the equipment is only *supposed* to be turned on when a valid
warrant is presented but hey).


-- 
Trixter http://www.0xdecafbad.com
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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RE: [Asterisk-Users] Grandstream ATA 286 problems

2005-04-25 Thread Gregory Wiktor - ADCom Corp.

Maybe you just got a broken one?

If you need high end, the multitechs are good products, with excellent
support.

Greg 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dana Olson
Sent: Monday, April 25, 2005 2:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Grandstream ATA 286 problems

I've had an issue with my 286 ever since I got it. Basically, the web
interface doesn't load, and I can't make any calls - although I get
dialtone. Also, I can call it and it will ring. But I get no audio.
The main issue is that I can't get into the web interface anymore... I
did once, but not anymore.

I contacted the vendor I bought it from, and they said to contact
Grandstream.

I contacted Grandstream, and they told me to hit refresh in my
browser

After sending them the Ethereal trace, I haven't heard back from them
yet.

I think it's the worst purchase I've ever made.

On 4/25/05, Anton Krall <[EMAIL PROTECTED]> wrote:
> Anobody had any problem with GS ata 286? The past few days Ive been 
> having some problem with it, while making a call or during a call, I 
> suddely hear a low noise like a car engine starting and then the ata 
> dies, as if it got stuck or frozen.
> 
> Anybody had these problems?
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Re: [Asterisk-Users] voip problems


Kerry Garrison wrote:
The quick and dirty method is to have the main lines come in over PSTN
lines, this can then be call-forwarded to a VOIP provider. This makes it
simple to change VOIP providers on-the-fly as well as turn off forwarding
and let the PSTN line come into the PBX. Setting up failover on outbound
calls through multiple providers is relativly simple but keeping the main
business number pointing to whatever service you are using is the tricky
part.
-Kerry
 

Doesn't that limit you to a single inbound call at a time?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gerard Marcel
Sent: Monday, April 25, 2005 1:04 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] voip problems
How do you guys deal with voip problems?  do you have multiple backups such
as land lines, and different voip providers?
Regards,
GM
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--
JD Austin
Twin Geckos Technology Services LLC
email: [EMAIL PROTECTED]
http://www.twingeckos.com
phone/fax: 480.344.2640 

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RE: [Asterisk-Users] Broadvoice Down?

2005-04-25 Thread Gregory Wiktor - ADCom Corp.

I am happy with teliax...

Keep in mind though, I am using 800 from opex, who can do an instant
change to another number. That goes to a local did.  This way, if
something happens and I am very busy, I can just call opex and they can
switch it.

Its like 2.9 cents per minute or so, on top of the did usage, but still
not bad.

Greg 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
Adamson
Sent: Monday, April 25, 2005 6:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Broadvoice Down?

Sounds like either BV or their pstn supplier did an upgrade last night
of some sort. (That's just a guess on my part. Certainly would not have
taken this long to fix a problem with that wide of a scope if it was
simply a so called bug. Or, maybe they forgot to pay their bill.)



> I guess 2 hours is 'soon' to them.  I'm still down.
> Is there a reliable voip provider out there that works with Asterisk?
> I can't have downtime like this.. it just makes my company look bad.
> 
> JD
> 
> JD Austin wrote:
> 
> I tried calling Broadvoice support.. on hold for 1/2 hour then it 
> hung up on me with a
reorder (fast busy), so I tried again.
> Just got through to a rep- they said it's a 'carrier issue' that
their 'partner carrier' 
was having issues and that it would be up
> soon.
> Makes me wonder if I should be signing up with their 'partner
carrier' instead.
> 
> JD
> 
> Jerry Geis wrote:
> 
> I am having the same broadvoice issue at the moment.
> 
> jerry
> 
> Is anyone else having difficulty with their Broadvoice
service? When I 
> dial my number right now it rings either fast busy or tells me
it cannot 
> complete the call.
> 
> I can make outgoing calls from my system through broadvoice
however. 
> Seems their inbound trunks hit capacity?
> 
> Am I alone in this?
> -Max
> 
> -- 
>Max Clark
>max [at] clarksys.com
>http://www.clarksys.com
> 
>  
> --
> 
> 
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> -- 
> JD Austin
> Twin Geckos Technology Services LLC
> email: [EMAIL PROTECTED]
> http://www.twingeckos.com
> phone/fax: 480.344.2640
> 
>  
> --
> 
> 
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> --
> JD Austin
> Twin Geckos Technology Services LLC
> email: [EMAIL PROTECTED]
> http://www.twingeckos.com
> phone/fax: 480.344.2640
---End of Original Message-


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RE: [Asterisk-Users] Broadvoice Down?




Inward or outward?
 
Outward, I am having the best luck with my dialing via 
VoipJet.
 
Inward, I have nto stabalized any one service yet so I am 
still open to suggestion.
 
Thanks,
Wiley
 


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of JD 
AustinSent: Monday, April 25, 2005 2:08 PMTo: Asterisk 
Users Mailing List - Non-Commercial DiscussionSubject: Re: 
[Asterisk-Users] Broadvoice Down?


On Mon, 2005-04-25 at 12:48, JD Austin wrote:
  
  I guess 2 hours is 'soon' to them.  I'm still down.
Is there a reliable voip provider out there that works with Asterisk?
I can't have downtime like this.. it just makes my company look bad.

JD

I just came up.. 4 hours down 
time.There are a few changes on broadvoices' web site: http://www.broadvoice.com/support_install_asterisk.htmlI 
made the changes, but they didn't immedately work so I don't think that was 
it.What does 'pedantic=no' mean in sip.conf?JD

  JD Austin wrote: 

I tried calling Broadvoice support.. on hold for 1/2 hour then it
hung up on me with a reorder (fast busy), so I tried again.
Just got through to a rep- they said it's a 'carrier issue' that
their 'partner carrier' was having issues and that it would be up
soon.
Makes me wonder if I should be signing up with their 'partner
carrier' instead.

JD

Jerry Geis wrote: 
  
  I am having the same broadvoice issue at the moment.

jerry
Is anyone else having difficulty with their Broadvoice service? When I 
dial my number right now it rings either fast busy or tells me it cannot 
complete the call.

I can make outgoing calls from my system through broadvoice however. 
Seems their inbound trunks hit capacity?

Am I alone in this?
-Max

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JD Austin
Twin Geckos Technology Services LLC
email: [EMAIL PROTECTED]
http://www.twingeckos.com
phone/fax: 480.344.2640 
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Re: [Asterisk-Users] Has anyone used Libretel DIDs with Asterisk?

2005-04-25 Thread William Suffill

FWD is availabe via iax as well as sip. Easiest solution would be to
sign up for a FWD acc and enable iax. You could even use sip plenty of
examples between the list and voip-info.org that should get you going.
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[Asterisk-Users] using goto to do selective dialing

2005-04-25 Thread David John Walsh

If I set up a context, which has lots of extensions eg

[ext-local]
exten => 3000,1,Dial(SIP/3000)
exten => 3001,1,Dial(SIP/3001)

.
exten => 3999,1,Dial(SIP/3999)

(I know the syntax is wrong, and it probably is not the best way to achive it)

then in another context, I use a goto like so

[selected-3000-numbers]
exten => _32XX,1,Goto(ext-local,${EXTEN},1)
exten => _34XX,1,Goto(ext-local,${EXTEN},1)

Will this allow me to only dial (from a phone in
selected-3000-numbers) the numbers starting 32 and 34??

Also is my "goto" syntax right?

Thanks for your input
David
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Re: [Asterisk-Users] Has anyone used Libretel DIDs with Asterisk?

2005-04-25 Thread Mark Charlton

On 4/25/05, beonice <[EMAIL PROTECTED]> wrote:
> Hello, all.
> 
> Google searches reveal that a couple of users have
> used Libretel port-of-call service via FWD using
> Asterisk. I don't have a FWD account. I don't have a
> SIP phone (no SIP setup at all, in fact).
> 
> I just purchased a DID in Brazil from Libretel and
> discovered that they assume that I either have a SIP
> phone or a VOIP number through BroadVoice. My provider
> is actually VoicepulseConnect in California. My server
> here is an Asterisk box connecting to a Voicepulse DID
> using IAX.
> 
> HELP. :( I was assuming everyone supports IAX. I guess
> I should have made sure first, huh?
> 
> Any help would be much appreciated.
> 
> Thanks,
> Maya
> 

Just set up a FWD account, (they're free), enable the IAX account
option, and you are away.  I have it set up and seems to be working
fine, not that it gets much use. Or just use the sip part of asterisk.

Regards
Mark
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Re: [Asterisk-Users] Broadvoice Down?

2005-04-25 Thread William Suffill

Seems what we all want but since it's new there are always problems
especially since we as a whole complain when they charge too much.
There will be a happy medium eventually but for now it's probably best
 not to having too important dependent on voip origination since
unlike termination you can't just use whoever it working properly at
the current time.

Atleast for me I've found giving out my cell # and doing all call
backs over voip as been the safest bet for now until voip origination
improves.
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Re: [Asterisk-Users] astrecipes v2.0

2005-04-25 Thread Daniel Bruce Lynes

On Monday 25 April 2005 08:07 am, lenz wrote:

> if anyone is interested, there is a new wiki about Asterisk "recipes",
> i.e. step-by-step descriptions on how to perform something with your *
> box. This is quite different from most * sites around, that are either
> questions-and-answers forums or are dedicated to documenting a feature.
> The point of AstRecipes is how to implement something.
>
> See http://www.oinko.net/astrecipes

Good idea.  Something like this is a good starting point for those of us that 
are just starting out, or other more advanced users that are just looking to 
add something new to their asterisk box.
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Re: [Asterisk-Users] Dial Plan - How to prepend a digit

2005-04-25 Thread Stiffe

Do you mean like this?

This is an example
Here I have 3 trunks (2 SIP and 1 IAX)

To dial out in rix, i'll have to dial a 0 before areaprefix, which
here always begins with another 0 (040xx)
 I.e  to dial that area on trunk "rix"  I'll have to dial 0040xxx = (00Z.)
EXTEN:1 will remove one (1) leading digit, the first 0 and I'm out on
that trunk.

Same with the digisip-trunk, instead I'll have to dial 9 to reach out
on that trunk
Same with the FWD-trunk, I'll have to dial 393 to reach the FWD network.
EXTEN:3 will remove the leading three digits (393)

To be able to dial another country I'll have to be able to dial 00 (000Z.)
Aka 0 00x  or 9 00x
´
I'm using [EMAIL PROTECTED] ver 0.9 with asterisk 1.0.7


;[outrt-001-rixroute]
;include => outrt-001-rixroute-custom
;exten => _000Z.,1,Macro(dialout-trunk,2,${EXTEN:1})
;exten => _000Z.,2,Macro(outisbusy) ; No available circuits
;exten => _00Z.,1,Macro(dialout-trunk,2,${EXTEN:1})
;exten => _00Z.,2,Macro(outisbusy)  ; No available circuits
;
;[outrt-002-digisiproute]
;include => outrt-002-digisiproute-custom
;exten => _900Z.,1,Macro(dialout-trunk,1,${EXTEN:1})
;exten => _900Z.,2,Macro(outisbusy) ; No available circuits
;exten => _90Z.,1,Macro(dialout-trunk,1,${EXTEN:1})
;exten => _90Z.,2,Macro(outisbusy)  ; No available circuits
;
;[outrt-003-fwdroute]
;include => outrt-003-fwdroute-custom
;exten => _393.,1,Macro(dialout-trunk,3,${EXTEN:3})
;exten => _393.,2,Macro(outisbusy)  ; No available circuits

Rgds

Stefan (Sweden)

On 4/25/05, Daniel Salama <[EMAIL PROTECTED]> wrote:
> I'd like to create a dial rule that when someone tries to dial a
> particular number, the same number is dialed, except that prefixed with
> some additional digit(s). How can this be specified on extensions.conf?
> 
> Thanks,
> Daniel
> 
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Re: [Asterisk-Users] Broadvoice Down?






Kerry Garrison wrote:

  
  
  
  While we have only been using
Broadvoice for a few months now, we have actually had better service
through them than with our PSTN provider. You could just as easily have
had a voice T1 go down which typically takes a few hours to replace
(and may be the actual problem). The issue is not with Broadvoice as a
service as much as it seems to be a peering problem. 
   
  Yes these issues make you and
your company look bad in so much as it shows that you do not have a
properly designed redundant system. Do you back up your server every
night even though you only lose a file once a year? So why wouldn't you
have a failover on your phone system?
   
  Its very easy to blame
Broadvoice for your phone service being out (and I am not defending
them) but that is ONE connection. Any business that is highly reliant
on their phones for business should have a backup system just for this
reason. Possibly even multiple backups depending on the critical nature
of the business.
   
  I have seen businesses
completely lose telephone service on standard PSTN lines for a day or
more at a time. So sh** happens, be prepared. Dont put all your eggs in
one basket. This is a perfect example of why.
   
  Kerry Garrison
  http://techdatapros.com
   

I do have failover; I am using another service that is not voip based. 

Fortunately only a few clients know about the new BV number.
This makes me reluctant to switch over completely.

How exactly could you failover using such a provider?
Outbound calling failover is easy; just have another provider and set
them up as a trunk.  
Inbound which relies on them and the DID provided to work is a bit more
complicated.
Sure you could do the same, but other than 'call this other number if
this number is down' how do you failover?  
The only way I can think of is get a regular pots line in and have it
forward on busy to BV and have both answer in asterisk.

Are DID's portable?

JD
-- 
JD Austin
Twin Geckos Technology Services LLC
email: [EMAIL PROTECTED]
http://www.twingeckos.com
phone/fax: 480.422.1250


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Re: [Asterisk-Users] Dial Plan - How to prepend a digit

2005-04-25 Thread Daniel Salama

Franco/Tony/Josiah,
Thanks for the feedback. That did it.
- Daniel
On Apr 25, 2005, at 3:53 PM, Franco Bellagamba wrote:
Daniel, try
_X.,1,Dial(123${EXTEN})
That will prefix "123" to the dialed number.
Franco
- Original Message -
From: "Daniel Salama" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Monday, April 25, 2005 4:05 PM
Subject: [Asterisk-Users] Dial Plan - How to prepend a digit

I'd like to create a dial rule that when someone tries to dial a
particular number, the same number is dialed, except that prefixed 
with
some additional digit(s). How can this be specified on 
extensions.conf?

Thanks,
Daniel
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RE: [Asterisk-Users] voip problems

Lower dial cost, cheaper redundancy, cheaper provisiojing (no $600 T1
Required)

W 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gerard
Marcel
Sent: Monday, April 25, 2005 1:46 PM
To: Wiley Siler
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] voip problems

In that case, where is the savings on telephone bills?


Regards,


GM



On 4/25/05, Wiley Siler <[EMAIL PROTECTED]> wrote:
> Seems to be the norm in most cases
> 
> I dial out on one ITSP who seems to always have good dialing with no 
> issues...
> 
> I receive on another that gives me DIDs and 800s (still working out 
> kinks)...
> 
> W
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Gerard 
> Marcel
> Sent: Monday, April 25, 2005 1:04 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] voip problems
> 
> How do you guys deal with voip problems?  do you have multiple backups

> such as land lines, and different voip providers?
> 
> Regards,
> 
> GM
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[Asterisk-Users] Cannot make outgoing calls on Mediatrix 1204 from Asterisk

2005-04-25 Thread bam

Mediatrix has been setup with automatic calling enabled as suggested
elsewhere with the four ports forwarding calls to extensions 1001, 1002,
1003, & 1004 respectively. 

Inbound traffic pretty much does what is expected albiet it takes a few
rings and some warnings before the call is passed to Asterisk.

Outbound calls head off to the 1204 before they loop back to asterisk
appearing as an imbound call which is not really what is expected.
Automatic calling is supposed to be inbound only. 

Any ideas?

The trace looks like this:

-- Executing Dial("SIP/212-acc5", "SIP/[EMAIL PROTECTED]") in new
stack
-- Called [EMAIL PROTECTED]
-- Executing Dial("SIP/1001-7f99", "SIP/210|20|tr") in new stack
-- Called 210
-- SIP/210-b655 is ringing


extensions.conf 

[BT_PSTN] ; Inbound calls

exten => _100X,1,Dial(SIP/210,20,tr)

[LOCAL_SIP] ; All internal extensions

exten => _0.,1,Dial(SIP/[EMAIL PROTECTED])


Mediatrix is version 4.4.13.88

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Re: [Asterisk-Users] voip problems

2005-04-25 Thread Rich Adamson

> How do you guys deal with voip problems?  do you have multiple backups
> such as land lines, and different voip providers?

Given the number of BV posts over the last several months, it would
certainly appear stability is not in their vocabulary. (I'm a past
user, but not any more.)

Guess I'd have to question why someone would commit a business to
using an "unsupported" service that is somewhat unstable, has limited
codec support, and that relies on the asterisk-user list for support.

Someone else already mentioned the need for backup arrangements.

What do I use? livevoip.com for 800 number, teliax.com for did's,
and, pstn analog's for backup and some outgoing calls. I can guarantee
not all three will be down at the same time (except for the local
fiber seeking backhoe). :)


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Re: [Asterisk-Users] Broadvoice Down?







  

On Mon, 2005-04-25 at 12:48, JD Austin wrote:
  
  
I guess 2 hours is 'soon' to them.  I'm still down.
Is there a reliable voip provider out there that works with Asterisk?
I can't have downtime like this.. it just makes my company look bad.

JD


  

I just came up.. 4 hours down time.
There are a few changes on broadvoices' web site: 
http://www.broadvoice.com/support_install_asterisk.html
I made the changes, but they didn't immedately work so I don't think
that was it.

What does 'pedantic=no' mean in sip.conf?


JD

  
JD Austin wrote: 


  I tried calling Broadvoice support.. on hold for 1/2 hour then it
hung up on me with a reorder (fast busy), so I tried again.
Just got through to a rep- they said it's a 'carrier issue' that
their 'partner carrier' was having issues and that it would be up
soon.
Makes me wonder if I should be signing up with their 'partner
carrier' instead.

JD

Jerry Geis wrote: 
  
  
I am having the same broadvoice issue at the moment.

jerry
Is anyone else having difficulty with their Broadvoice service? When I 
dial my number right now it rings either fast busy or tells me it cannot 
complete the call.

I can make outgoing calls from my system through broadvoice however. 
Seems their inbound trunks hit capacity?

Am I alone in this?
-Max



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-- 
JD Austin
Twin Geckos Technology Services LLC
email: [EMAIL PROTECTED]
http://www.twingeckos.com
phone/fax: 480.344.2640 


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RE: [Asterisk-Users] Broadvoice Down?

Who says on their site they are not taking new customers...

Bummer too, because I wam looking for a new set of DIDs from a reliable
source.

I was hoping they would be the ones

W
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Derek
Whitten
Sent: Monday, April 25, 2005 1:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Broadvoice Down?

I have had good luck with nufone...  

http://www.nufone.net

http://www.nufone.net/rates.csv


On Mon, 2005-04-25 at 12:48, JD Austin wrote:
> I guess 2 hours is 'soon' to them.  I'm still down.
> Is there a reliable voip provider out there that works with Asterisk?
> I can't have downtime like this.. it just makes my company look bad.
> 
> JD
> 
> JD Austin wrote: 
> > I tried calling Broadvoice support.. on hold for 1/2 hour then it 
> > hung up on me with a reorder (fast busy), so I tried again.
> > Just got through to a rep- they said it's a 'carrier issue' that 
> > their 'partner carrier' was having issues and that it would be up 
> > soon.
> > Makes me wonder if I should be signing up with their 'partner 
> > carrier' instead.
> > 
> > JD
> > 
> > Jerry Geis wrote: 
> > > I am having the same broadvoice issue at the moment.
> > > 
> > > jerry
> > > Is anyone else having difficulty with their Broadvoice service? 
> > > When I dial my number right now it rings either fast busy or tells

> > > me it cannot complete the call.
> > > 
> > > I can make outgoing calls from my system through broadvoice
however. 
> > > Seems their inbound trunks hit capacity?
> > > 
> > > Am I alone in this?
> > > -Max
--
Derek Whitten <[EMAIL PROTECTED]>
kFuQ Productions
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RE: [Asterisk-Users] Playing mp3's while recording voicemail

2005-04-25 Thread Rafal Kaniewski

apologies for the html of the last mail.. outlook at work...

Rafal Kaniewski

-- 
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.308 / Virus Database: 266.10.2 - Release Date: 21/04/2005
 

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[Asterisk-Users] Asterisk replacing CCM using Catalyst 6608

2005-04-25 Thread Aaron Daniel

Hey,
I'm currently developing asterisk for use on my university's system, and 
we're in the process of hooking it up to the main network to allow for 
on/off-campus calling.  Does anyone know of a way, through ip, of having 
the server communicate directly to a cisco catalyst 6608 blade to make 
calls that aren't on the server?  We would go through the CCM, but we're 
trying to get rid of the cisco server altogether, and we don't really 
want to bother with T1 cards (at least, not at this time).

Aaron Daniel
Senior Voice Analyst
Sam Houston State University
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[Asterisk-Users] Dialing to a remote extension

2005-04-25 Thread Robert G. Ristroph


Hi,

I am trying to automatically call out to an extension on another
PBX, which means that when asterisk dials it should get an answer, and
then dial an extension digit after a short wait.  I tried using the D(digits)
option to the Dial() application, and I also tried several other methods from
the wiki and google searches.  I cam not get it to dial the extension after
connecting.

For example, I made a call file with this in it:

Application: Dial
Data: Zap/1/5551212,D(103)

 And I also tried this:

Application: Dial
Data: Zap/1/5551212BYEXTENSION/www103

How can I accomplish this ?  People must do it pretty often,
especially for routing through pre-paid calling cards and stuff like
that.

Thanks in advance,

--Rob

-- 
Robert G. Ristroph
Airlink Systems
[EMAIL PROTECTED]
(512) 231-1240 x103





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[Asterisk-Users] Asterisk integration with Alcatel 4400

2005-04-25 Thread asterisk

I have just finished to look at the interesting documentation linked from
voip-info.org entitled:

PBX Interoperability: Alcatel 4400 R3.2 PBX with Cisco CallManager with
Analog FXS and FXO interfaces as an
MGCP Gateway

here it is explained how to use FXS and FXO interfaces to do integration.

Using * with TDM400P instead of a Cisco 3640 should be possible without big
problems, but...

...this is applicable in those cases where there are a small mumber of
lines to be routed
between the Alcatel PBX and Asterisk, and this will be the solution we will
try to set up for a pilot project.

If the requirement is to route an entire E1 trunk through *, having in any
case the E1 line coming out of the PBX,
is it possible to use some Digium hw (ie TE410P) to do the job?
In other words, which is the best way to integrate an * PBX into an
existing legacy environment?
I know it depends on the PBX, its features and its ability to be (further)
upgraded both in hw and sw...

A short term goal would be to have a number of branch offices (equipped
with IP phones) connected to a central Asterisk box through an IP WAN (ADSL
or MPLS), and have the asterisk box connected to the central PBX so that
any call directed to the root number of the company can be routed by the
PBX to * and then to the remote IP phones.

Then * could conquer also the typical domain of the legacy PBX, and then
...anyone know what else...


Francesco Pellegrini


++
|  Frame Srl |
|  Via Antonio Cantore 62/10 |
|  16149 Genova  |
|  Tel.   +39 010 8680570|
|  Fax.  +39 010 6591413 |
|  Cell.  +348 2237798   |
++





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RE: [Asterisk-Users] Alternatives to SpanDSP??

2005-04-25 Thread Jeremy Melanson

More like, I already have enough Digium cards, and I don't want purchase
a bunch of fax/modems and more Digium cards than I alrady have.
I have a PRI line that I'd like to support high-volume faxing on. I've
gotten SpanDSP to work with * over the PRI, but I need a more
reliability.
That, and I guess I'm probably just being cheap too :-)

-
Jeremy

On Mon, 2005-04-25 at 13:15 -0500, Anton Krall wrote:
> Maybe I started the day slow :) but let me see if I undertood correctly.
> 
> You say that you don't want to rely on having to buy Digums or any other
> type of cards in oder to tie everything into spandsp and * but you would
> rather have dedicated PSTN lines with faxes on them?
> 
> |-Original Message-
> |From: [EMAIL PROTECTED] 
> |
> |I guess I didn't word this right.
> |It's not that SpanDSP ties up extensions, as it definitely 
> |doesn't. I was more referring to the standard hardware-based 
> |solutions out there that need to have a dedicated line for an 
> |incoming fax. I need the ability to send and receive faxes 
> |with a good amount of reliability, and would love to integrate 
> |it with Asterisk. I'm just not keen on needing to buy a bunch 
> |of Digium TDM cards just to support such a solution.
> |
> |Don't get me wrong, SpanDSP is great! I'm just looking for 
> |something a little more "enterprise-ready". 
> |
> |On Mon, 2005-04-25 at 12:07 -0500, Eric Wieling aka ManxPower wrote:
> |> I wasn't aware that SpanDSP tied up a bunch of extensions.
> |> 
> |> Jeremy Melanson wrote:
> |>   > I'm trying to see if anyone knows of an alternative solution, 
> |> commercial or non-commercial, to SpanDSP. I'm specifically looking 
> |for another software-based, DSP fax that doesn't require me to add a tie up 
> a 
> |> > bunch of extensions on my PBX.
> |> > 
> |> > Has anyone ever seen such an animal, or gotten such it to play nice 
> |> > with Asterisk?
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Re: [Asterisk-Users] Broadvoice Down?

2005-04-25 Thread Luki

Broadvoice doesn't work for me in or out, regardless of proxy used. It
registers fine, but that's about it. Yep, looks like things are down.
And...

" Service Notes  
There are no service announcements at this time."

Indeed...
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RE: [Asterisk-Users] voip problems

Seems to be the norm in most cases

I dial out on one ITSP who seems to always have good dialing with no
issues...

I receive on another that gives me DIDs and 800s (still working out
kinks)...

W



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gerard
Marcel
Sent: Monday, April 25, 2005 1:04 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] voip problems

How do you guys deal with voip problems?  do you have multiple backups
such as land lines, and different voip providers?

Regards,

GM
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RE: [Asterisk-Users] voip problems

2005-04-25 Thread Kerry Garrison

The quick and dirty method is to have the main lines come in over PSTN
lines, this can then be call-forwarded to a VOIP provider. This makes it
simple to change VOIP providers on-the-fly as well as turn off forwarding
and let the PSTN line come into the PBX. Setting up failover on outbound
calls through multiple providers is relativly simple but keeping the main
business number pointing to whatever service you are using is the tricky
part.
-Kerry
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gerard Marcel
Sent: Monday, April 25, 2005 1:04 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] voip problems

How do you guys deal with voip problems?  do you have multiple backups such
as land lines, and different voip providers?

Regards,

GM
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Re: [Asterisk-Users] Broadvoice Down?

2005-04-25 Thread Rich Adamson

Sounds like either BV or their pstn supplier did an upgrade last
night of some sort. (That's just a guess on my part. Certainly
would not have taken this long to fix a problem with that wide
of a scope if it was simply a so called bug. Or, maybe they forgot
to pay their bill.)



> I guess 2 hours is 'soon' to them.  I'm still down.
> Is there a reliable voip provider out there that works with Asterisk?
> I can't have downtime like this.. it just makes my company look bad.
> 
> JD
> 
> JD Austin wrote:
> 
> I tried calling Broadvoice support.. on hold for 1/2 hour then it hung up 
> on me with a 
reorder (fast busy), so I tried again.
> Just got through to a rep- they said it's a 'carrier issue' that their 
> 'partner carrier' 
was having issues and that it would be up
> soon.
> Makes me wonder if I should be signing up with their 'partner carrier' 
> instead.
> 
> JD
> 
> Jerry Geis wrote:
> 
> I am having the same broadvoice issue at the moment.
> 
> jerry
> 
> Is anyone else having difficulty with their Broadvoice service? When 
> I 
> dial my number right now it rings either fast busy or tells me it 
> cannot 
> complete the call.
> 
> I can make outgoing calls from my system through broadvoice however. 
> Seems their inbound trunks hit capacity?
> 
> Am I alone in this?
> -Max
> 
> -- 
>Max Clark
>max [at] clarksys.com
>http://www.clarksys.com
> 
>  
> --
> 
> ___
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> 
> -- 
> JD Austin
> Twin Geckos Technology Services LLC
> email: [EMAIL PROTECTED]
> http://www.twingeckos.com
> phone/fax: 480.344.2640 
> 
>  
> --
> 
> ___
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> 
> -- 
> JD Austin
> Twin Geckos Technology Services LLC
> email: [EMAIL PROTECTED]
> http://www.twingeckos.com
> phone/fax: 480.344.2640 
---End of Original Message-


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Re: [Asterisk-Users] Broadvoice Down?

2005-04-25 Thread Derek Whitten

I have had good luck with nufone...  

http://www.nufone.net

http://www.nufone.net/rates.csv


On Mon, 2005-04-25 at 12:48, JD Austin wrote:
> I guess 2 hours is 'soon' to them.  I'm still down.
> Is there a reliable voip provider out there that works with Asterisk?
> I can't have downtime like this.. it just makes my company look bad.
> 
> JD
> 
> JD Austin wrote: 
> > I tried calling Broadvoice support.. on hold for 1/2 hour then it
> > hung up on me with a reorder (fast busy), so I tried again.
> > Just got through to a rep- they said it's a 'carrier issue' that
> > their 'partner carrier' was having issues and that it would be up
> > soon.
> > Makes me wonder if I should be signing up with their 'partner
> > carrier' instead.
> > 
> > JD
> > 
> > Jerry Geis wrote: 
> > > I am having the same broadvoice issue at the moment.
> > > 
> > > jerry
> > > Is anyone else having difficulty with their Broadvoice service? When I 
> > > dial my number right now it rings either fast busy or tells me it cannot 
> > > complete the call.
> > > 
> > > I can make outgoing calls from my system through broadvoice however. 
> > > Seems their inbound trunks hit capacity?
> > > 
> > > Am I alone in this?
> > > -Max
-- 
Derek Whitten <[EMAIL PROTECTED]>
kFuQ Productions


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RE: [Asterisk-Users] Broadvoice Down?

2005-04-25 Thread Andre Normandin

Just as a side note, I use the NYC broadvoice proxy.. Which one do you use
that seems to be down?

- Andre
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jerry Geis
Sent: Monday, April 25, 2005 1:14 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Broadvoice Down?

I am having the same broadvoice issue at the moment.

jerry

Is anyone else having difficulty with their Broadvoice service? When I
dial my number right now it rings either fast busy or tells me it cannot
complete the call.

I can make outgoing calls from my system through broadvoice however.
Seems their inbound trunks hit capacity?

Am I alone in this?
-Max

--
Max Clark
max [at] clarksys.com
http://www.clarksys.com

Just
as a side note, I use the NYC broadvoice proxy.. Which one do you use that seems
to be down?
- Andre

-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Jerry
GeisSent: Monday, April 25, 2005 1:14 PMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Broadvoice
Down?I am having the same broadvoice issue
at the moment.jerryIs anyone else having difficulty with their Broadvoice service? When I
dial my number right now it rings either fast busy or tells me it cannot
complete the call.

I can make outgoing calls from my system through broadvoice however.
Seems their inbound trunks hit capacity?

Am I alone in this?
-Max

--
Max Clark
max [at] clarksys.com
http://www.clarksys.com
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Re: [Asterisk-Users] voip problems

2005-04-25 Thread Gerard Marcel

In that case, where is the savings on telephone bills?


Regards,


GM



On 4/25/05, Wiley Siler <[EMAIL PROTECTED]> wrote:
> Seems to be the norm in most cases
> 
> I dial out on one ITSP who seems to always have good dialing with no
> issues...
> 
> I receive on another that gives me DIDs and 800s (still working out
> kinks)...
> 
> W
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Gerard
> Marcel
> Sent: Monday, April 25, 2005 1:04 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] voip problems
> 
> How do you guys deal with voip problems?  do you have multiple backups
> such as land lines, and different voip providers?
> 
> Regards,
> 
> GM
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[Asterisk-Users] Polycom IP4000 Conference Phone

Title: Polycom IP4000 Conference Phone






Can someone verify that this phone uses the same configs and sip.ld and other files as the IP 500 ?


I jus tgot one and I cannot get it provisioned yet.


Thanks,

Wiley



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Re: [Asterisk-Users] Broadvoice Down?

Looking at asterisk from the command line I notice that there is stuff 
after my number; -; I've never noticed this before.
Is it normal?  Why would this suddenly stop working? I haven't touched 
anything on the server for a few days.

JD
Connected to Asterisk 1.0.7 currently running on asterisk1 (pid = 2269)
Verbosity is at least 4
   -- Executing GotoIf("SIP/4804221250-dd40", 
"1?from-pstn-reghours|s|1:") in new stack
   -- Goto (from-pstn-reghours,s,1)
   -- Executing GotoIf("SIP/4804221250-dd40", 
"0?from-pstn-reghours-nofax|s|1:2") in new stack
   -- Goto (from-pstn-reghours,s,2)
   -- Executing Answer("SIP/4804221250-dd40", "") in new stack
   -- Executing NVFaxDetect("SIP/4804221250-dd40", "") in new stack
   -- Executing GotoIf("SIP/4804221250-bd48", 
"1?from-pstn-reghours|s|1:") in new stack
   -- Goto (from-pstn-reghours,s,1)
   -- Executing GotoIf("SIP/4804221250-bd48", 
"0?from-pstn-reghours-nofax|s|1:2") in new stack
   -- Goto (from-pstn-reghours,s,2)
   -- Executing Answer("SIP/4804221250-bd48", "") in new stack
   -- Executing NVFaxDetect("SIP/4804221250-bd48", "") in new stack
   -- Executing Hangup("SIP/4804221250-dd40", "") in new stack
 == Spawn extension (from-pstn-reghours, h, 1) exited non-zero on 
'SIP/4804221250-dd40'
   -- Executing Hangup("SIP/4804221250-bd48", "") in new stack
 == Spawn extension (from-pstn-reghours, h, 1) exited non-zero on 
'SIP/4804221250-bd48'
asterisk1*CLI> quit

 

Chuck Smith wrote:
I am having trouble with Broadvoice as well. My server sees the call but I
get no audio then the line drops but the call stays up at the asterisk
server. Goes to voicemail but on the far end the phone is already on the
hook.
Glad I wasn't the only one. Thought I was going crazy over here.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Max Clark
Sent: Monday, April 25, 2005 1:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Broadvoice Down?
Is anyone else having difficulty with their Broadvoice service? When I 
dial my number right now it rings either fast busy or tells me it cannot 
complete the call.

I can make outgoing calls from my system through broadvoice however. 
Seems their inbound trunks hit capacity?

Am I alone in this?
-Max
 

--
JD Austin
Twin Geckos Technology Services LLC
email: [EMAIL PROTECTED]
http://www.twingeckos.com
phone/fax: 480.344.2640 

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[Asterisk-Users] voip problems

2005-04-25 Thread Gerard Marcel

How do you guys deal with voip problems?  do you have multiple backups
such as land lines, and different voip providers?

Regards,

GM
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RE: [Asterisk-Users] Cisco's description of echo

2005-04-25 Thread Keith O'Brien




If you are running a Cisco VoIP 
gateway you can send a 0dBm 1000Khz test tone into or out of a voice port 
with:
 
"test voice 
port  inject-tone network 1000hz"
 
to measure the tone do 
a:
 
"sh call active voice 
brief"
 
>>Another common problem that 
causes echo in networks is not setting your>>loss plan correctly.    
You need to be sure that you aren't coming in too>>hot at any of your analog 
interfaces.   In general you should see a signal>>between -20dbm and -12dbm when someone 
is talking on the line.   If it is>>significantly hotter then you run the 
chance of having a larger reflected>>signal resulting in echo.   I 
typically try padding down analog levels by>>3dB at a time to see if echo is 
reduced.   >How do 
you measure the amplitude of a pstn line? As an audio engineer in a>previous life, I would love to be able to 
send standard level tones down a>pstn line and measure the amplitude at my 
end, then adjust the input gain>accurately. Is there a way to do 
this?


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RE: [Asterisk-Users] Broadvoice Down?

2005-04-25 Thread Kerry Garrison




While we have only been using Broadvoice for a few months 
now, we have actually had better service through them than with our PSTN 
provider. You could just as easily have had a voice T1 go down which typically 
takes a few hours to replace (and may be the actual problem). The issue is not 
with Broadvoice as a service as much as it seems to be a peering problem. 

 
Yes these issues make you and your company look bad in so 
much as it shows that you do not have a properly designed redundant system. Do 
you back up your server every night even though you only lose a file once a 
year? So why wouldn't you have a failover on your phone 
system?
 
Its very easy to blame Broadvoice for your phone service 
being out (and I am not defending them) but that is ONE connection. Any business 
that is highly reliant on their phones for business should have a backup system 
just for this reason. Possibly even multiple backups depending on the critical 
nature of the business.
 
I have seen businesses completely lose telephone service on 
standard PSTN lines for a day or more at a time. So sh** happens, be prepared. 
Dont put all your eggs in one basket. This is a perfect example of 
why.
 
Kerry Garrison
http://techdatapros.com
 


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of JD 
AustinSent: Monday, April 25, 2005 12:48 PMTo: Asterisk 
Users Mailing List - Non-Commercial DiscussionSubject: Re: 
[Asterisk-Users] Broadvoice Down?
I guess 2 hours is 'soon' to them.  I'm still down.Is there 
a reliable voip provider out there that works with Asterisk?I can't have 
downtime like this.. it just makes my company look bad.JDJD 
Austin wrote: 
I tried 
  calling Broadvoice support.. on hold for 1/2 hour then it hung up on me with a 
  reorder (fast busy), so I tried again.Just got through to a rep- they said 
  it's a 'carrier issue' that their 'partner carrier' was having issues and that 
  it would be up soon.Makes me wonder if I should be signing up with their 
  'partner carrier' instead.JDJerry Geis wrote: 
  I am having the same broadvoice issue at the 
moment.jerryIs anyone else having difficulty with their Broadvoice service? When I 
dial my number right now it rings either fast busy or tells me it cannot 
complete the call.

I can make outgoing calls from my system through broadvoice however. 
Seems their inbound trunks hit capacity?

Am I alone in this?
-Max

-- 
   Max Clark
   max [at] clarksys.com
   http://www.clarksys.com
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RE: [Asterisk-Users] Broadvoice Down?

2005-04-25 Thread Andre Normandin

Ahh,,

That explains alot..

Outbound service is VERY VERY distorted, like alot of packet loss is
occuring.. It was fine last night at 11:30PM, when I was talking outbound
via it.. I thought it was my Internet connection, but I checked and have
been unable to find anything wrong..

Haven't had any inbound from my broadvoice service today.. Not sure if that
is because no one tried, or no one has been able to get through :-)



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Max Clark
Sent: Monday, April 25, 2005 1:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Broadvoice Down?


Is anyone else having difficulty with their Broadvoice service? When I
dial my number right now it rings either fast busy or tells me it cannot
complete the call.

I can make outgoing calls from my system through broadvoice however.
Seems their inbound trunks hit capacity?

Am I alone in this?
-Max

--
   Max Clark
   max [at] clarksys.com
   http://www.clarksys.com
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RE: [Asterisk-Users] Broadvoice Down?

2005-04-25 Thread Darren Wright

Yup..major broadvoice issues here as well.  I can dial-in, but no dial out. 
 This is the first problem for me in 2 months.

-D

From: [EMAIL PROTECTED] on behalf of Sean Kennedy
Sent: Mon 4/25/2005 3:03 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Broadvoice Down?

JD Austin wrote:

> I tried calling Broadvoice support.. on hold for 1/2 hour then it hung
> up on me with a reorder (fast busy), so I tried again.
> Just got through to a rep- they said it's a 'carrier issue' that their
> 'partner carrier' was having issues and that it would be up soon.
> Makes me wonder if I should be signing up with their 'partner carrier'
> instead.
>
> JD

Actually, with all the threads I've seen on the mailing list, I'm weary
of anything having to do with broadvoice.

Personally.  Maybe it's just that they have such a large user base on
linux.  Who knows.

Voicepulse gets my business tho.  :)

Sean
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RE: [Asterisk-Users] Broadvoice Down?

2005-04-25 Thread Andre Normandin

I just checked (4:30PM EDT) , my inbound seems to be working fine, however,
I have the same issue as I have with outbound.. Very choppy audio..

I am having the same broadvoice issue at the moment.

jerry

Is anyone else having difficulty with their Broadvoice service? When I
dial my number right now it rings either fast busy or tells me it cannot
complete the call.

I can make outgoing calls from my system through broadvoice however.
Seems their inbound trunks hit capacity?

Am I alone in this?
-Max

--
Max Clark
max [at] clarksys.com
http://www.clarksys.com

I just
checked (4:30PM EDT) , my inbound seems to be working fine, however, I have the
same issue as I have with outbound.. Very choppy audio..
- Andre

I can make outgoing calls from my system through broadvoice however.
Seems their inbound trunks hit capacity?

Am I alone in this?
-Max

Re: [Asterisk-Users] Dial Plan - How to prepend a digit

2005-04-25 Thread Franco Bellagamba

Daniel, try

_X.,1,Dial(123${EXTEN})

That will prefix "123" to the dialed number.

Franco
- Original Message - 
From: "Daniel Salama" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Monday, April 25, 2005 4:05 PM
Subject: [Asterisk-Users] Dial Plan - How to prepend a digit


> I'd like to create a dial rule that when someone tries to dial a
> particular number, the same number is dialed, except that prefixed with
> some additional digit(s). How can this be specified on extensions.conf?
>
> Thanks,
> Daniel
>
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Re: [Asterisk-Users] Broadvoice Down?

2005-04-25 Thread Gerard Marcel

I am having the same issues.

Regards,

GM



On 4/25/05, Jerry Geis <[EMAIL PROTECTED]> wrote:
>  I am having the same broadvoice issue at the moment.
>  
>  jerry
>  Is anyone else having difficulty with their Broadvoice service? When I 
> dial my number right now it rings either fast busy or tells me it cannot 
> complete the call.
> 
> I can make outgoing calls from my system through broadvoice however. 
> Seems their inbound trunks hit capacity?
> 
> Am I alone in this?
> -Max
> 
> -- 
>  Max Clark
>  max [at] clarksys.com
>  http://www.clarksys.com
>  
>  
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[Asterisk-Users] Re: Dial Plan - How to prepend a digit

2005-04-25 Thread Tony Mountifield

In article <[EMAIL PROTECTED]>,
Daniel Salama <[EMAIL PROTECTED]> wrote:
> I'd like to create a dial rule that when someone tries to dial a 
> particular number, the same number is dialed, except that prefixed with 
> some additional digit(s). How can this be specified on extensions.conf?

Easy. Say you want to prepend 90 onto any number beginning with 7,
and 91 onto a particular number, and dial anything else normally:

[specials]
exten => _7.,1,Dial(Zap/g1/90${EXTEN})
exten => 1234567890,1,Dial(Zap/g1/91${EXTEN})

[outgoing]
include => specials
exten => _X.,1,Dial(Zap/g1/${EXTEN})

You need to put the exceptions into an included context, because that's
the only way to guarantee they are considered before the default case.
See http://www.voip-info.org/wiki-Asterisk+config+extensions.conf+sorting
for more discussion of this issue.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [Asterisk-Users] Broadvoice Down?


Sean Kennedy wrote:
Actually, with all the threads I've seen on the mailing list, I'm 
weary of anything having to do with broadvoice.

Personally.  Maybe it's just that they have such a large user base on 
linux.  Who knows.

Voicepulse gets my business tho.  :)
Sean
Voice pulse doesn't offer dids in my state (AZ)  :(
--
JD Austin
Twin Geckos Technology Services LLC
email: [EMAIL PROTECTED]
http://www.twingeckos.com
phone/fax: 480.344.2640 

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RE: [Asterisk-Users] Phone Recommendation.

Call waiting can be disabled in Asterisk via *71 regardless of the phone
used.

Cheers,
Wiley
  



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sean A.
Newton
Sent: Monday, April 25, 2005 11:56 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Phone Recommendation.


I'm looking for recommendations for a office phone that has the ability
to disable call-waiting.

Needs to be similar in features to a Polycom IP300. 

Thanks,

--Sean

-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
 sean a. newton  [EMAIL PROTECTED]
 louisville, ky, usa http://wewt.net 

 Another day, another convertible and another hotel 
 full of cops.-- Hunter S. Thompson
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-

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[Asterisk-Users] Has anyone used Libretel DIDs with Asterisk?

2005-04-25 Thread beonice

Hello, all.

Google searches reveal that a couple of users have
used Libretel port-of-call service via FWD using
Asterisk. I don't have a FWD account. I don't have a
SIP phone (no SIP setup at all, in fact).

I just purchased a DID in Brazil from Libretel and
discovered that they assume that I either have a SIP
phone or a VOIP number through BroadVoice. My provider
is actually VoicepulseConnect in California. My server
here is an Asterisk box connecting to a Voicepulse DID
using IAX.

HELP. :( I was assuming everyone supports IAX. I guess
I should have made sure first, huh?

Any help would be much appreciated.

Thanks,
Maya


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Re: [Asterisk-Users] Broadvoice Down?