date:20051022

I'm having the following recurring problem with asterisk:

When for any reason one of my SIP providers fails to register (i.e.
internet connection dropped), ALL SIP channels fail.  This means that,
for example, when my internet connection is out, none of my internal sip
phones register, and I'm unable to place outgoing calls (through IAX),
or to check voicemail.

Currently (and since yesterday evening), sipmedia.com/myphonecompany.com
is completely off the radar.  No DNS entry found -- not even a
name-server.  They've had this sort of massive failure before, but this
is one of the longest for all I can tell.  While that's a major problem,
it also meant that until I commented out the register =
sip.sipmedia.com statements, my entire phonesystem was unavailable.

1. Is there any way to get Asterisk to behave less absolute when one sip
registration fails?
2. Is anyone else experiencing the same sipmedia outtage, and/or has
information on when they'll be back?  Tech support seems affected, and
other direct numbers I have go into voicemail.

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Re: [Asterisk-Users] Does fwdout even work anymore?


  Mine stopped working sometime back in Feb.  I just made the changes  
  so everything points to fwdOUT.net now, but it still seems to fail.
  
  Using a sniffer, I see packets going out, but none coming back.  I  
  have a firewall, but 4569 has been opened, and I'm not seeing denys  
  on the firewall anyway.  I'm just not getting a response.
  
  Any ideas?
  
  ~jay
 
 FWD used work not to long ago, but is not working today.   IAX
 registration  to FWD is not going through.  Is anybody lucky?

As of 8:20 am CDT, both FWD and Iaxtel.com are unresponsive. It
appears the FWD iax server can be reached via a ping, but there
is no response from it for a iax register. That would imply their
asterisk crashed but the server is up.


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Re: [Asterisk-Users] One SIP dead, all SIP dead -- sipmedia gone?

 I'm having the following recurring problem with asterisk:
 
 When for any reason one of my SIP providers fails to register (i.e.
 internet connection dropped), ALL SIP channels fail.  This means that,
 for example, when my internet connection is out, none of my internal sip
 phones register, and I'm unable to place outgoing calls (through IAX),
 or to check voicemail.
 
 Currently (and since yesterday evening), sipmedia.com/myphonecompany.com
 is completely off the radar.  No DNS entry found -- not even a
 name-server.  They've had this sort of massive failure before, but this
 is one of the longest for all I can tell.  While that's a major problem,
 it also meant that until I commented out the register =
 sip.sipmedia.com statements, my entire phonesystem was unavailable.
 
 1. Is there any way to get Asterisk to behave less absolute when one sip
 registration fails?
 2. Is anyone else experiencing the same sipmedia outtage, and/or has
 information on when they'll be back?  Tech support seems affected, and
 other direct numbers I have go into voicemail.

FWIW, I don't see that same issue. I only have one sip provider (and
serveral iax links), but when the sip register fails I'm still able to
complete local and iax calls. This is with cvs-head as of yesterday.


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[Asterisk-Users] play a voice file voice for decline

2005-10-22 Thread Asterisk guy

When get sip respond 6xx ( such as 603 decline),  I want asterisk to
play a voice file to the caller,  how to do this in extensions ?

for example, when get 603 respond,  play  decline.gsm  to caller
   when get 604 respond, play doesnot-exit.gsm  to caller
when get 606 respond , play not-acceptalbe.gsm to caller







SIP response codes, class 6: Global failures


600 Busy Everywhere
603 Decline
604 Does Not Exist Anywhere
606 Not Acceptable
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Re: [Asterisk-Users] One SIP dead, all SIP dead -- sipmedia gone?

2005-10-22 Thread Joe Greco

 Currently (and since yesterday evening), sipmedia.com/myphonecompany.com
 is completely off the radar.  No DNS entry found -- not even a
 name-server.  They've had this sort of massive failure before, but this
 is one of the longest for all I can tell.  While that's a major problem,
 it also meant that until I commented out the register =
 sip.sipmedia.com statements, my entire phonesystem was unavailable.
 
[...]
 2. Is anyone else experiencing the same sipmedia outtage, and/or has
 information on when they'll be back?  Tech support seems affected, and
 other direct numbers I have go into voicemail.

Never heard of them.  The IP block which their NS1.SIPMEDIA.COM is in is

NetRange:   66.128.0.0 - 66.128.15.255
CIDR:   66.128.0.0/20
NetName:VITCOM-BLK
NetHandle:  NET-66-128-0-0-1
Parent: NET-66-0-0-0-0
NetType:Direct Allocation
NameServer: NS1.XCHANGETELE.COM
NameServer: NS3.XCHANGETELE.COM
Comment:ADDRESSES WITHIN THIS BLOCK ARE NON-PORTABLE
RegDate:2001-06-05
Updated:2005-03-11

and this appears to be off the air.  We have no route listed for this at
this time.  The IP block for NS3.SIPMEDIA.COM does have a route but
traceroutes to deadness.

10  POS7-0.GW12.NYC1.ALTER.NET (152.63.29.197)
11  band-x-gw.customer.alter.net (157.130.2.218)
12  * * *
13  * * *

NetRange:   69.1.236.0 - 69.1.237.255
CIDR:   69.1.236.0/23
NetName:XCHANGETELE
NetHandle:  NET-69-1-236-0-1
Parent: NET-69-1-224-0-1
NetType:Reassigned
NameServer: NS1.XCHANGETELE.COM
NameServer: NS3.XCHANGETELE.COM
NameServer: NS4.XCHANGETELE.COM
NameServer: NS5.XCHANGETELE.COM
Comment:
RegDate:2004-02-24
Updated:2004-02-24

The parent block containing this one is advertised in BGP right now, but
I see a whole bunch of /24's covering this block as well, and that route
is not covered by a /24 or /23 announcement.  This looks like they are
fully withdrawn at this time. 

I see a slew of withdrawls for that route around 10/21 23:42.

FixedOrbit sees them as singlehomed to Savvis, which isn't having any
problems that I'm aware of, other than this humorous bit:

http://www.nydailynews.com/news/local/story/357856p-304797c.html

:-)

I do notice that there seems to be no web site for www.xchangetele.com,
which suggests that possibly they tanked.  If your sipmedia.com was
reselling their services, then it is reasonable to think that they went
along for that ride.

... JG
-- 
Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net
We call it the 'one bite at the apple' rule. Give me one chance [and] then I
won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN)
With 24 million small businesses in the US alone, that's way too many apples.
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[Asterisk-Users] Linksys pap2 behind Linksys RT31

2005-10-22 Thread Sebastian Milioto

Hi all,

I have a public ip in Linksys RT31 (2 FXS port + 3 swtich port + 1
uplink port). I want to add behind it, a Linksys pap2 (uplink port + 2
FXS port) with private ip.
I understand that I have to configure Port forwarding or port
triggering (really I'm not sure which one).
Is someone already configured this toplogy? Could you help me with that, please?

Thanks very much in advance,

Sebastian
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Re: [Asterisk-Users] Does fwdout even work anymore?

2005-10-22 Thread trixter aka Bret McDanel

Initially I thought this may have been the fiasco last night or the
night before (I forget now) where level3 did a software upgrade and it
went awry.  With the pings responding I now wonder.

It still could be this (all symptoms from the same problem).  I am
thinking about signing up for FWD-out anyway, I might do that tonight
and see if it works for me.  

I dont know the exact routes that everyone is using to get there, which
would play a role in this.  Just flinging wild guesses based on current
events.

On Sat, 2005-10-22 at 08:21 -0600, Rich Adamson wrote:
   Mine stopped working sometime back in Feb.  I just made the changes  
   so everything points to fwdOUT.net now, but it still seems to fail.
   
   Using a sniffer, I see packets going out, but none coming back.  I  
   have a firewall, but 4569 has been opened, and I'm not seeing denys  
   on the firewall anyway.  I'm just not getting a response.
   
   Any ideas?
   
   ~jay
  
  FWD used work not to long ago, but is not working today.   IAX
  registration  to FWD is not going through.  Is anybody lucky?
 
 As of 8:20 am CDT, both FWD and Iaxtel.com are unresponsive. It
 appears the FWD iax server can be reached via a ping, but there
 is no response from it for a iax register. That would imply their
 asterisk crashed but the server is up.

-- 
Trixter http://www.0xdecafbad.com Bret McDanel
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] One SIP dead, all SIP dead -- sipmedia gone?

2005-10-22 Thread Olle E. Johansson

Jay Milk wrote:
 I'm having the following recurring problem with asterisk:
 
 When for any reason one of my SIP providers fails to register (i.e.
 internet connection dropped), ALL SIP channels fail.  This means that,
 for example, when my internet connection is out, none of my internal sip
 phones register, and I'm unable to place outgoing calls (through IAX),
 or to check voicemail.
 
 Currently (and since yesterday evening), sipmedia.com/myphonecompany.com
 is completely off the radar.  No DNS entry found -- not even a
 name-server.  They've had this sort of massive failure before, but this
 is one of the longest for all I can tell.  While that's a major problem,
 it also meant that until I commented out the register =
 sip.sipmedia.com statements, my entire phonesystem was unavailable.
 
 1. Is there any way to get Asterisk to behave less absolute when one sip
 registration fails?
 2. Is anyone else experiencing the same sipmedia outtage, and/or has
 information on when they'll be back?  Tech support seems affected, and
 other direct numbers I have go into voicemail.
 
DNS failures are way too catastrophical for Asterisk now. We need to fix
that urgently.

/O
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Re: [Asterisk-Users] Does fwdout even work anymore?

 Initially I thought this may have been the fiasco last night or the
 night before (I forget now) where level3 did a software upgrade and it
 went awry.  With the pings responding I now wonder.

 It still could be this (all symptoms from the same problem).  I am
 thinking about signing up for FWD-out anyway, I might do that tonight
 and see if it works for me.  

Don't know if FWD changed their internal procedures, but in the past
part of the iax signup required some manual action on their part. So
don't expect your sign up to work initially.

Given the relative instability associated with FWD (compared to other
itsp's), I generally comment out the register statement and only insert
it to play/validate service, etc.


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Re: [Asterisk-Users] Filenaming for Incoming Queue Call Recordings (Reposted from changing the filename of incoming call recordings)

2005-10-22 Thread Lenz



Hello,
I usually use

exten =  
s,1,SetVar(MONITOR_FILENAME=/var/spool/asterisk/q/QSAMPLE-${UNIQUEID})

exten = s,2,Queue(q-sample|nt|||60)

and it seems to work, then use QueueMetrics to keep track of who was  
talking to whom, instead of using the Agents monitoring.

Bye
l.



On Sat, 22 Oct 2005 12:19:42 +0200, KRTorio [EMAIL PROTECTED] wrote:


Regarding my previous post:
 Is there an easy way to modify the filename of an incoming call's
recording, or are we stuck to agent--unix timestamp format given  
to

us by Asterisk?


There seems to be neither an equivalent ChangeMonitor() application for

incoming, nor you can tweak the recording's filename in agents.conf.
 It seems that the only way to change the filenaming incoming queue call
recordings is by modifying this line here in chan_agent.c :
  snprintf(filename, sizeof(filename), agent-%s-%s,p-agent,
ast-uniqueid);
 But before I do that, is there a better way to do this, one that doesn't
require modifying the source code?




--
Loway Research - Home of QueueMetrics
http://queuemetrics.loway.it

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RE: [Asterisk-Users] Callerid on t1 lines

2005-10-22 Thread gw

I had this, my problem turned out to be in zapata.conf on the receiving
end.

I'll do the KS, right now I am using LS.  Any particular reason to use
KS?  The LSCPD on the adit seems to work fairly decently. 

Now I just need to work out some echo, although I have done milliwatt
tests to a local line, I still seem to get echo at the beginning of a
call regardless of how I set the training.

Thanks,
Greg

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: Monday, October 17, 2005 1:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Callerid on t1 lines

How are you checking if CallerID is received?
You should do at least a Noop(${CALLERIDNUM}) or if running head:
Noop(${CALLERID(NUM)}) so that you can verify that.
How do you know that your telco is giving you CID?
If you live in the US then setup the Adit to do LSCPD and Asteisk as
ks_fxs. and not loop start.

On 10/17/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
 Hello,
 That's what I really needed to know, that it was possible.

 Here is my setup:

 Telco Analog W/CID  FXO ADIT600 LoopStart  Loopstart Asterisk T1.

 Then LoopStart Asterisk T1  Loopstart Panasonic DBS PBX T1.

 At this point, I do not see any CID coming in from the telco into 
 asterisk.  Even when I increase the wait time, and the zapata.conf has

 asreceived set.

 I tried EM from the dbs to asterisk, but would get no dialtone from 
 asterisk as it was not working properly with immediate mode.

 The main purpose of the setup is to do call recording on 3 analog and 
 2 bri lines, and pass them to the pbx transparently.  Also to allow * 
 transfers and queuing.

 Thanks,
 Greg

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of C F
 Sent: Saturday, October 15, 2005 9:16 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Callerid on t1 lines

 What is the adit 600 doing? FXO? FXS? how you connected to the PSTN?
 I got an Adit 600 with both FXO and FXS as well as a PRI and I'm 
 getting CallerID on all three.

 On 10/14/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
  Hello All,
  Just a question, I have an adit600 and I am looking for a way to 
  pull the incoming cid into asterisk.
 
  Does anyone know if this is just not possible via t1? Or is it only 
  available on PRI?
 
  Thanks,
  Greg
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Re: [Asterisk-Users] voip provider in a box

2005-10-22 Thread Alistair Cunningham


Bret,

See my recent post:

http://lists.digium.com/pipermail/asterisk-users/2005-October/130542.html

I'll send you an email off list with the features and future roadmap.

Alistair Cunningham,
Integrics Ltd,
+44 (0)7870 699 479
http://integrics.com/


trixter aka Bret McDanel wrote:

I am tasked with evaluating ready made solutions for a voip provider.
Does anyone have any recommendations for software, specifically the
environment will be a chargable voip provider (ie broadvoice, vonage,
etc).  They wanted me to see what was there and write something if
nothing they like exists.

Thanks





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RE: [Asterisk-Users] Terrible echo with Te110P and Adit 600

2005-10-22 Thread gw

I have a similar setup... I set the canceller on the incoming PSTN
lines, but turn it off on the FXS.  

I have no local internal echo over the t1, but moderate over the PSTN.
I managed to tweak it a little and most of my  outbound (local side)
echo is minimized, but still there a little. I have no incoming echo.

You mind elaborating on where you are getting the echo? 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darren
Wright
Sent: Tuesday, October 18, 2005 10:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Terrible echo with Te110P and Adit 600


8 POTS lines - Adit 600 - TE110P - Dell Precision 530 Dual Xeon running
1.0.9 and 1.2 (tried both)


The echo is insurmountable.  I have tried everything, and the pots lines
are clean.  If I go from an FXO on the Adit 600 straight to an FXS, I
get no echo from an analog phone.  

I put an 128ms T1 echo canceller in between the adit and the TE110P, and
the echo was still horrible.  

I finally disabled the Zapata echo cancellerand WHAMMO!  It's
perfect now.  

The TE110P is on it's own IRQ.. and the machine has PLENTY of
horsepower.

Any ideas so I don't have to spend $1000 on an echo canceller?

-Darren




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Re: [Asterisk-Users] voip provider in a box



Hey ho,

We have something like that (tailored for huge installations), contact
me off list for more info.

zoa.

trixter aka Bret McDanel wrote:


I am tasked with evaluating ready made solutions for a voip provider.
Does anyone have any recommendations for software, specifically the
environment will be a chargable voip provider (ie broadvoice, vonage,
etc).  They wanted me to see what was there and write something if
nothing they like exists.

Thanks





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Re: [Asterisk-Users] Filenaming for Incoming Queue Call Recordings (Reposted from changing the filename of incoming call recordings)

2005-10-22 Thread Kevin Bockman


KRTorio wrote:

 Is there an easy way to modify the filename of an incoming call's
 recording, or are we stuck to agent--unix timestamp format given 
to us by Asterisk? 


Your answer was in queues.conf that's why you only got 1 reply.


Kevin
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Re: [Asterisk-Users] Does fwdout even work anymore?

2005-10-22 Thread Sergey Okhapkin





I just tried to place a call thru fwdout, works fine.

On Sat, 2005-10-22 at 06:40 -0700, trixter aka Bret McDanel wrote:


Initially I thought this may have been the fiasco last night or the
night before (I forget now) where level3 did a software upgrade and it
went awry.  With the pings responding I now wonder.

It still could be this (all symptoms from the same problem).  I am
thinking about signing up for FWD-out anyway, I might do that tonight
and see if it works for me.  

I dont know the exact routes that everyone is using to get there, which
would play a role in this.  Just flinging wild guesses based on current
events.

On Sat, 2005-10-22 at 08:21 -0600, Rich Adamson wrote:
   Mine stopped working sometime back in Feb.  I just made the changes  
   so everything points to fwdOUT.net now, but it still seems to fail.
   
   Using a sniffer, I see packets going out, but none coming back.  I  
   have a firewall, but 4569 has been opened, and I'm not seeing denys  
   on the firewall anyway.  I'm just not getting a response.
   
   Any ideas?
   
   ~jay
  
  FWD used work not to long ago, but is not working today.   IAX
  registration  to FWD is not going through.  Is anybody lucky?
 
 As of 8:20 am CDT, both FWD and Iaxtel.com are unresponsive. It
 appears the FWD iax server can be reached via a ping, but there
 is no response from it for a iax register. That would imply their
 asterisk crashed but the server is up.

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[Asterisk-Users] Re: uable to establish link between asterisk to external phone

2005-10-22 Thread kotesh m

I was able to resolve the problem to some extent. My out going calls are 
working fine when I included the external IP adress in sip_nat.conf file. But 
my incoming calls are going voice mail instead of ringing the telephone 
attached to my sipura device.

Any help is appreciated.

--kotesh
 
On 10/19/05, kotesh m [EMAIL PROTECTED] wrote: 
 
My mistake it is [EMAIL PROTECTED] 1.5. 
  
 --k

 
  
On 10/19/05, kotesh m [EMAIL PROTECTED]   wrote: 
 
 
 Hi,
  
 I am new Asterisk. I configured asterisk1.5 and be able to communicate from 
iaxComm dial pad to external computer i.e out side my router/LAN. When I make 
call from iamComm of external computer to my cell phone, I am getting the ring 
but not able to listen voice on both sides. Do I need to make any special 
configuration to make voice link.
  
 I found the same problem when used Sipura SIP device. 
  
 Please let me know if I am missing anything.
  
 Appreciate any help
  
 --k

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RE: [Asterisk-Users] One SIP dead, all SIP dead -- sipmedia gone?

Thanks for the wealth of information.  I knew they were off-air
DNS-wise, and this happened before a couple of times.  It's just bad
juju to have all your IPs in one block.

I don't think they were reselling, and I actually thought I had a pretty
good report with them -- just have been unable to get anyone on the
phone today.  Other than that, had several DIDs with them at $5/month,
and really haven't seen major issues other than DTMF not working -- but
then, they are SIP-only.  But between pricing and availability, they've
been the best provider out of all I tried (Vonage, Broadvoice,
voicepulse, iax.cc, ... )

 -Original Message-
 From: Joe Greco [mailto:[EMAIL PROTECTED] 
 Sent: Saturday, October 22, 2005 8:34 AM
 To: asterisk-users@lists.digium.com
 Subject: Re: [Asterisk-Users] One SIP dead, all SIP dead -- 
 sipmedia gone?
 
 
  Currently (and since yesterday evening), 
  sipmedia.com/myphonecompany.com is completely off the 
 radar.  No DNS 
  entry found -- not even a name-server.  They've had this sort of 
  massive failure before, but this is one of the longest for 
 all I can 
  tell.  While that's a major problem, it also meant that until I 
  commented out the register = sip.sipmedia.com statements, 
 my entire 
  phonesystem was unavailable.
  
 [...]
  2. Is anyone else experiencing the same sipmedia outtage, 
 and/or has 
  information on when they'll be back?  Tech support seems 
 affected, and 
  other direct numbers I have go into voicemail.
 
 Never heard of them.  The IP block which their 
 NS1.SIPMEDIA.COM is in is
 
 NetRange:   66.128.0.0 - 66.128.15.255
 CIDR:   66.128.0.0/20
 NetName:VITCOM-BLK
 NetHandle:  NET-66-128-0-0-1
 Parent: NET-66-0-0-0-0
 NetType:Direct Allocation
 NameServer: NS1.XCHANGETELE.COM
 NameServer: NS3.XCHANGETELE.COM
 Comment:ADDRESSES WITHIN THIS BLOCK ARE NON-PORTABLE
 RegDate:2001-06-05
 Updated:2005-03-11
 
 and this appears to be off the air.  We have no route listed 
 for this at this time.  The IP block for NS3.SIPMEDIA.COM 
 does have a route but traceroutes to deadness.
 
 10  POS7-0.GW12.NYC1.ALTER.NET (152.63.29.197)
 11  band-x-gw.customer.alter.net (157.130.2.218)
 12  * * *
 13  * * *
 
 NetRange:   69.1.236.0 - 69.1.237.255
 CIDR:   69.1.236.0/23
 NetName:XCHANGETELE
 NetHandle:  NET-69-1-236-0-1
 Parent: NET-69-1-224-0-1
 NetType:Reassigned
 NameServer: NS1.XCHANGETELE.COM
 NameServer: NS3.XCHANGETELE.COM
 NameServer: NS4.XCHANGETELE.COM
 NameServer: NS5.XCHANGETELE.COM
 Comment:
 RegDate:2004-02-24
 Updated:2004-02-24
 
 The parent block containing this one is advertised in BGP 
 right now, but I see a whole bunch of /24's covering this 
 block as well, and that route is not covered by a /24 or /23 
 announcement.  This looks like they are fully withdrawn at this time. 
 
 I see a slew of withdrawls for that route around 10/21 23:42.
 
 FixedOrbit sees them as singlehomed to Savvis, which isn't 
 having any problems that I'm aware of, other than this humorous bit:
 
http://www.nydailynews.com/news/local/story/357856p-304797c.html

:-)

I do notice that there seems to be no web site for www.xchangetele.com,
which suggests that possibly they tanked.  If your sipmedia.com was
reselling their services, then it is reasonable to think that they went
along for that ride.

... JG
-- 
Joe Greco - sol.net Network Services - Milwaukee, WI -
http://www.sol.net We call it the 'one bite at the apple' rule. Give me
one chance [and] then I won't contact you again. - Direct Marketing
Ass'n position on e-mail spam(CNN) With 24 million small businesses in
the US alone, that's way too many apples.
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[Asterisk-Users] Queue Join Event

2005-10-22 Thread Tressler, Joshua A











I apologize if
this is posted again as I sent it last night, but I dont see it anywhere
in the list as of now.







I did a quick Google search of the lists and I hope that I
am not asking a question that has already been answered recently.



I have been working on a interface to use with our CRM software.
I am using the manager interface and mysql to store the changes. The only issue
I am having is when a caller joins the queue.

Currently, I can show the status of phones (ready, not
ready, ringing, ringing ack, in call, etc). What I am wanting to do is to be
able to track the status of the call in the database and do things with it
accordingly. I am able to accomplish this and make it work exactly as I want,
but it requires a modification to the source. For some reason, the JOIN event
in the manager interface doesnt seem to have the unique call id. Almost
every other event does, but JOIN doesnt for some reason. Can anyone
explain why it doesnt?



My boss asked us to remove our hack to the source and find
another way as it we want to be able to update versions of asterisk and not
modify the source. I thought that I could get around this by using the NEWEXTEN
event that happens just before the join, but I cant tie the two events
together.



Basically, with the hack modified, heres what I do:



Call comes in, enter the info into the database with
uniqueid as the key. When a call is answered, I update that record in the
database and so on. Without the uniqueid on the JOIN event I am stuck.



Any suggestions on a way around this, or a better way of
doing it? I would also be curious if anyone would share their setup if the are
attempting the same.



Thanks,


Josh






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[Asterisk-Users] FXO no release useing fxsks disconnect supervision from telco

2005-10-22 Thread steve casto


Adtran 750
d4,ami
loop start
fxs 5x4 port
fxo 1x4 port
TE405P
Asterisk 1.0.9 stable
span 1 - CAC channel bank all FXS
sand 2 - Adtran 750 20 FXS 4 FXO (new FXO card)
span 34 - outgoing only EM

A call comes in from the telco on the FXO port  answered on a FXS 
(fxoks). If the FXS hangs up, ok, if the FXO (fxsks) hangs up first:

1) CPC is receive at the FXO (an open from telco)
2) the A bit for the FXO port - 0
3) FXS port get short reorder then Asterisk dial tone
4) the FXO port is not dropped by Asterisk
5) telco recording like 'Please hang up to make a call' no busy then 
dead air (on the telco line connected to the FXO)
6) to reset,the FXO port needs to be opened at the telco demark  3 sec. 
opened less then 3 sec. the FXO port is still in an off hook  draws 
dial tone from the telco  does not drop
This happens every time. Being that that the FXS, still off hook when 
the FXO hangs up, drops that connection to that FXO, Asterisk must know 
the FXO has hung up but Asterisk does not release the FXO.

thanks
Steve Casto

zaptel.conf
loadzone = us
defaultzone=us
span=1,0,0,d4,ami
#em=1-24
fxoks=1-24
span=2,0,0,d4,ami
#em=25-48
fxoks=25-44
fxsks=45-48
span=3,2,0,d4,ami
em=49-72
span=4,1,0,d4,ami
em=73-96
fxsks=100
fxoks=97

zapata.conf
[channels]
musiconhold=default
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
tranfer=yes
cancallforward=no
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0

group=1
immediate=no
context=dialout
threewaycalling=yes
tranfer=yes

signalling=fxo_ks
group=1
channel = 1-44


threewaycalling=yes
transfer=yes

usecallerid=yes
callerid=  101
mailbox=101
channel=1
callerid=102
mailbox=102
channel=2
callerid=103
mailbox=103
channel=3
callerid=104
mailbox=104
channel=4
callerid=105
mailbox=105
channel=5
callerid=106
mailbox=106
channel=6
callerid=107
mailbox=107
channel=7
callerid=108
mailbox=108
channel=8
callerid=109
mailbox=109
channel=9
callerid=110
mailbox=110
channel=10
callerid=111
mailbox=111
channel=11
callerid=112
mailbox=112
channel=12
callerid=113
mailbox=113
channel=13
callerid=114
mailbox=114
channel=14
callerid=115
mailbox=115
channel=15
callerid=116
mailbox=116
channel=16
callerid=117
mailbox=117
channel=17
callerid=118
mailbox=118
channel=18
callerid=119
mailbox=119
channel=19
callerid=120
mailbox=120
channel=20
callerid=121
mailbox=121
channel=21
callerid=122
mailbox=122
channel=22
callerid=123
mailbox=123
channel=23
callerid=124
mailbox=124
channel=24
callerid=125
mailbox=125
channel=25
callerid=126
mailbox=126
channel=26
callerid=127
mailbox=127
channel=27
callerid=128
mailbox=128
channel=28
callerid=129
mailbox=129
channel=29
callerid=130
mailbox=130
channel=30
callerid=131
mailbox=131
channel=31
callerid=132
mailbox=132
channel=32
callerid=133
mailbox=133
channel=33
callerid=134
mailbox=134
channel=34
callerid=135
mailbox=135
channel=35
callerid=136
mailbox=136
channel=36
callerid=137
mailbox=137
channel=37
callerid=138
mailbox=138
channel=38
callerid=139
mailbox=139
channel=39
callerid=140
mailbox=140
channel=40

group=5
callwaiting=no

signalling=fxo_ks
callerid=151
channel=41
callerid=152
channel=42
callerid=153
channel=43
callerid=154

context=incoming_9
callwaiting=no
threewaycalling=no
transfer=no
usecallerid=no
group=
signalling=fxs_ks
channel=45
channel=46
channel=47
channel=48

context=admin
immediate=no
signalling=fxo_ks
callerid=yes
callerid=190
threewaycalling=no
transfer=no
channel=97

context=bell
signalling=em
group=2
channel =49-96
group=
signalling=fxs_ks
channel=100


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RE: [Asterisk-Users] Queue_log multiple entries

2005-10-22 Thread David Craigon

Sorry, I've worked this out if anybody is scratching their heads on my
behalf. 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 David Craigon
 Sent: 21 October 2005 15:46
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] Queue_log multiple entries

 Hi there,
   I'm having trouble with queue_log. Whenever somebody 
 makes a queue call, I always get multiple rows in queue_log- 
 several ENTERQUEUE records. Then when somebody picks up, I 
 get one CONNECT and several ABANDONs. Anybody got any idea 
 why this could be? My queues are all set to ring all, and I'm 
 using Realtime configuration with MySQL.

   David

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[Asterisk-Users] Do the quantity of hardware timing devices go up as call volume increases?

2005-10-22 Thread Jason Lixfeld


Hi,

Is there any difference in the amount of hardware timing  
something like a Wildcard X100P can provide over something like a  
Wildcard TE411P?  If someone has a machine that pretty much just does  
very low volume MeetMe, Voicemail, SIP + IAX and 2 or 3 channels  
worth of codec translation at most, could they get by on a X100P?


Assuming an environment where this particular box gets it's PSTN  
channels from via an IAX trunk, what needs timing? Just meetme or are  
there other applications that would benefit from Zaptel hardware?

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[Asterisk-Users] CDMA card or module made for Asterisk?

2005-10-22 Thread Widyachacra Rajapaksha

dear friends,

what is de best CDMA card or module made for Asterisk?-- ---the path to freedom.--- 2.6.13-gentoo-r4
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Re: [Asterisk-Users] Do the quantity of hardware timing devices go up as call volume increases?

2005-10-22 Thread Andrew Kohlsmith

On Saturday 22 October 2005 14:07, Jason Lixfeld wrote:
  Is there any difference in the amount of hardware timing
 something like a Wildcard X100P can provide over something like a
 Wildcard TE411P?  If someone has a machine that pretty much just does
 very low volume MeetMe, Voicemail, SIP + IAX and 2 or 3 channels
 worth of codec translation at most, could they get by on a X100P?

Nope, an X100P timing source is just as good as a DS3000P timing source.  
There's been some argument that a software timer should work just fine, but 
nobody's stepped up and provided the patches and test cases.

 Assuming an environment where this particular box gets it's PSTN
 channels from via an IAX trunk, what needs timing? Just meetme or are
 there other applications that would benefit from Zaptel hardware?

Meetme's the big thing, I think the IAX2 trunk timing has been moved to a 
software timer now that the jitter buffer's got PLC and needs to be 
self-timed now anyway.

-A.
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Re: [Asterisk-Users] slow translations for ilbc and lpc10 on x86_64

2005-10-22 Thread Soner Tari

OK, yet another thread is closing, where I am the only poster :). For the 
record, I find out that starting asterisk with -p option (realtime) gives 
the following table, which now makes more sense. But what is surprising for 
me is that the load of the server was close to 0 in my original post, thus I 
would expect similar results. So I'll assume that the previous translation 
table (without -p) does not show the real capabilities of this server. 
Lesson learned: start asterisk with -p for close-to-actual translation 
values.


g723   gsm  ulaw  alaw  g726 adpcm  slin lpc10  g729 speex  ilbc
  g723 - - - - - - - - - - -
   gsm - - 2 2 2 2 1 3 - -13
  ulaw - 3 - 1 2 2 1 3 - -13
  alaw - 3 1 - 2 2 1 3 - -13
  g726 - 3 2 2 - 2 1 3 - -13
 adpcm - 3 2 2 2 - 1 3 - -13
  slin - 2 1 1 1 1 - 2 - -12
 lpc10 - 4 3 3 3 3 2 - - -14
  g729 - - - - - - - - - - -
 speex - - - - - - - - - - -
  ilbc - 4 3 3 3 3 2 4 - - -

Cheers,
Soner

- Original Message - 
From: Soner Tari [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: Thursday, October 20, 2005 9:34 PM
Subject: [Asterisk-Users] slow translations for ilbc and lpc10 on x86_64



Hi All,

When I do 'show translation' on a Linux asterisk 2.6.9-11.EL #1 Wed Jun 8 
16:40:06 CDT 2005 x86_64 x86_64 x86_64 GNU/Linux

and
Asterisk CVS HEAD built by [EMAIL PROTECTED] on a x86_64 running Linux on 
2005-10-19 19:07:08 UTC


I have very strange lpc10 and ilbc rows (sorry the columns are mixed up, I 
don't want to use html):


g723   gsm  ulaw  alaw  g726 adpcm  slin lpc10  g729 speex  ilbc
  g723 - - - - - - - - - - -
   gsm - - 2 2 2 2 1 3 - -14
  ulaw - 3 - 1 2 2 1 3 - -14
  alaw - 3 1 - 2 2 1 3 - -14
  g726 - 3 2 2 - 2 1 3 - -14
 adpcm - 3 2 2 2 - 1 3 - -14
  slin - 2 1 1 1 1 - 2 - -13
 lpc10 -484747474746 - - -59
  g729 - - - - - - - - - - -
 speex - - - - - - - - - - -
  ilbc -35343434343335 - - -

But, even on a Linux asterisk 2.6.9-11.EL #1 Wed Jun 8 16:59:52 CDT 2005 
i686 athlon i386 GNU/Linux

and
Asterisk CVS HEAD built by [EMAIL PROTECTED] on a i686 running Linux on 
2005-09-16 19:34:57 UTC

(which is a much slower machine, Athlon 2000+, 32bit anyway)
I have much better result on rows for iLBC and lpc10:

g723   gsm  ulaw  alaw  g726 adpcm  slin lpc10  g729 speex  ilbc
  g723 - - - - - - - - - - -
   gsm- - 2 2 4 2 1 5- -25
  ulaw- 5 - 1 4 2 1 5- -25
  alaw- 5 1 - 4 2 1 5- -25
  g726- 7 4 4 - 4 3 7- -27
 adpcm- 5 2 2 4 - 1 5- -25
  slin- 4 1 1 3 1 - 4- -24
 lpc10- 8 5 5 7 5 4 -- -28
  g729 - - - - - - - - - - -
 speex - - - - - - - - - - -
  ilbc- 8 5 5 7 5 4 8- - -

I've checked asterisk/codecs/ilbc and lpc10 dirs to see any optimizations 
for x86_64, but could not find anything.


Should I compile with some different flags? What could be wrong? Or x86_64 
is not a supported platform? Any pointers please?


Thanks in advance,
Soner

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Re: [Asterisk-Users] DTMF detection

2005-10-22 Thread Ryan

Hello all,
yes there is a lot of information about this on the wiki and in past posts on 
this list but have not found anything that has solved my problem.
setup is:
phone--PAP2-na--asterisk v1.0.9(in house on local subnet dtmf is 
inband)---PSTN---Telisipasterisk box at colo v1.0.9 VoIP only.  I have 
only access to dial up so can not go VoIP out of the house.
In extensions.conf  on colo * i have some logic that based on callerid lets me 
hit a single digit to get to DISA, this work every time.
the problem is that when i enter a number for DISA to dial i get duplicate 
digits, example i enter 6037862111 and disa tries to dial 6003778621.  I have 
tried setting relaxdtmf=yes in sip.conf with no luck.  I have read on the 
wiki that RFC2833 should work, but alas its a no go.  I am also using ulaw 
which should not be distorting the dtmf through compresion, correct? Also 
with RFC2833 it should not matter? Everything works great otherwise. sip.conf 
for colo * is posted below:

I just setup telasip and I'm having the same issue. I captured some RTP
packets and realized that when I get duplicate numbers it is because an
RTP packet has arrived out of order. In all my test cases it was just
one packet coming 1 packet too late, but the sequence number was
correct. It seems that * instead of putting the packets back in order
(using the seq numbers) makes a duplicate digit.

I'm not sure if this is a bug or not (I haven't read the rfc).

Thanks,
Ryan
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[Asterisk-Users] Modem Over IP: solutions ?

2005-10-22 Thread Jean-Michel Hiver


Hi,

I have a potential client who has legacy alarm systems which use modems 
to transmit encoded data to a remote location through the PSTN. They 
wish to replace the 'PSTN' bit with an IP link.


I am aware that it would be best if the data was transmitted directly 
over IP rather than modulated and then sent on the internet, but that is 
not possible because of the legacy equipment.


I was wondering if there was some specialized ATAs of some kind that 
would do TDMoIP and which could be used for this purpose?


Link latency is about 300ms with no more than 10ms jitter. If you have a 
solution please let me know!


Cheers,
Jean-Michel.

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Re: [Asterisk-Users] CDMA card or module made for Asterisk?

2005-10-22 Thread [EMAIL PROTECTED]


Widyachacra Rajapaksha wrote:


dear friends,

what is de best CDMA card or module made for Asterisk?

--
---

the path to freedom.
--- 2.6.13-gentoo-r4



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Hello,
Did u mean PCI cards?

~Madhawa

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Re: [Asterisk-Users] One SIP dead, all SIP dead -- sipmedia gone?

2005-10-22 Thread Eric \ManxPower\ Wieling


Jay Milk wrote:

I'm having the following recurring problem with asterisk:

When for any reason one of my SIP providers fails to register (i.e.
internet connection dropped), ALL SIP channels fail.  This means that,
for example, when my internet connection is out, none of my internal sip
phones register, and I'm unable to place outgoing calls (through IAX),
or to check voicemail.

Currently (and since yesterday evening), sipmedia.com/myphonecompany.com
is completely off the radar.  No DNS entry found -- not even a
name-server.  They've had this sort of massive failure before, but this
is one of the longest for all I can tell.  While that's a major problem,
it also meant that until I commented out the register =
sip.sipmedia.com statements, my entire phonesystem was unavailable.

1. Is there any way to get Asterisk to behave less absolute when one sip
registration fails?
2. Is anyone else experiencing the same sipmedia outtage, and/or has
information on when they'll be back?  Tech support seems affected, and
other direct numbers I have go into voicemail.


Asterisk handles DNS failures VERY badly.  Make sure your phones and 
Asterisk always refer to devices by IP, not by DNS name.

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Re: [Asterisk-Users] voip provider in a box

2005-10-22 Thread Paul Mahler

We have a turn-key solution available that does exactly what you are asking
for. You can reach someone for more information at 415.442.4010. 

TKS

Paul

[EMAIL PROTECTED]

 
 trixter aka Bret McDanel wrote:
 
 I am tasked with evaluating ready made solutions for a voip provider.
 Does anyone have any recommendations for software, specifically the
 environment will be a chargable voip provider (ie broadvoice, vonage,
 etc).  They wanted me to see what was there and write something if
 nothing they like exists.
 
 Thanks


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[Asterisk-Users] iax softphones

2005-10-22 Thread Hector medina

can anyone recomend a good iax softphone??___
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Re: [Asterisk-Users] Multiple instances of asterisk showing from 'ps aux'

2005-10-22 Thread BJ Weschke

strace?

valgrind?

Therearen't any profiling tools or the sort within the Asterisk suite that will deliver the information you're looking for about what each thread is doing. 
On 10/20/05, Jason Walker [EMAIL PROTECTED] wrote:





When I run 'ps aux' I get this:

root 964 0.0 0.4 47836 8280 ? S 00:02 0:00 asterisk -vvvg -croot 965 0.0 0.4 47836 8280 ? S 00:02 0:00 asterisk -vvvg -croot 967 
0.0 0.4 47836 8280 ? S 00:02 0:00 asterisk -vvvg -croot 975 0.0 0.4 47836 8280 ? S 00:02 0:00 asterisk -vvvg -croot 982 0.0 0.4 47836 8280 ? S 00:02 0:00 asterisk -vvvg -c
root 984 0.0 0.4 47836 8280 ? S 00:02 0:12 asterisk -vvvg -croot 986 0.0 0.4 47836 8280 ? S 00:02 0:00 asterisk -vvvg -croot 987 0.1 0.4 47836 8280 ? S 00:02 1:10 asterisk -vvvg -c
root 988 0.1 0.4 47836 8280 ? S 00:02 1:24 asterisk -vvvg -croot 989 0.0 0.4 47836 8280 ? S 00:02 0:00 asterisk -vvvg -croot 993 0.0 0.4 47836 8280 ? S 00:02 0:00 asterisk -vvvg -c
root 994 0.0 0.4 47836 8280 ? S 00:02 0:17 asterisk -vvvg -croot 996 0.0 0.4 47836 8280 ? S 00:02 0:00 asterisk -vvvg -croot 997 0.0 0.4 47836 8280 ? S 00:02 0:02 asterisk -vvvg -c
root 24202 1.2 0.4 47836 8280 ? S 09:04 6:52 asterisk -vvvg -croot 29417 1.6 0.4 47836 8280 ? S 11:07 6:54 asterisk -vvvg -croot 6555 1.0 0.4 47836 8280 ? S 14:44 2:04 asterisk -vvvg -c
root 8463 1.1 0.4 47836 8280 ? S 15:29 1:53 asterisk -vvvg -croot 14405 1.0 0.4 47836 8280 ? S 17:47 0:15 asterisk -vvvg -c

My question is, why are there 21 instances of asterisk running?

I understand the concept of a multi-threaded app in Linux (such as httpd). I am just looking for possible avenues and explanations of where I could look to figure out what each instance (or some of the instances) are actually doing.


* 1.0.9; FC1 

Thanks in advance

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[Asterisk-Users] iax softphone

2005-10-22 Thread Hector medina

can someone tell me about a good iax softphone ??

thanks
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Re: [Asterisk-Users] TDMoE and Badness in Kernel

2005-10-22 Thread asterisk groups

On Fri, 2005-10-21 at 09:26 -0700, trixter aka Bret McDanel wrote:
 On Fri, 2005-10-21 at 09:27 -0700, [EMAIL PROTECTED] wrote:
  I received some postings back, as I was trying to do the same thing.
  
  it' is a problem with Kernel 2.6... 2.4 works fine .. this is the summary
  I got from reading the posts before.
  
  I hope that helps... I dont have the ability to go DOWn in kernel to 2.4..
  
 
 the wiki suggested that it was a problem with softirq.c in the kernel
 and that this was fixed at some point.  What 2.6 version are you running
 that you have this problem?

I've seen this on just about everything 2.6.9 and above and up to
2.6.13.

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Re: [Asterisk-Users] One SIP dead, all SIP dead -- sipmedia gone?

2005-10-22 Thread Joe Greco

 Thanks for the wealth of information.  I knew they were off-air
 DNS-wise, and this happened before a couple of times.  It's just bad
 juju to have all your IPs in one block.

Actually, they didn't have all their DNS servers in one block.

It's also a fallacy that having DNS servers in a single block (or, 
worse, sequentially numbered) isnecessarily a bad thing.  For a long 
while, we ran with sequentially numbered servers that were in 
completely different cities, thanks to the magic of OSPF and not using
Ethernet IP addresses as service addresses.  There are arguments for 
and against certain kinds of diversity, of course, all of which have to
do with the available failure modes.

However, in this case, both their networks are dead, and with only two
name servers, that's zero for two, and of course that /will/ be a bad
thing.

 I don't think they were reselling, and I actually thought I had a pretty
 good report with them -- just have been unable to get anyone on the
 phone today.  Other than that, had several DIDs with them at $5/month,
 and really haven't seen major issues other than DTMF not working -- but
 then, they are SIP-only.  But between pricing and availability, they've
 been the best provider out of all I tried (Vonage, Broadvoice,
 voicepulse, iax.cc, ... )

Best price is occasionally a bad sign.

... JG
-- 
Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net
We call it the 'one bite at the apple' rule. Give me one chance [and] then I
won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN)
With 24 million small businesses in the US alone, that's way too many apples.
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Re: [Asterisk-Users] iax softphone

2005-10-22 Thread Time Bandit

 can someone tell me about a good iax softphone ??
Shameless plug : http://www.marccharbonneau.com/asterisk/mediaxphone.php

works only on windows


for one that works on Windows and Linux :
http://iaxclient.sourceforge.net/iaxcomm/index.html

there is also DIAX : http://www.laser.com/dante/diax/diax.html
and idefisk : http://www.asteriskguru.com/tools/idefisk_beta.php

hth
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Re: [Asterisk-Users] iax softphone



Idefisk is currently only for windows, but a native linux version is
nearly ready and will be released soon,
others i can also recommend
:
- iaxphone by ipsando
- firefly by virbiage.

Time Bandit wrote:


can someone tell me about a good iax softphone ??



Shameless plug : http://www.marccharbonneau.com/asterisk/mediaxphone.php

works only on windows


for one that works on Windows and Linux :
http://iaxclient.sourceforge.net/iaxcomm/index.html

there is also DIAX : http://www.laser.com/dante/diax/diax.html
and idefisk : http://www.asteriskguru.com/tools/idefisk_beta.php

hth
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[Asterisk-Users] Problem with asterisk 1.0.9 and sip and dtmf

2005-10-22 Thread Mike Bernson


I have asterisk running with sipura 3000 connect to PSTN and
sipura 2000 connected to phones.

On inbound calls I am getting what sounds like DTMF tone when
someone is talking on the PSTN side of the phone. It sound like
someone is hitting key on the phone while talking.

Is there any way to stop this from happing.

Here is the PSTN and one ext from the sip.conf

PSTN line
[199]
username=
type=friend
secret=
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5061
nat=never
[EMAIL PROTECTED]
host=dynamic
dtmfmode=rfc2833
context=incomming
canreinvite=no

ext
[206]
username=
type=friend
secret=
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5061
nat=never
[EMAIL PROTECTED]
host=dynamic
dtmfmode=rfc2833
context=from-sip
canreinvite=no



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RE: [Asterisk-Users] Modem Over IP: solutions ?

2005-10-22 Thread Carlos

Hey jean,

I have had to deal with the same situation many times what we have come to
the conclusion is you can actully get them to work only under the 4-2
signaling that the alarm companys use.  Just about all alarms now are set to
use contact id which we have found out that we send the data ok but we don't
get a kiss back telling the alarm to silent.  Note the different between 4-2
and contact id are 4-2 basically calls and says there is a issue here.
Contact id will call and say there is an issue on the 4th floor room 410
closet.  Basically with contact id you get spacifics of the issue.  Note set
the alarm speed to as low as possible.  And in some cases the alarm comapnys
are ass's and wont change it to 4-2 because they have to re id everything.
Oh yea and when testing make sure you call the alarm company and put it in
test mode the fire dept dosnt exactly like it when the get false calls.

Carlos Alcantar
Race Technologies, Inc.
101 Haskins Way
South San Francisco, CA 94080
P: 650.246.8900
F: 650.246.8901
E: [EMAIL PROTECTED]
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel
Hiver
Sent: Saturday, October 22, 2005 12:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Commercial and
Business-Oriented Asterisk Discussion
Subject: [Asterisk-Users] Modem Over IP: solutions ?

Hi,

I have a potential client who has legacy alarm systems which use modems to
transmit encoded data to a remote location through the PSTN. They wish to
replace the 'PSTN' bit with an IP link.

I am aware that it would be best if the data was transmitted directly over
IP rather than modulated and then sent on the internet, but that is not
possible because of the legacy equipment.

I was wondering if there was some specialized ATAs of some kind that would
do TDMoIP and which could be used for this purpose?

Link latency is about 300ms with no more than 10ms jitter. If you have a
solution please let me know!

Cheers,
Jean-Michel.

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Re: [Asterisk-Users] emacs syntax/keyowrd highlighting: asterisk-mode

2005-10-22 Thread Dave Grey



On Oct 22, 2005, at 12:18 AM, Dave Grey wrote:


On Oct 21, 2005, at 5:50 PM, JP Carballo wrote:

Dave Grey wrote:
 I hacked together an emacs general/minor mode for basic font- 
locking (syntax shading) support. Feel free to grab it here:

http://homepage.mac.com/lydanynom/asterisk-mode.el.zip


Good work Dave!
I suggest you post this in www.voip-info.org for future emacs/ 
asterisk users.


Thanks, JP.  That's what I intended to do, but I wanted to give it  
a little time for refinement before I threw it up there.  If I  
don't find, and no one points out to me, anything glaringly  
unworkable in a week or so I will post it.


Speaking of glaringly unworkable, like a numb-skull I edited and  
tested the thing with default my green-on-black color scheme.  I  
happened to open something up in a raw black-on-white xterm and  
realized that I had created a nightmare.  I have made the appropriate  
changes, so if you looked at it and said,  Hey, this is  
horrible..., then my apologies and give it another look.  Same url  
and filename above, updated version noted in the comments.


lyd

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[Asterisk-Users] Re: [Asterisk-biz] Modem Over IP: solutions ?

2005-10-22 Thread alex

On Sat, 22 Oct 2005, Jean-Michel Hiver wrote:

 I have a potential client who has legacy alarm systems which use modems
 to transmit encoded data to a remote location through the PSTN. They
 wish to replace the 'PSTN' bit with an IP link.
 
 I am aware that it would be best if the data was transmitted directly
 over IP rather than modulated and then sent on the internet, but that is
 not possible because of the legacy equipment.
 
 I was wondering if there was some specialized ATAs of some kind that
 would do TDMoIP and which could be used for this purpose?
 
 Link latency is about 300ms with no more than 10ms jitter. If you have a
 solution please let me know!
No.

Terminate the connection on the remote side. Equipment such as Lucent MAX 
to do that is a dime a dozen now (4-port max6000 is ~200$)

-alex


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Re: [Asterisk-Users] iax softphone

2005-10-22 Thread Matt Florell

We use the Firefly ThirdParty softphone on our windows laptops. It's
free, easy to configure and will do IAX2 and SIP:
http://www.virbiage.com/firefly/download/firefly-thirdparty.exe

MATT---



On 10/22/05, Zoa [EMAIL PROTECTED] wrote:

 Idefisk is currently only for windows, but a native linux version is
 nearly ready and will be released soon,
 others i can also recommend
 :
 - iaxphone by ipsando
 - firefly by virbiage.

 Time Bandit wrote:

 can someone tell me about a good iax softphone ??
 
 
 Shameless plug : http://www.marccharbonneau.com/asterisk/mediaxphone.php
 
 works only on windows
 
 
 for one that works on Windows and Linux :
 http://iaxclient.sourceforge.net/iaxcomm/index.html
 
 there is also DIAX : http://www.laser.com/dante/diax/diax.html
 and idefisk : http://www.asteriskguru.com/tools/idefisk_beta.php
 
 hth
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[Asterisk-Users] Unable to negotiate codec???

2005-10-22 Thread clint_in_sydney




Hi All,I get the following when trying to dial in to my asterisk 
box.Oct 22 13:58:11 WARNING[3599]: Don't know any of 0xf800 
formatsOct 22 13:58:11 ERROR[3599]: No best format in 0xf800???Oct 22 
13:58:11 NOTICE[3599]: Rejected connect attempt from 203.98.83.19, 
requested/capability 0x200/0xfe00 incompatible with our capability 
0xf900.and I get the following when I try to dial out.Oct 22 
14:06:31 WARNING[3599]: Call rejected by 203.98.83.19: Unable to negotiate 
codecI'm using a brand new g729 codec from Digium.Any ideas on 
what my problem might be?Cheers,Clint 
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[Asterisk-Users] redirecting incoming calls to external phone (cell)

2005-10-22 Thread Jay Christopherson


Hi-

I am attempting to setup Asterisk for the first time, and I think I am about 
99% there.   I am using vonage softphone, and want to use asterisk to 
redirect incoming calls to my cell phone primarily, and maybe other remote 
lines.


Right now, I am able to register with vonage, and trap incoming calls.  The 
only issue I have is that I don't think my syntax in extensions.conf is 
correct to dial out to my cell:


XX -- my vonage softphone number
YY- my cell phone number


sip.conf:
[sphone.vopr.vonage.net]
username=1XX
port=5060
nat=yes
type=friend
secret=X
host=sphone.vopr.vonage.net
fromuser=1XX
fromdomain=sphone.vopr.vonage.net
dtmfmode=rfc2833
auth=md5
canreinvite=no
context=out
;
[vonage-in]
username=1XX
type=friend
port=5060
nat=yes
secret=21V9bkQ5MR
host=sphone.vopr.vonage.net
insecure=very
fromuser=1XX
fromdomain=sphone.vopr.vonage.net
context=in
canreinvite=no
auth=md5

extensions.conf
[in]
;exten = s,1,Dial(SIP/1YY,25,rt)
;exten = s,2,Ringing()
;exten = s,3,Wait(60)
exten = _1XX,1,dial(sip/1YY,20,r)

[out]
exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED],25,rt)

Right now, when I get an incoming, this is the message I get:

From: ZZZ-ZZZ- 
sip:[EMAIL PROTECTED]:5061;user=phone;tag=1939305037

To: sip:[EMAIL PROTECTED]:5061;user=phone
Call-ID: 
[EMAIL PROTECTED]

CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0

to 216.115.25.198:5060
   -- Executing Dial(SIP/1XX-6e16, sip/1YY|20|r) in new 
stack
Oct 23 00:58:20 WARNING[6933]: chan_sip.c:1401 create_addr: No such host: 
1YY

Destroying call '[EMAIL PROTECTED]'
Oct 23 00:58:20 NOTICE[6933]: app_dial.c:764 dial_exec: Unable to create 
channel of type 'sip'

 == Everyone is busy/congested at this time
Oct 23 00:58:30 WARNING[6933]: pbx.c:1948 ast_pbx_run: Timeout, but no rule 
't' in context 'in'


Thanks-
Jay


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RE: [Asterisk-Users] Unable to negotiate codec???




What codec are you using on the client and the server? From 
my understanding, you have to have a license for both ends of the G.729 call. 
Are you passing this through one server to another and the call is being 
rejected at the server level?


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
clint_in_sydneySent: Saturday, October 22, 2005 5:44 
PMTo: Asterisk Users Mailing List - Non-Commercial 
DiscussionSubject: [Asterisk-Users] Unable to negotiate 
codec???

Hi All,I get the following when trying to dial in to my asterisk 
box.Oct 22 13:58:11 WARNING[3599]: Don't know any of 0xf800 
formatsOct 22 13:58:11 ERROR[3599]: No best format in 0xf800???Oct 22 
13:58:11 NOTICE[3599]: Rejected connect attempt from 203.98.83.19, 
requested/capability 0x200/0xfe00 incompatible with our capability 
0xf900.and I get the following when I try to dial out.Oct 22 
14:06:31 WARNING[3599]: Call rejected by 203.98.83.19: Unable to negotiate 
codecI'm using a brand new g729 codec from Digium.Any ideas on 
what my problem might be?Cheers,Clint 
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[Asterisk-Users] IAX registration with FWD and Teliax - Lost

2005-10-22 Thread Joseph

Earlier today I lost registration with FWD, now Teliax registration is
down as well.
I can ping both networks but can not register.

Can anybody check on their end?

-- 
#Joseph
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Re: [Asterisk-Users] IAX registration with FWD and Teliax - Lost


 Earlier today I lost registration with FWD, now Teliax registration is
 down as well.
 I can ping both networks but can not register.
 
 Can anybody check on their end?

Of the two servers:
 host=voip-co1.teliax.com
 host=voip-co2.teliax.com  

the co2 was not processing any calls a few minutes ago, but co1 server
is just fine. (at 9pm CDT)



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Re: [Asterisk-Users] Goiax.com DID not working anymore?

2005-10-22 Thread Blake Krone

Good lord, it's part asterisk part goiax.com

If you have an issue with it ignore the thread. At first I thought I had asterisk config'd wrong.

Now I know better than to waste my time with a list that has people like you on it.On 10/21/05, Robert Webb
[EMAIL PROTECTED] wrote:On Fri, 21 Oct 2005 10:25:59 -0400Paul
[EMAIL PROTECTED] wrote: Kanuri, Seshu (Company IT) wrote:[EMAIL PROTECTED] wrote
It's a free service. It belongs on this list.Olle is right. Even if it is a free service it does notbelong here.This forum is for any Asterisk related user issues, not
some DID issueof one of a hundred such service providers.Take it off this list. Now that makes 2 of you who are wrong.
Goiax.com isproviding a valuable free service to asterisk users. Forone thing it enables users to do some free testing ofPSTN-asterisk setup. I believe the posters to thisthread are likely 100% asterisk users so what is so bad
about using the asterisk users mailing list fordiscussion? There are lots of unwarranted posts to all the listsfrom the totally clueless. Why don't you pick on theminstead?
No, this belongs on the asterisk-biz list as this is anissue of business practice not an operational issue of theAsterisk software itself.The -users list is for those that are having issues with
getting Asterisk up and running or trying to figure outhow to do certain software realated tasks or scripting.Can you not comprehend the difference??___--Bandwidth and Colocation sponsored by
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[Asterisk-Users] Satellite receiver over IP

2005-10-22 Thread Chris Mason

I need my satellite receivers to call home to avoid problems with the 
service. I have hooked them up through an IAXY and tried a SPA2002 set 
to G711 and made sure the transport is 711 all the way. However, it does 
not work at all, the receivers cannot make the connection work.

Has anyone made this work?

Chris Mason
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[Asterisk-Users] Unable to negotiate codec???

2005-10-22 Thread clint_in_sydney




I use IAX and have a license for G729 at my end and 
OZTell, my provider, use G729 as their main codec.

My box rejects connections from my provider due to 
incompatible codecs and vice versa.

I'm waiting for them to get back to me on 
this.

Clint.



  - Original Message - 
  From: 
  Jason 
  Walker 
  To: 'Asterisk Users Mailing List - 
  Non-Commercial Discussion' 
  Sent: Sunday, October 23, 2005 10:52 
  AM
  Subject: [other] RE: [Asterisk-Users] 
  Unable to negotiate codec???
  
  What codec are you using on the client and the server? 
  From my understanding, you have to have a license for both ends of the G.729 
  call. Are you passing this through one server to another and the call is being 
  rejected at the server level?
  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of 
  clint_in_sydneySent: Saturday, October 22, 2005 5:44 
  PMTo: Asterisk Users Mailing List - Non-Commercial 
  DiscussionSubject: [Asterisk-Users] Unable to negotiate 
  codec???
  
  Hi All,I get the following when trying to dial in to my asterisk 
  box.Oct 22 13:58:11 WARNING[3599]: Don't know any of 0xf800 
  formatsOct 22 13:58:11 ERROR[3599]: No best format in 0xf800???Oct 22 
  13:58:11 NOTICE[3599]: Rejected connect attempt from 203.98.83.19, 
  requested/capability 0x200/0xfe00 incompatible with our capability 
  0xf900.and I get the following when I try to dial out.Oct 22 
  14:06:31 WARNING[3599]: Call rejected by 203.98.83.19: Unable to negotiate 
  codecI'm using a brand new g729 codec from Digium.Any ideas on 
  what my problem might be?Cheers,Clint
  
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Re: [Asterisk-Users] IAX registration with FWD and Teliax - Lost

2005-10-22 Thread Joseph

On Sat, 2005-10-22 at 21:01 -0600, Rich Adamson wrote:
  Earlier today I lost registration with FWD, now Teliax registration is
  down as well.
  I can ping both networks but can not register.
  
  Can anybody check on their end?
 
 Of the two servers:
  host=voip-co1.teliax.com
  host=voip-co2.teliax.com  
 
 the co2 was not processing any calls a few minutes ago, but co1 server
 is just fine. (at 9pm CDT)

You are right, I change registration to co1 and it went through.  So the
problem seems to be on their end.
Now, FWD needs to fix something on their end.  I've tired
iax2.fwdnet.net but it is not going through.
Do they have any other alternative?

-- 
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RE: [Asterisk-Users] Unable to negotiate codec???







From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
clint_in_sydneySent: Saturday, October 22, 2005 7:15 
PMTo: Asterisk Users Mailing List - Non-Commercial 
DiscussionSubject: [Asterisk-Users] Unable to negotiate 
codec???

I use IAX and have a license for G729 at my end and 
OZTell, my provider, use G729 as their main codec.

My box rejects connections from my provider due to 
incompatible codecs and vice versa.

I'm waiting for them to get back to me on 
this.

Clint.



  - Original Message - 
  From: 
  Jason 
  Walker 
  To: 'Asterisk Users Mailing List - 
  Non-Commercial Discussion' 
  Sent: Sunday, October 23, 2005 10:52 
  AM
  Subject: [other] RE: [Asterisk-Users] 
  Unable to negotiate codec???
  
  What codec are you using on the client and the server? 
  From my understanding, you have to have a license for both ends of the G.729 
  call. Are you passing this through one server to another and the call is being 
  rejected at the server level?
  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of 
  clint_in_sydneySent: Saturday, October 22, 2005 5:44 
  PMTo: Asterisk Users Mailing List - Non-Commercial 
  DiscussionSubject: [Asterisk-Users] Unable to negotiate 
  codec???
  
  Hi All,I get the following when trying to dial in to my asterisk 
  box.Oct 22 13:58:11 WARNING[3599]: Don't know any of 0xf800 
  formatsOct 22 13:58:11 ERROR[3599]: No best format in 0xf800???Oct 22 
  13:58:11 NOTICE[3599]: Rejected connect attempt from 203.98.83.19, 
  requested/capability 0x200/0xfe00 incompatible with our capability 
  0xf900.and I get the following when I try to dial out.Oct 22 
  14:06:31 WARNING[3599]: Call rejected by 203.98.83.19: Unable to negotiate 
  codecI'm using a brand new g729 codec from Digium.Any ideas on 
  what my problem might be?Cheers,Clint
  
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RE: [Asterisk-Users] Unable to negotiate codec???




Sorry for the blank response - 
before...

From your output below, what looks weird are the hex values 
for the codecs:

[snip]
requested/capability 
0x200/0xfe00 incompatible with our capability 0xf900.

From one of my servers, when I do a 'show codecs' on the 
console, I get 

sfsip01*CLI show codecsDisclaimer: this command is 
for informational purposes only. 
It does not indicate anything about your 
configuration. 
INT BINARY 
HEX TYPE NAME 
DESC 
1 (1  0) (0x1) 
audio g723 
(G.723.1) 2 (1 
 1) (0x2) 
audio gsm 
(GSM) 4 (1 
 2) (0x4) 
audio ulaw (G.711 
u-law) 8 (1 
 3) (0x8) 
audio alaw (G.711 
A-law) 16 (1  
4) (0x10) audio g726 
(G.726) 32 (1  
5) (0x20) audio adpcm 
(ADPCM) 64 (1  
6) (0x40) audio slin 
(16 bit Signed Linear PCM) 128 (1 
 7) (0x80) audio 
lpc10 (LPC10) 256 (1 
 8) (0x100) audio 
g729 (G.729A) 512 (1 
 9) (0x200) audio 
speex (SpeeX) 1024 (1 
 10) (0x400) audio 
ilbc (iLBC) 65536 (1  
16) (0x1) image jpeg (JPEG 
image) 131072 (1  17) (0x2) 
image png (PNG 
image) 262144 (1  18) (0x4) 
video h261 (H.261 
Video) 524288 (1  19) (0x8) 
video h263 (H.263 Video)




0x200 would be speex. G.729 - in hex, from this display - would be 0x100. 
>From your output, I don't see 0x100 at all. Am I confused?



From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
clint_in_sydneySent: Saturday, October 22, 2005 7:15 
PMTo: Asterisk Users Mailing List - Non-Commercial 
DiscussionSubject: [Asterisk-Users] Unable to negotiate 
codec???

I use IAX and have a license for G729 at my end and 
OZTell, my provider, use G729 as their main codec.

My box rejects connections from my provider due to 
incompatible codecs and vice versa.

I'm waiting for them to get back to me on 
this.

Clint.



  - Original Message - 
  From: 
  Jason 
  Walker 
  To: 'Asterisk Users Mailing List - 
  Non-Commercial Discussion' 
  Sent: Sunday, October 23, 2005 10:52 
  AM
  Subject: [other] RE: [Asterisk-Users] 
  Unable to negotiate codec???
  
  What codec are you using on the client and the server? 
  From my understanding, you have to have a license for both ends of the G.729 
  call. Are you passing this through one server to another and the call is being 
  rejected at the server level?
  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of 
  clint_in_sydneySent: Saturday, October 22, 2005 5:44 
  PMTo: Asterisk Users Mailing List - Non-Commercial 
  DiscussionSubject: [Asterisk-Users] Unable to negotiate 
  codec???
  
  Hi All,I get the following when trying to dial in to my asterisk 
  box.Oct 22 13:58:11 WARNING[3599]: Don't know any of 0xf800 
  formatsOct 22 13:58:11 ERROR[3599]: No best format in 0xf800???Oct 22 
  13:58:11 NOTICE[3599]: Rejected connect attempt from 203.98.83.19, 
  requested/capability 0x200/0xfe00 incompatible with our capability 
  0xf900.and I get the following when I try to dial out.Oct 22 
  14:06:31 WARNING[3599]: Call rejected by 203.98.83.19: Unable to negotiate 
  codecI'm using a brand new g729 codec from Digium.Any ideas on 
  what my problem might be?Cheers,Clint
  
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Re: [Asterisk-Users] Satellite receiver over IP

2005-10-22 Thread Tom Vile

Try changing at least on the sipura the RTP Packet Size: to 0.020 or 0.010
it should be under the admin login and then the SIP tab.On 10/22/05, Chris Mason [EMAIL PROTECTED]
 wrote:I need my satellite receivers to call home to avoid problems with the
service. I have hooked them up through an IAXY and tried a SPA2002 setto G711 and made sure the transport is 711 all the way. However, it doesnot work at all, the receivers cannot make the connection work.Has anyone made this work?
Chris Mason___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list
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http://lists.digium.com/mailman/listinfo/asterisk-users-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephony
www.baldwintechsolutions.comPhone: 518-631-2855 x205Phone: 845-652-4578 x205Phone: 978-203-3848 x205Fax: 518-631-2856
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Re: [Asterisk-Users] iax softphone

2005-10-22 Thread Tom Vile

Idefisk for me. I love how it does not clutter the screen and it works.On 10/22/05, Matt Florell [EMAIL PROTECTED]
 wrote:We use the Firefly ThirdParty softphone on our windows laptops. It's
free, easy to configure and will do IAX2 and SIP:http://www.virbiage.com/firefly/download/firefly-thirdparty.exeMATT---
On 10/22/05, Zoa [EMAIL PROTECTED] wrote: Idefisk is currently only for windows, but a native linux version is nearly ready and will be released soon,
 others i can also recommend : - iaxphone by ipsando - firefly by virbiage. Time Bandit wrote: can someone tell me about a good iax softphone ?? 
  Shameless plug : http://www.marccharbonneau.com/asterisk/mediaxphone.php  works only on windows
   for one that works on Windows and Linux : http://iaxclient.sourceforge.net/iaxcomm/index.html 
 there is also DIAX : http://www.laser.com/dante/diax/diax.html and idefisk : http://www.asteriskguru.com/tools/idefisk_beta.php
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-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephonywww.baldwintechsolutions.com
Phone: 518-631-2855 x205Phone: 845-652-4578 x205Phone: 978-203-3848 x205Fax: 518-631-2856
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RE: [Asterisk-Users] iax softphone




Tom - do you end up with that phone shutting down with 
an error on Windows XP? I downloaded the latest. After about 3 minutes on a 
call, the other end can no longer hear me and then the phone just 
dies.


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Tom 
VileSent: Saturday, October 22, 2005 8:21 PMTo: Asterisk 
Users Mailing List - Non-Commercial DiscussionSubject: Re: 
[Asterisk-Users] iax softphone
Idefisk for me. I love how it does not clutter the screen and 
it works.
On 10/22/05, Matt 
Florell [EMAIL PROTECTED]  
wrote:
We 
  use the Firefly ThirdParty softphone on our windows laptops. It'sfree, 
  easy to configure and will do IAX2 and SIP:http://www.virbiage.com/firefly/download/firefly-thirdparty.exeMATT---On 
  10/22/05, Zoa [EMAIL PROTECTED] 
  wrote: Idefisk is currently only for windows, but a native 
  linux version is nearly ready and will be released soon,  
  others i can also recommend : - iaxphone by ipsando - 
  firefly by virbiage. Time Bandit wrote: 
  can someone tell me about a good iax softphone ??  
Shameless plug : http://www.marccharbonneau.com/asterisk/mediaxphone.php 
   works only on windows   
  for one that works on Windows and Linux : http://iaxclient.sourceforge.net/iaxcomm/index.html 
   there is also DIAX : http://www.laser.com/dante/diax/diax.html 
  and idefisk : http://www.asteriskguru.com/tools/idefisk_beta.php 
hth 
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  -- Tom VileBaldwin 
Technology Solutions, IncConsulting - Web Design - VoIP Telephonywww.baldwintechsolutions.com 
Phone: 518-631-2855 x205Phone: 845-652-4578 x205Phone: 
978-203-3848 x205Fax: 518-631-2856 
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Re: [Asterisk-Users] DTMF detection

2005-10-22 Thread Ryan

On Sat, Oct 22, 2005 at 12:53:39PM -0600, Ryan exclaimed:

snip

I just setup telasip and I'm having the same issue. I captured some RTP
packets and realized that when I get duplicate numbers it is because an
RTP packet has arrived out of order. In all my test cases it was just
one packet coming 1 packet too late, but the sequence number was
correct. It seems that * instead of putting the packets back in order
(using the seq numbers) makes a duplicate digit.

I'm not sure if this is a bug or not (I haven't read the rfc).


I found this in mantis at: http://bugs.digium.com/view.php?id=4659
Unfortunately this will require upstream providers to patch asterisk
before this will work (which will happen over time).
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[Asterisk-Users] Asterisk CDR records when a call is transferred

2005-10-22 Thread Brad .

I have an Asterisk server that peers with a VoiP provider via IAX2. I have 
10 local SIP users.


I record the CDR data into a MySQL database, and use that to bill the 10 
local SIP users.


The problem I have is, one of my local users (User 5, for example) has there 
handset forwarded to a mobile phone number (which is fine, I have no problem 
with that). But the problem is, when another local user (User 8, for 
example) calls User 5, the call to the mobile number in the MySQL CDR is 
recorded with the accountcode and src of User 8, instead of User 5.


So, the caller (User 8) gets billed for the call to the mobile, instead of 
the user (User 5) who's handset was forwarded to the mobile.


Can anyone tell me how I can get around this?

Thanks very much

BJ

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Re: [Asterisk-Users] CDMA card or module made for Asterisk?

2005-10-22 Thread Widyachacra Rajapaksha

sure...On 10/23/05, [EMAIL PROTECTED] [EMAIL PROTECTED]
 wrote:Widyachacra Rajapaksha wrote: dear friends, what is de best CDMA card or module made for Asterisk?
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Re: [Asterisk-Users] iax softphone



Jason, i didn't hear about that problem before (several thousand people
are using that version), could you please send a copy of your config
files + the exact version and language localisation of windows xp to
[EMAIL PROTECTED]
Does it happen with one specific version of asterisk ?

Whatever the problem is, it should not be there. Please help us find the
bug.

Joachim.

Jason Walker wrote:


Tom - do you end up with that phone shutting down with an error on
Windows XP? I downloaded the latest. After about 3 minutes on a call,
the other end can no longer hear me and then the phone just dies.


*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of *Tom Vile
*Sent:* Saturday, October 22, 2005 8:21 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [Asterisk-Users] iax softphone

Idefisk for me.  I love how it does not clutter the screen and it works.

On 10/22/05, *Matt Florell* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:

We use the Firefly ThirdParty softphone on our windows laptops. It's
free, easy to configure and will do IAX2 and SIP:
http://www.virbiage.com/firefly/download/firefly-thirdparty.exe

MATT---



On 10/22/05, Zoa [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:

 Idefisk is currently only for windows, but a native linux version is
 nearly ready and will be released soon,
 others i can also recommend
 :
 - iaxphone by ipsando
 - firefly by virbiage.

 Time Bandit wrote:

 can someone tell me about a good iax softphone ??
 
 
 Shameless plug :
http://www.marccharbonneau.com/asterisk/mediaxphone.php
 
 works only on windows
 
 
 for one that works on Windows and Linux :
 http://iaxclient.sourceforge.net/iaxcomm/index.html
 
 there is also DIAX : http://www.laser.com/dante/diax/diax.html
 and idefisk :
http://www.asteriskguru.com/tools/idefisk_beta.php
http://www.asteriskguru.com/tools/idefisk_beta.php
 
 hth
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www.baldwintechsolutions.com http://www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Phone: 845-652-4578 x205
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Re: [Asterisk-Users] iax softphone

2005-10-22 Thread Tom Vile

Nope, I do not have that issue.On 10/23/05, Jason Walker [EMAIL PROTECTED] wrote:





Tom - do you end up with that phone shutting down with 
an error on Windows XP? I downloaded the latest. After about 3 minutes on a 
call, the other end can no longer hear me and then the phone just 
dies.


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED]] On Behalf Of Tom 
VileSent: Saturday, October 22, 2005 8:21 PMTo: Asterisk 
Users Mailing List - Non-Commercial DiscussionSubject: Re: 
[Asterisk-Users] iax softphone
Idefisk for me. I love how it does not clutter the screen and 
it works.
On 10/22/05, Matt 
Florell [EMAIL PROTECTED]  
wrote:
We 
  use the Firefly ThirdParty softphone on our windows laptops. It'sfree, 
  easy to configure and will do IAX2 and SIP:http://www.virbiage.com/firefly/download/firefly-thirdparty.exe
MATT---On 
  10/22/05, Zoa [EMAIL PROTECTED] 
  wrote: Idefisk is currently only for windows, but a native 
  linux version is nearly ready and will be released soon,  
  others i can also recommend : - iaxphone by ipsando - 
  firefly by virbiage. Time Bandit wrote: 
  can someone tell me about a good iax softphone ??  
Shameless plug : http://www.marccharbonneau.com/asterisk/mediaxphone.php
 
   works only on windows   
  for one that works on Windows and Linux : http://iaxclient.sourceforge.net/iaxcomm/index.html
 
   there is also DIAX : http://www.laser.com/dante/diax/diax.html 
  and idefisk : http://www.asteriskguru.com/tools/idefisk_beta.php 
hth 
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  -- Tom VileBaldwin 
Technology Solutions, IncConsulting - Web Design - VoIP Telephonywww.baldwintechsolutions.com 
Phone: 518-631-2855 x205Phone: 845-652-4578 x205Phone: 
978-203-3848 x205Fax: 518-631-2856 

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http://lists.digium.com/mailman/listinfo/asterisk-users-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephony
www.baldwintechsolutions.comPhone: 518-631-2855 x205Phone: 845-652-4578 x205Phone: 978-203-3848 x205Fax: 518-631-2856
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RE: [Asterisk-Users] One SIP dead, all SIP dead -- sipmedia gone?

 -Original Message-
 From: Joe Greco [mailto:[EMAIL PROTECTED] 
 
  juju to have all your IPs in one block.
 
 Actually, they didn't have all their DNS servers in one block.

Touche... I didn't even double-check, just assumed.
 
 while, we ran with sequentially numbered servers that were in 
 completely different cities, thanks to the magic of OSPF and 

True.

  I don't think they were reselling, and I actually thought I had a 
 ...
  availability, they've been the best provider out of all I tried 
  (Vonage, Broadvoice, voicepulse, iax.cc, ... )
 
 Best price is occasionally a bad sign.

Granted.  But I did say between pricing and availability.  Their BYOD
plan is $5/month and includes 60 outgoing minutes, unlimited incoming,
and sip-only access.  After explaining that I wanted several lines and a
discount, they asked me quite bluntly what kind of incoming volume I
expected; when they learned that I'm a residential user with minimal
volume, they allowed me to drop the included outgoing minutes and
adjusted their pricing south.  Seems fair and well-calculated to me.  I
could have gotten cheaper lines from the yokels at sixtel/iax.cc...
Well, if they ever fixed their instant 3-months ordering system.

For sipmedia, good response, quick set-up, proper fraud-protection...
All around good, just some recurring problems with their upstream
(bandwidth) providers from what I can tell.  They're reselling L3.
Three failures in seven months isn't great, and the response on this
last one was lacking -- but they're still doing better than the local
utility, which leaves us an average of eight hours/month without
electricity :)

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RE: [Asterisk-Users] Satellite receiver over IP

If that's dishnetwork and they keep charging you their $5 programming
access fee or whatever they call it, just plug it in and confirm that
you get a dial-tone.  Then call tech-support and have them adjust
billing -- all they check is that the receiver gets a dial-tone and they
take your word for it.

 -Original Message-
 From: Chris Mason [mailto:[EMAIL PROTECTED] 
 Sent: Saturday, October 22, 2005 9:12 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] Satellite receiver over IP

 I need my satellite receivers to call home to avoid problems with the 
 service. I have hooked them up through an IAXY and tried a 
 SPA2002 set 
 to G711 and made sure the transport is 711 all the way. 
 However, it does 
 not work at all, the receivers cannot make the connection 
 work. Has anyone made this work?

 Chris Mason
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RE: [Asterisk-Users] iax softphone

Done - 

Joachim, I cc'd you on the email so you could see what I sent.

Let me know if more info is needed. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zoa
Sent: Saturday, October 22, 2005 10:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] iax softphone


Jason, i didn't hear about that problem before (several thousand people are
using that version), could you please send a copy of your config files + the
exact version and language localisation of windows xp to
[EMAIL PROTECTED] Does it happen with one specific version of
asterisk ?

Whatever the problem is, it should not be there. Please help us find the
bug.

Joachim.

Jason Walker wrote:

 Tom - do you end up with that phone shutting down with an error on 
 Windows XP? I downloaded the latest. After about 3 minutes on a call, 
 the other end can no longer hear me and then the phone just dies.

 --
 --
 *From:* [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] *On Behalf Of *Tom 
 Vile
 *Sent:* Saturday, October 22, 2005 8:21 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [Asterisk-Users] iax softphone

 Idefisk for me.  I love how it does not clutter the screen and it works.

 On 10/22/05, *Matt Florell* [EMAIL PROTECTED] 
 mailto:[EMAIL PROTECTED] wrote:

 We use the Firefly ThirdParty softphone on our windows laptops. It's
 free, easy to configure and will do IAX2 and SIP:
 http://www.virbiage.com/firefly/download/firefly-thirdparty.exe

 MATT---



 On 10/22/05, Zoa [EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED] wrote:
 
  Idefisk is currently only for windows, but a native linux version is
  nearly ready and will be released soon,
  others i can also recommend
  :
  - iaxphone by ipsando
  - firefly by virbiage.
 
  Time Bandit wrote:
 
  can someone tell me about a good iax softphone ??
  
  
  Shameless plug :
 http://www.marccharbonneau.com/asterisk/mediaxphone.php
  
  works only on windows
  
  
  for one that works on Windows and Linux :
  http://iaxclient.sourceforge.net/iaxcomm/index.html
  
  there is also DIAX : http://www.laser.com/dante/diax/diax.html
  and idefisk :
 http://www.asteriskguru.com/tools/idefisk_beta.php
 http://www.asteriskguru.com/tools/idefisk_beta.php
  
  hth
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 --
 Tom Vile
 Baldwin Technology Solutions, Inc
 Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com 
 http://www.baldwintechsolutions.com
 Phone: 518-631-2855 x205
 Phone: 845-652-4578 x205
 Phone: 978-203-3848 x205
 Fax: 518-631-2856

---
-

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RE: [Asterisk-Users] iax softphone




Are you running on XP SP2just curious? How about the 
version of *?


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Tom 
VileSent: Saturday, October 22, 2005 10:03 PMTo: Asterisk 
Users Mailing List - Non-Commercial DiscussionSubject: Re: 
[Asterisk-Users] iax softphone
Nope, I do not have that issue.
On 10/23/05, Jason 
Walker [EMAIL PROTECTED] wrote:

  Tom - do 
  you end up with that phone shutting down with an error on Windows XP? I 
  downloaded the latest. After about 3 minutes on a call, the other end can no 
  longer hear me and then the phone just dies.
  
  
  From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of 
  Tom VileSent: Saturday, October 22, 2005 8:21 PMTo: 
  Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: 
  [Asterisk-Users] iax softphone
  
  Idefisk for me. I love how it does not clutter the screen and 
  it works.
  On 10/22/05, Matt 
  Florell [EMAIL PROTECTED]  
  wrote: 
  We 
use the Firefly ThirdParty softphone on our windows laptops. It'sfree, 
easy to configure and will do IAX2 and SIP:http://www.virbiage.com/firefly/download/firefly-thirdparty.exe 
MATT---On 10/22/05, Zoa [EMAIL PROTECTED] wrote: Idefisk is 
currently only for windows, but a native linux version is nearly 
ready and will be released soon,  others i can also 
recommend : - iaxphone by ipsando - firefly by 
virbiage. Time Bandit wrote: can 
someone tell me about a good iax softphone ??   
 Shameless plug : http://www.marccharbonneau.com/asterisk/mediaxphone.php 
  works only on windows  
 for one that works on Windows and Linux : http://iaxclient.sourceforge.net/iaxcomm/index.html 
  there is also DIAX : http://www.laser.com/dante/diax/diax.html and 
idefisk : http://www.asteriskguru.com/tools/idefisk_beta.php 
  hth 
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-- Tom VileBaldwin 
  Technology Solutions, IncConsulting - Web Design - VoIP Telephonywww.baldwintechsolutions.com Phone: 518-631-2855 
  x205Phone: 845-652-4578 x205Phone: 978-203-3848 
  x205Fax: 518-631-2856 
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- Web Design - VoIP Telephonywww.baldwintechsolutions.comPhone: 
518-631-2855 x205Phone: 845-652-4578 x205Phone: 978-203-3848 
x205Fax: 518-631-2856 
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Re: [Asterisk-Users] iax softphone



I'm running it on sp2 myself, never had a crash with it so far.


Jason Walker wrote:


Are you running on XP SP2just curious? How about the version of *?


*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of *Tom Vile
*Sent:* Saturday, October 22, 2005 10:03 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [Asterisk-Users] iax softphone

Nope,  I do not have that issue.

On 10/23/05, *Jason Walker* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:

Tom - do you end up with that phone shutting down with an error on
Windows XP? I downloaded the latest. After about 3 minutes on a
call, the other end can no longer hear me and then the phone just
dies.


*From:* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]] *On Behalf Of
*Tom Vile
*Sent:* Saturday, October 22, 2005 8:21 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [Asterisk-Users] iax softphone

Idefisk for me.  I love how it does not clutter the screen and it
works.

On 10/22/05, *Matt Florell* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:

We use the Firefly ThirdParty softphone on our windows
laptops. It's
free, easy to configure and will do IAX2 and SIP:
http://www.virbiage.com/firefly/download/firefly-thirdparty.exe
http://www.virbiage.com/firefly/download/firefly-thirdparty.exe

MATT---



On 10/22/05, Zoa [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:


Idefisk is currently only for windows, but a native linux

version is

nearly ready and will be released soon,
others i can also recommend
:
- iaxphone by ipsando
- firefly by virbiage.

Time Bandit wrote:

can someone tell me about a good iax softphone ??


Shameless plug :

http://www.marccharbonneau.com/asterisk/mediaxphone.php
http://www.marccharbonneau.com/asterisk/mediaxphone.php


works only on windows


for one that works on Windows and Linux :
http://iaxclient.sourceforge.net/iaxcomm/index.html

http://iaxclient.sourceforge.net/iaxcomm/index.html


there is also DIAX : http://www.laser.com/dante/diax/diax.html
and idefisk :

http://www.asteriskguru.com/tools/idefisk_beta.php
http://www.asteriskguru.com/tools/idefisk_beta.php


hth
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Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com http://www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Phone: 845-652-4578 x205
Phone: 978-203-3848 x205
Fax: 518-631-2856

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Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com http://www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Phone: 845-652-4578 x205
Phone: 978-203-3848 x205
Fax: 518-631-2856

Re: [Asterisk-Users] emacs syntax/keyowrd highlighting: asterisk-mode

2005-10-22 Thread JP Carballo


Dave Grey wrote:



Speaking of glaringly unworkable, like a numb-skull I edited and  
tested the thing with default my green-on-black color scheme.  I  
happened to open something up in a raw black-on-white xterm and  
realized that I had created a nightmare.  I have made the appropriate  
changes, so if you looked at it and said,  Hey, this is  
horrible..., then my apologies and give it another look.  Same url  
and filename above, updated version noted in the comments.


lyd


Is it? :)
I work in a myriad of colored terms. green-on-black is the default for 
my local machines, other schemes are used identify remote machines...
Not a show stopper for me. I have a few screen sessions open for ages. 
Sometimes, the only time I see a prompt is when I do M-x shell :)

I'll check out your changes the next time I need to edit a .conf file.

--
JP Carballo

http://www.netfone2x.com
Bringing the world closer.

It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. 


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RE: [Asterisk-Users] iax softphone