[Asterisk-Users] LiveVoip status report

[David Josephson]

There has been improvement in the quality of LiveVoip connections. Still 
some packet loss and resultant choppy audio, a little worse than with 
Vonage or Broadvoice. As noted in several posts over the past months, 
they still don't handle indication of ringing on an IAX channel if the 
caller has dialed a number in the Asterisk switch (for instance with the 
DISA app). The workaround previously suggested, to Answer() and then run 
Ringing() doesn't work in this case, because it still sends the IAX 
command for ringing which LiveVoip doesn't recognize. However, 
Playtones(ring) does work and represents a usable workaround for the 
price. They claim to be working on a new session controller that will 
fix this and other problems. We'll see.
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Re: [Asterisk-Users] What do I need to get started?

[Wilson Pickett]

> My message to the list was definitely flame bait for my
> ignorance, thanks for just giving me links and a point in the right
> direction. 

In an ideal world, where insecure people wouldn't need constant
validation on mailing lists to bolster their self worth, what just
happened here would be more common:

If someone feels like answering a question with useful info, they do.
Otherwise, they move on.

Unfortunately, the world is not ideal, so in this case the gods must
be smiling on you.

Welcome to the asterisk community :)
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[Asterisk-Users] zap to sip caller id "forwarding"

[Tom Makulski]

Hello,
I have 8 zap channels connected to my local teleco company, Id like to
get callerid working on my sip phones but asterisk always sets the
callerid as "asterisk" eventho I have callerid=asrecieved in the
zapata.conf and in my dialplan I set the callerid before dialing the
sip chan. Could anyone give me any pointers?
  

-- 
Best regards,
 Tom  mailto:[EMAIL PROTECTED]

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RE: [Asterisk-Users] Re: Email to Fax

[Anton Krall]

How are you doing it Justin? 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Justin Newman
Sent: Jueves, 21 de Abril de 2005 10:53 p.m.
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: Email to Fax

> Message: 11
> Date: Thu, 21 Apr 2005 20:39:22 -0500
> From: "Anton Krall" <[EMAIL PROTECTED]>
> Subject: [Asterisk-Users] Email to Fax
> 
> Anybody doing email to fax using spandsp?
> 

Yep...

Justin Newman
Newman Telecom, Inc.
[EMAIL PROTECTED]

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Re: [Asterisk-Users] 503 Error

[doug]

After speaking with out provider, they believe it has something to do with
the silence suppression tag in the SIP headers Asterisk is sending.  Is
there a way to remove the silence suppression tag completely?

Thanks,
Doug

> When trying to send calls from our Asterisk PBX to our upstream
> termination provider, I am getting
>
> Got SIP response 503 "Service Unavailable" back from PROVIDER
>
> We are sending the calls without registration, there is no username and
> password.  When we were using SER it would send them without a problem.
>
> Can someone tell me if the problem is on my side or the provider's?
>
> What would cause this problem?
>
> Thanks
> Doug
>
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[Asterisk-Users] Provider offering IAX and T.38 origination+termination?

[Adam Megacz]

Are there any IAX providers out there that also offer T.38 real-time
fax-over-IP?  Failing that, any SIP+T.38 providers?

  - a
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RE: [Asterisk-Users] Asterisk Restart after crash

[David Phelan]

After a crash of what??
Linux...asterisk??

Depends on how you have it setup

If you start asterisk with safe_asterisk, then if asterisk crashes it will
start again.
If you run safe_asterisk from say...your rc.local then it will start when
linux restarts.

Dave
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chuck Smith
Sent: Friday, 22 April 2005 1:56 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Asterisk Restart after crash


Does Asterisk restart itself if it crashes? If not is there a way to make
linux do it?



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--
No virus found in this incoming 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] Digium Card Issues

[Sahil Gupta]

Hi,
I'm trying to configure a digium card here.  Got everything working 
sweetly apart from the last bit..

dmesg shows:
TE110P: Span configured for ESF/B8ZS
Calling startup (flags is 4099)
Registered tone zone 1 (Australia)
whilst /etc/zaptel.conf has:
span = 1,1,1,ccs,hdb3,crc4
bchan = 1-10
dchan = 16
defaultzone = au
loadzone = au
Any ideas?
Regards,
Sahil Gupta
VoiceValley
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Re: [Asterisk-Users] BYOD provider other than broadvoice

[Luki]

Mark,

well, your ping results look pretty damn good. I get good quality with
up to 20 ms jitter and 80 ms pings. If on a very bad day jitter on my
line is 30+ ms, calls to PSTN sound "under the water" but calls to
other SIP devices through Broadvoice sound fine. I guess the jitter
buffer/handling is better in SIP adapters than what Broadvoice uses to
terminate to PSTN.

Anyway, I don't see why you get bad quality (it's not that I don't
believe you) so can't help you more. You're probably sick and tired
and ready to try someone else, which I understand if you've been
dealing with this for over a year. If not, you could try changing the
RTP packet size in asterisk (rtp.c) to either 10, 30 or 40 ms and see
if this makes a difference...

--Luki
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Re: [Asterisk-Users] Playing mp3's while recording voicemail

[trixter http://www.0xdecafbad.com]

On Fri, 2005-04-22 at 00:18 +0100, Rafal Kaniewski wrote:
> Anyone got a good idea on how to do this?
> 
> (its for a singing down the phone thing)
> 
> thanks
> 
Off the top of my head I would have to say monitor mp3player and record
would be your friends.


-- 
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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[Asterisk-Users] Asterisk Restart after crash

[Chuck Smith]

Does Asterisk restart itself if it crashes? If not is there a way to make
linux do it?



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Re: [Asterisk-Users] Playing mp3's while recording voicemail

[Rod Bacon]

I need to do something similar and I was thinking about starting a 
conference call using an AGI script and MeetMe.

Basically, one leg of the call is MP3, one is a recorder and the other is a 
live call.

I have no idea how to implement it in code, but it sounds logical to me.

- Original Message - 
From: "Rafal Kaniewski" <[EMAIL PROTECTED]>
To: 
Sent: Friday, April 22, 2005 9:18 AM
Subject: [Asterisk-Users] Playing mp3's while recording voicemail


Anyone got a good idea on how to do this?
(its for a singing down the phone thing)
thanks

Rafal Kaniewski
Rafal#movingimagearts.com

--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.308 / Virus Database: 266.10.1 - Release Date: 20/04/2005
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[Asterisk-Users] Re: Email to Fax

[Justin Newman]

> Message: 11
> Date: Thu, 21 Apr 2005 20:39:22 -0500
> From: "Anton Krall" <[EMAIL PROTECTED]>
> Subject: [Asterisk-Users] Email to Fax
> 
> Anybody doing email to fax using spandsp?
> 

Yep...

Justin Newman
Newman Telecom, Inc.
[EMAIL PROTECTED]

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Re: [Asterisk-Users] asterisk@home 0.9 zap problems

[Henry Devito]

-- Executing Dial("SIP/3001-e13a", "ZAP/1/65869804") in new stack
This is what's wrong I think. The line is missing the 'g' for the trunk 
group.  On all of my [EMAIL PROTECTED] boxes the cli shows

  -- Executing Dial("SIP/227-a4dd", "ZAP/g0/3428463") in new stack 

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[Asterisk-Users] TE110p - universal voltage?

[Craig Guy]

Can anyone with a TE110p confirm that it will fit and work in both a 3.3 and
5 volt pci slot?  From photos it looks to be a universal card but the digium
literature makes no mention of voltage requirements.

Craig

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Re: [Asterisk-Users] Bug?

[Ronald Wiplinger]

Kevin Bockman wrote:
From all what I can see here is that there might be a bug in the php 
program.
Have you already checked the line 47 and 92 in the php program if you 
have missed the semicolon at the end?
If that is not the case, can you dump the kernel to the list along with 
your bank account info, so that we have more fun in guessing and still 
get paid for.
   

I found another way to accomplish what I wanted without recording the
file in the AGI.  It would still be cool to debug it though.
Like I said, it works fine if I call locally with SIP.  It only crashes
if I call over the internet using IAX.
Could you tell me how you arrived at the line number of 47 and 92?  I
use a few AGIs but it seems to be the vm.agi that it has a problem
with.   It is a  67 line file though. :-)
 

Good that you take it with humor, ...
I wanted to say that it is completely useless to send the log files if 
we have no idea what your php is doing, ...
As you stated it did not complete the script, so it is obviously (for 
me) that the reason is in the php and not in Asterisk, or at least you 
can only find the reason if you know the part of the php where it stops.
Since you wrote the php, maybe you can make some print statments (to a 
file), like "I am at line xxx, my values are x"
At least this could give you a clue where it may stuck.
It still can be just a syntax error, which you only come to in certain 
situations, 

bye
Ronald
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RE: [Asterisk-Users] Cisco 7920 - chan_sccp - asterisk@home .9

[Chuck Smith]

Sccp.conf


[general]
keepalive = 5
context = default
dateFormat = D-M-Y
bindaddr = IP_ADDRESS * Server  
port = 2000 


[SEP000d288e] 
type = 7920
autologin = LoginID 
description = Wireless 
context = from-internal

[LoginID] 
id = 2005
Label = Wireless 
description = Wireless 
context = from-internal
callwaiting = 1 
mailbox =  
callerid = "Wireless", <2005> 



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul A Brown
Sent: Thursday, April 21, 2005 7:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Cisco 7920 - chan_sccp - [EMAIL PROTECTED] .9

Can I see your config files to help me sort mine?

Thanks
- Original Message - 
From: "Chuck Smith" <[EMAIL PROTECTED]>
To: "'Andy Hamilton'" <[EMAIL PROTECTED]>; "'Asterisk Users Mailing 
List - Non-Commercial Discussion'" 
Sent: Thursday, April 21, 2005 3:32 PM
Subject: RE: [Asterisk-Users] Cisco 7920 - chan_sccp - [EMAIL PROTECTED] .9


>I got it to work. I didn't set the source ip address of my Asterisk server
> in the sccp.conf file. They work like a charm now. Thanks for the offer
> though.
>
> Chuck
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Andy 
> Hamilton
> Sent: Wednesday, April 20, 2005 5:48 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Cisco 7920 - chan_sccp - [EMAIL PROTECTED] .9
>
> Chuck:
>
> I have been able to use a 7920 with Asterisk. Never used [EMAIL PROTECTED]
>
> If you post your config files (sccp.conf, SEPX.cnf, etc), I can
> have a look at them for any suggestions.
>
> -Andy
>
> On 4/19/05, Chuck Smith <[EMAIL PROTECTED]> wrote:
>> Has anyone been able to get chan_sccp to work with a 7920 on
> [EMAIL PROTECTED]
>>
>> I was able to compile chan_sccp and I see it as a running module in
> Asterisk
>> but I don't see any debugs on the console that shows the phone even 
>> making
>> an attempt. I followed the examples to the T but still no luck.
>>
>> Thanks
>>
>> -Original Message-
>> From: [EMAIL PROTECTED]
>> [mailto:[EMAIL PROTECTED] On Behalf Of Matt 
>> Darnell
>> Sent: Tuesday, April 19, 2005 10:39 PM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [Asterisk-Users] US$200 bounty for * paging feature
>>
>> On 4/19/05, Mike <[EMAIL PROTECTED]> wrote:
>> > >> . close source and we own the code.
>> > You are no better then Microsoft.
>>
>> Speaking of an over reaction
>>
>> -Matt
>> ___
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RE: [Asterisk-Users] Bug?

[Kevin Bockman]

>  From all what I can see here is that there might be a bug in the php 
> program.
> Have you already checked the line 47 and 92 in the php program if you 
> have missed the semicolon at the end?
> If that is not the case, can you dump the kernel to the list along with 
> your bank account info, so that we have more fun in guessing and still 
> get paid for.

I found another way to accomplish what I wanted without recording the
file in the AGI.  It would still be cool to debug it though.

Like I said, it works fine if I call locally with SIP.  It only crashes
if I call over the internet using IAX.

Could you tell me how you arrived at the line number of 47 and 92?  I
use a few AGIs but it seems to be the vm.agi that it has a problem
with.   It is a  67 line file though. :-)

Thanks,

Kevin Bockman

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Re: [Asterisk-Users] Asterisk Netgear FSM7326P and Cisco 7960 on VLAN

[Allen Niven]

we had problem with 7960 and Netgear but do not recall the version of 
Netgear.it was solved by upgrading the Netgear to latest rev...

SCollins wrote:
I had and issue with the FSM732P and DHCP Requests. I would get  
intermitant connectivity and eventually I would loose connectivity on a  
port, move the IP phone to a new port and eventually loose 
connectivity.  The Netgear sounds suspect.

If all ports are a member of Default VLAN 1, untagged, and the ports 
are  set to Admit All, the Switch will assign all packets received 
(untagged  and those with priotization set) to PVID 1. Could it be a 
VLAN  configuration or Routing problem?

Maybe it's a known bug that has been resolved with Firmware?
" FSM7326P Software Version 3.0.3.2
Issues This Firmware Resolves
1. The VLAN priority setting as configured were not working.
2. When setting fixed speed at 1000Mbps full-duplex for copper ports, 
the  switch issued a warning.
3. The switch only allowed 8 route entries instead of 16 route entries"

Hope something I said helps.
Sean



 On Tue, 19 Apr 2005 21:16:07 +0100, Ron Wellsted <[EMAIL PROTECTED]>  
wrote:

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
I have just finished a couple of very frustrating days trying to resolve
an issue with Asterisk 1.0.5, Netgear FSM7326P and Cisco 7960 phones
under SIP using a VLAN for the VoIP traffic.
The switch was configured with a default VLAN of 1 and the 7960s were
configured for a VLAN of 2.  The 7960s were tested with SIP firmware 7.3
and 7.4
The symptoms were as follows:
1/ When the phones were idle, every few minutes, the asterisk server
would loose MAC/IP address mapping for a phone.  Approx 10 seconds later
the mapping would reappear. (checked with "arp -avn")
2/ When in a call, we would randomly loose sound TO the phone.  Sound
FROM the phone was OK.  This loss would continue for about 10 seconds.
Using ethereal, we sniffed the traffic in and out of the Asterisk box
and saw that we were not getting replies to the targeted ARP packets
being sent from the asterisk box, only replies to the broadcast ARP  
packets.

Sniffing a phone revealed that although the ARP packets were correctly
tagged as VLAN 2, the phone was replying to the targeted ARP packets on
VLAN 1 (!?).  As a result the Asterisk box was (correctly) expiring the
arp cache entry when the ttl was reached, then it would send a broadcast
 ARP packet which would be responded to by the phone on VLAN 2.
I have worked around this be removing the VLANs and by using DHCP
reservations to put the phones on a different subnet from the rest of
the network.
Has anyone else found themselves in a similar situation?  If so, how did
you work around it?
Is it worth raising a TAC with Cisco?
- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
FWD:519961  Gossiptel:9309811
N 52.567623, W 2.137621
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Version: GnuPG v1.4.1 (GNU/Linux)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
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=JmsL
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--
Allen Niven
GlobalFone
350 Fifth Avenue #6206
Empire State Building
New York, NY 10118
+1-212-678-4381 office
+1-646-246-7415 cell
http://www.GlobalFone.biz
Instant Messaging Accounts
ICQ 137763656
Yahoo Messenger [EMAIL PROTECTED]
MSN Messenger [EMAIL PROTECTED]
PLEASE NOTE  I NEVER NEVER NEVER RECEIVE EMAILS ON HOTMAIL OR YAHOO
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Re: [Asterisk-Users] Bug?

[Ronald Wiplinger]

Kevin Bockman wrote:
Hello, I have found a possible bug in Asterisk.  The reason I say this
is that it does not coredump when I call locally over SIP, only when it
goes over the net via IAX.
I'm running an AGI (in PHP) which does not apparantly complete.  It
shows in the console that it runs, but never completes.  The agi runs
RECORD FILE.
I also tried running -STABLE (from yesterday) along with a few other
versions of -HEAD and it also crashes.  I assume that the "output
fwrites" are from PHP.
Sorry if I have done something wrong with the reporting, this is my
first time reporting a bug.  I think there is a procedure about posting
to the bugs database, but I want to make sure this is a bug and not
something I'm doing wrong.
Here's the console output at the time of the crash:
asterisk in free(): error: chunk is already free
output: fwrite: Broken pipe
Abort (core dumped)
output: fwrite: Broken pipe
output: fwrite: Broken pipe
 

From all what I can see here is that there might be a bug in the php 
program.
Have you already checked the line 47 and 92 in the php program if you 
have missed the semicolon at the end?
If that is not the case, can you dump the kernel to the list along with 
your bank account info, so that we have more fun in guessing and still 
get paid for.

(Sorry, I could not resist - PLEASE DO NOT send the kernel to the list!!!)
And a gdb bt full:
(gdb) bt full
#0  0x282adf17 in pthread_testcancel () from /usr/lib/libpthread.so.1
No symbol table info available.
#1  0x2829f0b5 in sigaction () from /usr/lib/libpthread.so.1
No symbol table info available.
#2  0x282991e1 in pthread_kill () from /usr/lib/libpthread.so.1
No symbol table info available.
#3  0x28298bb0 in raise () from /usr/lib/libpthread.so.1
No symbol table info available.
#4  0x28369627 in abort () from /lib/libc.so.5
No symbol table info available.
#5  0x2830d389 in ldexp () from /lib/libc.so.5
No symbol table info available.
#6  0x2830d3cd in ldexp () from /lib/libc.so.5
No symbol table info available.
#7  0x2830e2c1 in ldexp () from /lib/libc.so.5
No symbol table info available.
#8  0x2830e513 in ldexp () from /lib/libc.so.5
No symbol table info available.
#9  0x2830e644 in free () from /lib/libc.so.5
No symbol table info available.
#10 0x283c8bd6 in handle_recordfile (chan=0x283cfce0, agi=0x1,
argc=142206976, argv=0x283caf39) at res_agi.c:908
   fs = (struct ast_filestream *) 0x879e800
   f = (struct ast_frame *) 0x2836fae6
   tv = {tv_sec = 3000, tv_usec = 142205952}
   start = {tv_sec = 0, tv_usec = 3030}
   sample_offset = 1114132012
   res = 64
   ms = 730313
   sildet = (struct ast_dsp *) 0xbf307428
   dspsilence = 1
   silence = -1087343568
   gotsilence = 0
   silencestr = 0x2836fae6 "chunk is already free\n"
   rfmt = -1087343580
   __PRETTY_FUNCTION__ = "handle_recordfile"
#11 0x283caf39 in agi_exec_full (chan=0x8733c00, data=0x1, enhanced=0,
dead=1) at res_agi.c:1711
   uc = (struct localuser *) 0x17e
   ul = (struct localuser *) 0x283cfce0
   u = (struct localuser *) 0x876d430
   argv = {0xbf308048 "dial/vm.agi", 0xbf308058 "54390", 0x0
}
   buf = "dial/vm.agi\00054390", '\0' 
   tmp = 0x0
   argc = 675085536
   fds = {52, 56}
   efd = -1
   pid = 19445
   stringp = 0x17e 
   agi = {fd = 56, audio = -1, ctrl = 52}
   __PRETTY_FUNCTION__ = "agi_exec_full"
#12 0x283cb410 in deadagi_exec (chan=0x17e, data=0x6) at res_agi.c:1970
No locals.
#13 0x0807fdb2 in pbx_extension_helper (c=0x8733c00, con=0x0,
context=0xbf30ee88 "", exten=0x8733d3c "dial-dial",
   priority=0, label=0x8080bf8 "\203Ä \205À\211Å\017\205÷þÿÿ\213\203À",
callerid=0xbf30cc28 "dial/vm.agi|54390",
   action=0) at pbx.c:530
   e = (struct ast_exten *) 0x0
   sw = (struct ast_switch *) 0x81633a0
   data = 0x1 
   foundcontext = 0x0
   newstack = 141770288
   res = 0
   status = 675066876
   incstack = {0x0, 0x0, 0x0, 0x0, 0x81083c0 "\020", 0x40 
0x40 out of bounds>,
 0x55 , 0x2829e3b2 "[\201Ãê5\001",
0x282b199c "Äø\001",
 0x8107000 "@P\021(\025¡\233ÐÐ\001\020\b", 0xbf30ecd0 "", 0x2829ec8f
"\203Ä\020é\031\003", 0x28115050 "",
 0x2 , 0x0, 0x2829ea22 "[\201Ãz/\001",
0x2838d818 "\200U\021\b", 0x8761c50 "",
 0xbf30eca0 "Ðì0¿_6*(", 0x282a4d4a "[\201ÃRÌ",
 0x80a13bc
"\203ì\f¡\030W\017\b\205Àu\030\203ì\bh¼\023\n\bÿt$\034è;\"ûÿ\203Ä\034Ã\215v",
0x0,
 0x2 , 0x0, 0x2838d818 "\200U\021\b",
0x87af200 "@¸s(\025¡\233Ð",
 0xbf30ecc0 "Á\236*(*Z*(\234\031+(", 0x282a4089 "ÿ\217¬", 0x8115580
"P°\021\bP°\021\b\002", 0x87af248 "@\023v\b",
 0xf , 0x282a3b6a "[\201Ã2Þ", 0x2838d818
"\200U\021\b", 0x87af200 "@¸s(\025¡\233Ð",
 0xbf30ecd0 "", 0x282a365f
"\203Ä\020\215eô[^_ÉÃ\211öU\211åWVS\203ì\fè", 0x87af200
"@¸s(\025¡\233Ð", 0x0, 0x0,
 0x282aecde "[\201þ,", 0x28379458 "", 0x81000f0 "H\030\020\b",
0x282a9ec1 "[\201ÃÛz", 0x282a5a2a "[\201Ãr¿",
 0x282b199c "Äø\001", 0x8101000 "\200ó\017\b", 0xbf30ed00
"0í0¿ò\221)(",
 0x282ae635 "\203Ä\020\215eô[^_

RE: [Asterisk-Users] Bug?

[Kevin Bockman]

Darn, I forgot to say that I'm running:
Asterisk CVS-HEAD-04/21/05-16:53:20 
FreeBSD 5.3

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[Asterisk-Users] 100 & AAH .9

[Daniel Dziubanski]

Hi try adding the following line to zapata.conf at the bottom:- 
 
#include zapata-channels.conf 

http://sourceforge.net/forum/forum.php?thread_id=1266472&forum_id=420324
  
asterisk-users@lists.digium.com
asterisk-users@lists.digium.com

When ever I try to dial a pstn number, I get this message.  I did a yum update 
and also recompiled the zaptel and asterisk modules in /usr/src.  I used the 
same machine as I was using it for [EMAIL PROTECTED] 0.6.  

 

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[Asterisk-Users] Bug?

[Kevin Bockman]

Hello, I have found a possible bug in Asterisk.  The reason I say this
is that it does not coredump when I call locally over SIP, only when it
goes over the net via IAX.

I'm running an AGI (in PHP) which does not apparantly complete.  It
shows in the console that it runs, but never completes.  The agi runs
RECORD FILE.

I also tried running -STABLE (from yesterday) along with a few other
versions of -HEAD and it also crashes.  I assume that the "output
fwrites" are from PHP.

Sorry if I have done something wrong with the reporting, this is my
first time reporting a bug.  I think there is a procedure about posting
to the bugs database, but I want to make sure this is a bug and not
something I'm doing wrong.

Here's the console output at the time of the crash:
asterisk in free(): error: chunk is already free
output: fwrite: Broken pipe
Abort (core dumped)
output: fwrite: Broken pipe
output: fwrite: Broken pipe

And a gdb bt full:
(gdb) bt full
#0  0x282adf17 in pthread_testcancel () from /usr/lib/libpthread.so.1
No symbol table info available.
#1  0x2829f0b5 in sigaction () from /usr/lib/libpthread.so.1
No symbol table info available.
#2  0x282991e1 in pthread_kill () from /usr/lib/libpthread.so.1
No symbol table info available.
#3  0x28298bb0 in raise () from /usr/lib/libpthread.so.1
No symbol table info available.
#4  0x28369627 in abort () from /lib/libc.so.5
No symbol table info available.
#5  0x2830d389 in ldexp () from /lib/libc.so.5
No symbol table info available.
#6  0x2830d3cd in ldexp () from /lib/libc.so.5
No symbol table info available.
#7  0x2830e2c1 in ldexp () from /lib/libc.so.5
No symbol table info available.
#8  0x2830e513 in ldexp () from /lib/libc.so.5
No symbol table info available.
#9  0x2830e644 in free () from /lib/libc.so.5
No symbol table info available.
#10 0x283c8bd6 in handle_recordfile (chan=0x283cfce0, agi=0x1,
argc=142206976, argv=0x283caf39) at res_agi.c:908
fs = (struct ast_filestream *) 0x879e800
f = (struct ast_frame *) 0x2836fae6
tv = {tv_sec = 3000, tv_usec = 142205952}
start = {tv_sec = 0, tv_usec = 3030}
sample_offset = 1114132012
res = 64
ms = 730313
sildet = (struct ast_dsp *) 0xbf307428
dspsilence = 1
silence = -1087343568
gotsilence = 0
silencestr = 0x2836fae6 "chunk is already free\n"
rfmt = -1087343580
__PRETTY_FUNCTION__ = "handle_recordfile"
#11 0x283caf39 in agi_exec_full (chan=0x8733c00, data=0x1, enhanced=0,
dead=1) at res_agi.c:1711
uc = (struct localuser *) 0x17e
ul = (struct localuser *) 0x283cfce0
u = (struct localuser *) 0x876d430
argv = {0xbf308048 "dial/vm.agi", 0xbf308058 "54390", 0x0
}
buf = "dial/vm.agi\00054390", '\0' 
tmp = 0x0
argc = 675085536
fds = {52, 56}
efd = -1
pid = 19445
stringp = 0x17e 
agi = {fd = 56, audio = -1, ctrl = 52}
__PRETTY_FUNCTION__ = "agi_exec_full"
#12 0x283cb410 in deadagi_exec (chan=0x17e, data=0x6) at res_agi.c:1970
No locals.
#13 0x0807fdb2 in pbx_extension_helper (c=0x8733c00, con=0x0,
context=0xbf30ee88 "", exten=0x8733d3c "dial-dial",
priority=0, label=0x8080bf8 "\203Ä \205À\211Å\017\205÷þÿÿ\213\203À",
callerid=0xbf30cc28 "dial/vm.agi|54390",
action=0) at pbx.c:530
e = (struct ast_exten *) 0x0
sw = (struct ast_switch *) 0x81633a0
data = 0x1 
foundcontext = 0x0
newstack = 141770288
res = 0
status = 675066876
incstack = {0x0, 0x0, 0x0, 0x0, 0x81083c0 "\020", 0x40 ,
  0x55 , 0x2829e3b2 "[\201Ãê5\001",
0x282b199c "Äø\001",
  0x8107000 "@P\021(\025¡\233ÐÐ\001\020\b", 0xbf30ecd0 "", 0x2829ec8f
"\203Ä\020é\031\003", 0x28115050 "",
  0x2 , 0x0, 0x2829ea22 "[\201Ãz/\001",
0x2838d818 "\200U\021\b", 0x8761c50 "",
  0xbf30eca0 "Ðì0¿_6*(", 0x282a4d4a "[\201ÃRÌ",
  0x80a13bc
"\203ì\f¡\030W\017\b\205Àu\030\203ì\bh¼\023\n\bÿt$\034è;\"ûÿ\203Ä\034Ã\215v",
0x0,
  0x2 , 0x0, 0x2838d818 "\200U\021\b",
0x87af200 "@¸s(\025¡\233Ð",
  0xbf30ecc0 "Á\236*(*Z*(\234\031+(", 0x282a4089 "ÿ\217¬", 0x8115580
"P°\021\bP°\021\b\002", 0x87af248 "@\023v\b",
  0xf , 0x282a3b6a "[\201Ã2Þ", 0x2838d818
"\200U\021\b", 0x87af200 "@¸s(\025¡\233Ð",
  0xbf30ecd0 "", 0x282a365f
"\203Ä\020\215eô[^_ÉÃ\211öU\211åWVS\203ì\fè", 0x87af200
"@¸s(\025¡\233Ð", 0x0, 0x0,
  0x282aecde "[\201þ,", 0x28379458 "", 0x81000f0 "H\030\020\b",
0x282a9ec1 "[\201ÃÛz", 0x282a5a2a "[\201Ãr¿",
  0x282b199c "Äø\001", 0x8101000 "\200ó\017\b", 0xbf30ed00
"0í0¿ò\221)(",
  0x282ae635 "\203Ä\020\215eô[^_ÉÃU\211åS\203ì\024è", 0x2873b850
"\003P", 0x8101034 "\020\001\020\b", 0x0,
  0x282ae531 "\211Eðe\2135@", 0x2838d818 "\200U\021\b", 0x283795ec
"(5\f", 0x2873b850 "\003P", 0x282b199c "Äø\001", 0x0,
  0x10 , 0xbf30ed30 "\001", 0x282991f2
"\203Ä\020\215v", 0x87af200 "@¸s(\025¡\233Ð",
  0x8107000 "@P\021(\025¡\233ÐÐ\001\020\b", 0x0, 0x2829917a
"[\201Ã\"\210\001",
  0x80f21a5 "- IAX2/asterlink-13 is ringing\n", 0x80

[Asterisk-Users] Email to Fax

[Anton Krall]

Anybody doing email to fax using spandsp?

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[Asterisk-Users] Looking for an IAX(2) or SIP DID provider for LA, Orlando and Chicago areas.

[Carmi Weinzweig]

I am once again looking for someone that can provide a block of DIDs in 
Chicagoland, LA and Orlando with a pricing model that aligns with ISDN 
PRI service. I would like to pay a monthly fee per simultaneous call 
(equivalent to a trunk) with blocks of DID numbers priced at around 
$0.10 or so (what I would pay for those DIDs as a retail customer in 
SBC territory).

Thanks in advance.
/carmi
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[Asterisk-Users] asterisk@home 0.9 zap problems

[Dinesh]

Hi all,

 

Trying to upgrade to the new version, and all went well.  I
am having some problems with the zaptel config.  

 

When ever I try to dial a pstn number, I get this message. 
I did a yum update and also recompiled the zaptel and asterisk modules in
/usr/src.  I used the same machine as I was using it for [EMAIL PROTECTED]
0.6.  

 

genzaptelconf  
autoconfig Zaptel cards

 

I used as per what the [EMAIL PROTECTED] manual told me.  I
see all the modules for zaptel loaded

 

[EMAIL PROTECTED] root]# dmesg | grep Zap

Zapata Telephony Interface Registered on major 196

 

[EMAIL PROTECTED] root]# lsmod | grep zap

zaptel   
178560   4  [ztdummy wcfxs]

 

but still getting this annonying error saying all circuits
are busy.  I also get this error when I do a zaptel restart.

 

[EMAIL PROTECTED] zaptel]# /etc/init.d/zaptel restart

Unloading zaptel hardware drivers: wcfxs ztdummy 

Removing zaptel module:  zaptel: Device or resource
busy

[FAILED]

Loading zaptel framework:  [  OK  ]

Waiting for zap to come online ...OK

Loading zaptel hardware modules: wcfxs ztdummy 

Running ztcfg:  [  OK  ]

 

Is this normal?

 

I have tested the phone socket on the port 1 its working
fine.  I also tried to play around with the trunks g1 g0..etc, doesn’t
work.

 

My Zapata.conf

 

;

; Zapata telephony interface

;

; Configuration file

 

[trunkgroups]

 

[channels]

 

language=en

context=from-pstn

signalling=fxs_ks

rxwink=300 
; Atlas seems to use long (250ms) winks

;

; Whether or not to do distinctive ring detection on FXO
lines

;

;usedistinctiveringdetection=yes

 

usecallerid=yes

hidecallerid=no

callwaiting=yes

usecallingpres=yes

callwaitingcallerid=yes

threewaycalling=yes

transfer=yes

cancallforward=yes

callreturn=yes

echocancel=yes

echocancelwhenbridged=no

echotraining=800

rxgain=0.0

txgain=0.0

group=1

callgroup=1

pickupgroup=1

immediate=no

 

;faxdetect=both

faxdetect=incoming

;faxdetect=outgoing

;faxdetect=no

 

[EMAIL PROTECTED] root]# asterisk -r

Asterisk 1.0.7, Copyright (C) 1999-2004 Digium.

Written by Mark Spencer <[EMAIL PROTECTED]>

=

Connected to Asterisk 1.0.7 currently running on owl (pid =
1707)

Verbosity is at least 3

    -- Executing
Macro("SIP/3001-e13a", "dialout-trunk|1|65869804") in new
stack

    -- Executing
GotoIf("SIP/3001-e13a", "1?4") in new stack

    -- Goto (macro-dialout-trunk,s,4)

    -- Executing
GotoIf("SIP/3001-e13a", "0?6") in new stack

    -- Executing
SetCallerID("SIP/3001-e13a", "64789451") in new stack

    -- Executing
SetGroup("SIP/3001-e13a", "OUT_1") in new stack

    -- Executing
CheckGroup("SIP/3001-e13a", "1") in new stack

    -- Executing
SetVar("SIP/3001-e13a", "DIAL_NUMBER=65869804") in new
stack

    -- Executing
SetVar("SIP/3001-e13a", "DIAL_TRUNK=1") in new stack

    -- Executing
AGI("SIP/3001-e13a", "fixlocalprefix") in new stack

    -- Launched AGI Script
/var/lib/asterisk/agi-bin/fixlocalprefix

    -- AGI Script fixlocalprefix completed,
returning 0

    -- Executing
Dial("SIP/3001-e13a", "ZAP/1/65869804") in new stack

  == Everyone is busy/congested at this time

    -- Executing
NoOp("SIP/3001-e13a", "dial failed") in new stack

    -- Executing
Macro("SIP/3001-e13a", "outisbusy") in new stack

    -- Executing
Playback("SIP/3001-e13a", "allison7/all-circuits-busy-now")
in new stack

    -- Playing
'allison7/all-circuits-busy-now' (language 'en')

    -- Executing
Playback("SIP/3001-e13a", "allison7/pls-try-call-later") in
new stack

    -- Playing 'allison7/pls-try-call-later'
(language 'en')

    -- Executing
Macro("SIP/3001-e13a", "hangupcall") in new stack

    -- Executing
ResetCDR("SIP/3001-e13a", "w") in new stack

    -- Executing
NoCDR("SIP/3001-e13a", "") in new stack

    -- Executing
Wait("SIP/3001-e13a", "5") in new stack

  == Spawn extension (macro-hangupcall, s, 3) exited
non-zero on 'SIP/3001-e13a' in macro 'hangupcall'

  == Spawn extension (macro-outisbusy, s, 3) exited
non-zero on 'SIP/3001-e13a' in macro 'outisbusy'

  == Spawn extension (from-internal, 65869804, 2)
exited non-zero on 'SIP/3001-e13a'

    -- Executing
Macro("SIP/3001-e13a", "hangupcall") in new stack

    -- Executing
ResetCDR("SIP/3001-e13a", "w") in new stack

    -- Executing
NoCDR("SIP/3001-e13a", "") in new stack

    -- Executing
Wait("SIP/3001-e13a", "5") in new stack

  == Spawn extension (macro-hangupcall, s, 3) exited
non-zero on 'SIP/3001-e13a' in macro 'hangupcall'

  == Spawn extension (from-internal, h, 1) exited
non-zero on 'SIP/3001-e13a'

 

[EMAIL PROTECTED] zaptel]# genzaptelconf -s

 

 

STOPPING ASTERISK

 

Disconnected from Asterisk server

Asterisk Stopped

 

STOPPING FOP SERVER

FOP Server Stopped

 

SETTING FILE PERMISSIONS

Permissions OK

 

STARTING ASTERISK

Asterisk Started

 

STARTING FOP SERVER

 

** SIP/3001 in position 2

** IAX2/3004 in position 3

FOP Server St

[Asterisk-Users] Music Onhold Problem

[kritikus Araklidas]

Hi every one:
I have * RT  latest CVS version, the problem is when i make calls from soft 
SIP phone to other soft phone (Between extensions) if the receiver put on 
hold i cannot hear the music in the other end, if the caller put the call on 
hold, happen the same thing.

I'm using SJphone version 2.48 and the error messages is the following:
Apr 21 21:13:17 WARNING[10144]: res_musiconhold.c:865 local_ast_moh_start: 
No class: default

Regards.
Kritikus.
_
Don’t just search. Find. Check out the new MSN Search! 
http://search.msn.click-url.com/go/onm00200636ave/direct/01/

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[Asterisk-Users] GS ATA 286 goes deaf.

[Chuck Keeter]

Hi all,
I hope that someone can help me with this problem, I am running a GS ATA 
286 that connects to mt Asteriis box at work, up till a few days ago, it 
was working fine. Now it seems to dial out, but I get no audio from the 
phone. If I call my cell phone from it, the cell will ring but I cannot 
talk or hear on the phone connected to the GS ATA 286.

I just updated the firmware and built a 2nd Asterisk to test with.. Same 
problem.. anyone had this happen or any suggestions?

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RE: [Asterisk-Users] FAX

[Anton Krall]

Ronald.

I got the asterisk extension working... (asterisk doesn't auto decte
incoming faxes yet though) 

But the problem I have is that the faxes are coming in as blank pages with
one black thick line at the top... Why is this?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ronald
Wiplinger
Sent: Jueves, 21 de Abril de 2005 07:48 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] FAX

Anton Krall wrote:

>Guys.. Anybody sucessfully using spandsp on cvs head?
>  
>
A few days I asked the same question and got a working answer from Jesse
Guardiani Please look for a message with the subject: Fax questions from
April 16,
2005
He also referes to the web site: 
http://scottstuff.net/scott/archives/000152.html

Just one of my experience. I copied the extension line from the web, which
is in TWO lines combined with a "\" at the end of the first line.
That does not work in Asterisk,  it must be on one line. That was the
only really difficulty I had.

Two things I want to extend:
1. to get the caller name somehow. At the moment I get only the
Caller-ID- number. Maybe it is possible from the fax header or via own
database.
2. to allow EACH extension to get a fax and use the voicemail.conf email
address as the receiver of the fax

bye

Ronald

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Re: [Asterisk-Users] FAX

[Ronald Wiplinger]

Anton Krall wrote:
Guys.. Anybody sucessfully using spandsp on cvs head?
 

A few days I asked the same question and got a working answer from Jesse 
Guardiani
Please look for a message with the subject: Fax questions from April 16, 
2005
He also referes to the web site: 
http://scottstuff.net/scott/archives/000152.html

Just one of my experience. I copied the extension line from the web, 
which is in TWO lines combined with a "\" at the end of the first line.
That does not work in Asterisk,  it must be on one line. That was 
the only really difficulty I had.

Two things I want to extend:
1. to get the caller name somehow. At the moment I get only the 
Caller-ID- number. Maybe it is possible from the fax header or via own 
database.
2. to allow EACH extension to get a fax and use the voicemail.conf email 
address as the receiver of the fax

bye
Ronald
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[Asterisk-Users] FAX

[Anton Krall]

Guys.. Anybody sucessfully using spandsp on cvs head?

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[Asterisk-Users] dynamic port problem !!

[CM Rahman Jr.]

Is there a way to fix this problem? I am using cisco 5300 to connect to
asterisk but it is failing to recognize by IP due to port number changing.
Unfortunately, Cisco send sip request with changing port number. What can be
done? No matter what I put on sip.conf, I couldn't get it to match. Any
help?

Here is the debug

Found no matching peer or user for '216.236.160.15:50896'


Thanks

&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*
C.M. Rahman Jr.
IT Manager
CCNP, MCSE Security"Secure your self by securing your System"
CompTI Security Plus Certified
CCS Internet
http://www.ccsi.com
13706 Research Blvd. Suite 100
Austin, TX 78750
Tel: 512-257-2274 Ex: 115




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[Asterisk-Users] IAX peer

[MDM]

To All
I am a new to Asterisk and dialplans have me stumped
I just inherited 2 Asterisk server conected as IAX peers.
Now from what i can tell when Asterisk Server (ask-chic) needs to make a 
call to an extension
which resides on the other server (ask-MAIN) it goes over a IAX channel.

Now i am trying to add that third asterisk server to mix (ask-SD) and i 
figured i would do it in baby steps.
The first thing i did was configure two local SIP client so they could 
call each other and leave voicemail.

Then i tried to  add (ask-SD) to (ask-Main) as a IAX peer. Just like 
(ask-chic).
To test i tried dialing an (ask-SD) ext from a phone off the (ask-MAIN) 
server.

The call did not go through. However i watched it from the CLI> and 
captured the following output. What could be wrong. i am so stumped.

'[EMAIL PROTECTED]' in 15000 ms
-- Accepting call from '' to '7101' on channel 0/23, span 2
-- Executing Dial("Zap/47-1", "IAX2/ask-SD/7101") in new stack
-- Called ask-SD/7101
Apr 21 13:52:12 WARNING[147465]: chan_iax2.c:5495 socket_read: Call 
rejected by ask-SD: No such context/extension
-- IAX2/ask-SD/2 is circuit-busy
-- Hungup 'IAX2/ask-SD/2'
  == Everyone is busy/congested at this time
-- Executing Congestion("Zap/47-1", "") in new stac

here is server (ask-sd) iax.conf and extension.conf files
iax.conf:
[EMAIL PROTECTED] asterisk]# cat iax.conf
[general]
allow=all
jitterbuffer=no
tos=lowdelay
[guest]
type=user
context=guest
callerid="Gust  User"
; BMS-ask-Main-asterisk - Incoming -
;
[ask-mail]
type=user
secret=ask-mail
context=from-ask-main
disallow=all
allow=ulaw
; bmc-asl-main - Outgoing
;
[telx-nyc]
type=peer
username=ask-sd   ; our username
secret=ask-sd; our password
host=192.168.11.30  ; host to connect to
;qualify=yes
;trunk=yes  ; use trunking
[EMAIL PROTECTED] asterisk]#
Extension.conf
[EMAIL PROTECTED] asterisk]#  cat extensions.conf
[general]
static=yes
writeprotect=yes
[bogon-calls]
exten => _.,1,Congestion
[from-sip]
exten => 7101,1,Dial(SIP/7101,20)
exten => 7101,2,Voicemail(u7101)
exten => 7101,102,Voicemail(b7101)
exten => 7101,103,Hangup
exten => 7102,1,Dial(SIP/7102,20)
exten => 7102,2,Voicemail(u7102)
exten => 7102,102,Voicemail(b7102)
exten => 7102,103,Hangup
exten => 7199,1,VoicemailMain(${CALLERIDNUM})
[macro-telx-nyc]
exten => s,1,Noop()
exten => s,2,Dial(IAX2/ask-mail/${ARG1})
[outgoing]
;ingnorepat => 9
exten => _9NXXNXX,1,Noop()
exten => _9NXXNXX,2,Macro(ask-main,${EXTEN})
exten => _9NXXNXX,3,Playback(invalid)
exten => _9NXXNXX.4,Hangup
[EMAIL PROTECTED] asterisk]#
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[Asterisk-Users] Demo phones with advertisement announcements

[Ronald Wiplinger]

I am asked very often to let the user try for a while. And of course 
they want to have it for free. However, there is nothing like free lunch 
out there.

I got the idea to "bother" these people, by playing an advertisement 
before they actually make the call and even after a certain time.
Has anybody done that before?

Ideas I got for that is:
1. put the caller into a conference call with the advertisment channel
2. let the caller listen to the first advertisement before inviting the 
other party to the conference
3. keep playing ads, till the called party is in the conference too.
4. immediately silent the ads, when called party pickes up
5. wait the desired time and start to play the next advertisement block.

Advanced feature:
1. give the caller the chance to pay for the call by key in a  
password(?), that means:
a. it kicks out the advertisement
or
b. forward the call () so that the call is now directly connected.

`?' means I am not sure if that is a good idea nor if that is possible
bye
Ronald
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Re: [Asterisk-Users] Problems with app_dbodbc.c

[Matt]

Ok
And what exactly is asterisk realtime?  I'm familiar with it being
that it reads out of a database... but is it a new version of
asterisk?  Something you can add to an existing 1.0.7 install?  Just
some changes you make in config file?

On 4/21/05, Matthew Boehm <[EMAIL PROTECTED]> wrote:
> Matt wrote:
> > Hi,
> > Has anyone experience with app_dbodbc.c ?  If so.. any thoughts what
> > the issue might be trying to compile this?
> 
> The author of dbodbc just said on the -dev list that this app is
> deprecated and you shouldn't use it anymore. Use RealTime instead.
> 
> -Matthew
> 
>
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RE: [Asterisk-Users] Queues configuration

[Anton Krall]

Hi Henry

How do you do it? I mean, if a caller is already on the queue and suddenly
all agents logoff.. How do you make the caller fall out of the queue and
into an IVR where he can leave a message?

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Henry Devito
Sent: Jueves, 21 de Abril de 2005 01:56 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Queues configuration


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

Sent: Thursday, April 21, 2005 9:31 AM
Subject: [Asterisk-Users] Queues configuration


> 1) If I understand correctly, an agent can belong to more than one queue. 
> Does that mean that as soon as the agent logs in, that agent can receive 
> calls from either of the queues s/he belongs to?
Agents/members can login to different queue or be defined statically


> 2) When the agent is making outbound calls and I wish to identify the 
> calls as belonging to one campaign or another, is there a way to identify 
> that?
That is just a function of the dial plan  should now be an issue.

> 3) When callers call into the * box and the agents are busy, they will be 
> put on the queue. Now, I wish to be able to tell the callers they are in 
> position X in the queue. However, if after, say, 30 seconds no agent is 
> available yet, I wish to give the caller the opportunity to listen to a 
> IVR where s/he can make a decision whether to continue holding, be 
> transfered to another agent group, or leave a message. Is this possible? 
> Does anyone have any sample of how to do this?

This is possible.  The person can jump out of queue at anytime or you can 
set an 'overflow' to an IVR menu.



> Thanks,
> Daniel
>
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Re: [Asterisk-Users] Cisco 7920 - chan_sccp - asterisk@home .9

[Mojo with Horan & Company, LLC]

Paul, here's what I'm doing.  I'm using the easter2005-testing 
chan_sccp, and I have the following sccp.conf:
So far, everything's Just Worked :)  I have no idea how to select codecs 
though.  The 7920 reports to chan_sccp that it is capable of ulaw and 
alaw, g729a and g729b, and g729a+b, and I've purchased a g729a license.  
mebbe I need to purchase two licenses instead of one?  mebbe I need to 
unlock the 729 capability in the phone??
Mojo
-
[general]
keepalive = 5
context = internal
dateFormat = M-D-Y  ; M-D-Y in any order (5 chars max)
bindaddr = 0.0.0.0; replace 1.2.3.4 with the ip address of the 
asterisk box.
port = 2000; listen on port 2000 (Skinny, default)

[SEP001121BC4E97]  
description = Moj's Wireless
type  = 7920
context   = internal
tzoffset  = 0
autologin = mojowireless

[mojowireless]
id  = 1001
context = internal
callwaiting = 1
mailbox = 22
callerid= "Mojo Wireless", <22>
-
Paul A Brown wrote:
Can I see your config files to help me sort mine?
Thanks
- Original Message - From: "Chuck Smith" <[EMAIL PROTECTED]>
To: "'Andy Hamilton'" <[EMAIL PROTECTED]>; "'Asterisk Users 
Mailing List - Non-Commercial Discussion'" 

Sent: Thursday, April 21, 2005 3:32 PM
Subject: RE: [Asterisk-Users] Cisco 7920 - chan_sccp - [EMAIL PROTECTED] .9


I got it to work. I didn't set the source ip address of my Asterisk 
server
in the sccp.conf file. They work like a charm now. Thanks for the offer
though.

Chuck

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andy 
Hamilton
Sent: Wednesday, April 20, 2005 5:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Cisco 7920 - chan_sccp - [EMAIL PROTECTED] .9

Chuck:
I have been able to use a 7920 with Asterisk. Never used [EMAIL PROTECTED]
If you post your config files (sccp.conf, SEPX.cnf, etc), I can
have a look at them for any suggestions.
-Andy
On 4/19/05, Chuck Smith <[EMAIL PROTECTED]> wrote:
Has anyone been able to get chan_sccp to work with a 7920 on
[EMAIL PROTECTED]
I was able to compile chan_sccp and I see it as a running module in
Asterisk
but I don't see any debugs on the console that shows the phone even 
making
an attempt. I followed the examples to the T but still no luck.

Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt 
Darnell
Sent: Tuesday, April 19, 2005 10:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] US$200 bounty for * paging feature

On 4/19/05, Mike <[EMAIL PROTECTED]> wrote:
> >> . close source and we own the code.
> You are no better then Microsoft.
Speaking of an over reaction
-Matt
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[Asterisk-Users] Playing mp3's while recording voicemail

[Rafal Kaniewski]

Anyone got a good idea on how to do this?

(its for a singing down the phone thing)

thanks


Rafal Kaniewski
Rafal#movingimagearts.com



-- 
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.308 / Virus Database: 266.10.1 - Release Date: 20/04/2005
 

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Re: [Asterisk-Users] Recording Queue agents

[kyle Hagan]

We use the AgentCallBackLogin option, that will work for you too.
Kyle
David John Walsh wrote:
Hi all
We have a psudo-switchboard in the offices, with receptionists who
login into the switchboard queue, all is well.
I have been asked to look into the possibilty of recording all the
agents conversations.
Legal issues aside:
Can anyone give me any pointers as to how this might be achived.  The
basic rules are:
We only want the conversation, otherwise i'd just monitor all thru the
wait period :)
when the agent transfers it off, the recording needs to stop.
-
Am I opening a can of worms here, or is it as simple as setting a flag
in one of the commands?
All the agents phones are SIP, the queue is an internal extention (so
that all can use the switchboard) so just monitoring the trunks is out
as well (sadly)
Any help (even if its just ideas or sample code) would be greatly apreciated.
We run [EMAIL PROTECTED] - to keep things simple (0.6 in production) (0.9
on my test machine)
thanks
David
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Re: [Asterisk-Users] Cisco 7920 - chan_sccp - asterisk@home .9

[Paul A Brown]

Can I see your config files to help me sort mine?
Thanks
- Original Message - 
From: "Chuck Smith" <[EMAIL PROTECTED]>
To: "'Andy Hamilton'" <[EMAIL PROTECTED]>; "'Asterisk Users Mailing 
List - Non-Commercial Discussion'" 
Sent: Thursday, April 21, 2005 3:32 PM
Subject: RE: [Asterisk-Users] Cisco 7920 - chan_sccp - [EMAIL PROTECTED] .9


I got it to work. I didn't set the source ip address of my Asterisk server
in the sccp.conf file. They work like a charm now. Thanks for the offer
though.
Chuck

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andy 
Hamilton
Sent: Wednesday, April 20, 2005 5:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Cisco 7920 - chan_sccp - [EMAIL PROTECTED] .9

Chuck:
I have been able to use a 7920 with Asterisk. Never used [EMAIL PROTECTED]
If you post your config files (sccp.conf, SEPX.cnf, etc), I can
have a look at them for any suggestions.
-Andy
On 4/19/05, Chuck Smith <[EMAIL PROTECTED]> wrote:
Has anyone been able to get chan_sccp to work with a 7920 on
[EMAIL PROTECTED]
I was able to compile chan_sccp and I see it as a running module in
Asterisk
but I don't see any debugs on the console that shows the phone even 
making
an attempt. I followed the examples to the T but still no luck.

Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt 
Darnell
Sent: Tuesday, April 19, 2005 10:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] US$200 bounty for * paging feature

On 4/19/05, Mike <[EMAIL PROTECTED]> wrote:
> >> . close source and we own the code.
> You are no better then Microsoft.
Speaking of an over reaction
-Matt
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RE: [Asterisk-Users] using * for Internet call waiting

[jltaylor]

I'm an ISP, what I would like is a client for the dialup customer to run.
They would use call fwd busy to my did on an asterisk box.
I'd signal and they could click on button (URL) to download .wav file in
asterisk voice mail.
James

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Mojo with
Horan & Company, LLC
Sent: Thursday, April 21, 2005 5:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] using * for Internet call waiting


I once tried the pagoo service.  Seems I had to ask the telco for Call
Forward Busy, and provide them with the toll free number pagoo gave me
for their service.  When the forwarded call is received by their
systems, they  would see _my_ callerid information, and thus know to
contact my computer for the notification purpose.

Also, not sure if this is on track with what you want, but I've used
jabber_client.pl tied into my dialplan to popup the callerid info of an
incoming call on my screen..  I could then choose to answer the call or
let it ring to voicemail.  Seems the jabber client Neos has
well-designed popups.

links:
http://jabberd.jabberstudio.org/2/
for the jabber_alert.pl script, allows sending jabber msgs from cmd line.

http://www.neosmt.com/
for a jabber client that pops up incoming messages. Note, this is also
an H.323 client.  Haven't tried it with * yet, but I have been meaning to.

Here's the specific Dialplan line I use:
[inpstn]
exten => s,2,TrySystem(echo "Incoming call from :${CALLERID}" |
jabber_alert.pl -e [EMAIL PROTECTED] -n [EMAIL PROTECTED] -w
senders_password)

Because it can sometimes take 2 or 3 seconds to send the jabber message
on my network, I use TrySystem instead of System, which blocks, waiting
for the return code from the command I passed.  Because the return code
is prolly irrelevant, you'd most likely want to use TrySystem too...

hope this helps :)
Moj


>-Original Message-
>From: [EMAIL PROTECTED]
>[mailto:[EMAIL PROTECTED] On Behalf Of Gary Carr
>Sent: Thursday, April 21, 2005 4:50 PM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: [Asterisk-Users] using * for Internet call waiting
>
>Wondering if it is possible or if something already exist to setup * to
>offer Internet Call Waiting. For those that do not know what it is, it's
>a small application that runs on a users computer that will pop up a
>window letting them know they have a incoming call and who it is from
>then they can choose to take the call which will disconnect their dialup
>modem and ring their phone or send the call to voice mail.
>
>
>Thanks,
>
>
>Gary
>
>
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>
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>
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Re: [Asterisk-Users] One touch voicemail on Cisco 7940/60

[William Suffill]

I prefer to use a numerical exten. but same result.
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RE: [Asterisk-Users] wcte11xp digium card

[Nathaniel Angelo A. Torres (247talk)]

I'll typed make for libunicall then this error come up.
Any reason and solution for this error:

ranlib .libs/libunicall.a
creating libunicall.la
(cd .libs && rm -f libunicall.la && ln -s ../libunicall.la libunicall.la)
if gcc -DHAVE_CONFIG_H -I. -I. -I. -DPROTOCOLDIR="\"/usr/lib/unicall\""
-g -O2 -MT testcall.o -MD -MP -MF ".deps/testcall.Tpo" -c -o testcall.o
testcall.c; \
then mv -f ".deps/testcall.Tpo" ".deps/testcall.Po"; else rm -f
".deps/testcall.Tpo"; exit 1; fi
/bin/sh ./libtool --mode=link gcc  -g -O2   -o testcall  testcall.o
-lunicall -lpthread -laudiofile -ldl
gcc -g -O2 -o .libs/testcall testcall.o
/usr/src/libunicall-0.0.2/.libs/libunicall.so -lpthread -laudiofile -ldl
testcall.o(.text+0x85): In function `channel_read_file':
/usr/src/libunicall-0.0.2/testcall.c:124: undefined reference to `dtmf_rx'
testcall.o(.text+0xa4):/usr/src/libunicall-0.0.2/testcall.c:125: undefined
reference to `dtmf_get'
testcall.o(.text+0xb03): In function `handle_uc_event':
/usr/src/libunicall-0.0.2/testcall.c:420: undefined reference to
`dtmf_rx_init'
collect2: ld returned 1 exit status
make[1]: *** [testcall] Error 1
make[1]: Leaving directory `/usr/src/libunicall-0.0.2'
make: *** [all] Error 2

Thanks.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Titux
Sent: Thursday, April 21, 2005 9:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] wcte11xp digium card

Hi, do you have libtiff installed?
also libtiff-dev is needed for RH clones...


On 4/18/05, Nathaniel Angelo A. Torres (247talk) <[EMAIL PROTECTED]> wrote:
> Any idea what's the cause of this:
> 
> configure: WARNING: spandsp.h: present but cannot be compiled
> configure: WARNING: spandsp.h: check for missing prerequisite headers?
> configure: WARNING: spandsp.h: see the Autoconf documentation
> configure: WARNING: spandsp.h: section "Present But Cannot Be
Compiled"
> configure: WARNING: spandsp.h: proceeding with the preprocessor's result
> configure: WARNING: spandsp.h: in the future, the compiler will take
> precedence
> configure: WARNING: ## -- ##
> configure: WARNING: ## Report this to the AC_PACKAGE_NAME lists.  ##
> configure: WARNING: ## -- ##
> 
> Cheers,
> Angelo
> 
> -Original Message-
> From: Nathaniel Angelo A. Torres (247talk) [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, April 19, 2005 8:38 AM
> To: 'Nathaniel Angelo A. Torres (247talk)'; [EMAIL PROTECTED];
'Asterisk
> Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] wcte11xp digium card
> 
> Hi, please help me solve this problem in compiling libsupertone,
libunicall
> and libmfcr2 to accommodate E1-R2 using wcte11xp.
> 
> Thanks.
> 
> Cheers,
> Angelo
> 
> -Original Message-
> From: Nathaniel Angelo A. Torres (247talk) [mailto:[EMAIL PROTECTED]
> Sent: Monday, April 18, 2005 10:51 PM
> To: '[EMAIL PROTECTED]'; 'Asterisk Users Mailing List - Non-Commercial
> Discussion'
> Subject: RE: [Asterisk-Users] wcte11xp digium card
> 
> Hi Matteo,
> 
> Please find attached excerpts of the error below:
> 
> supertone.c:337: invalid type argument of `->'
> supertone.c:337: syntax error before "xmlChar"
> supertone.c: At top level:
> supertone.c:344: redefinition of `cur'
> supertone.c:263: `cur' previously defined here
> supertone.c:344: invalid type argument of `->'
> supertone.c:344: warning: data definition has no type or storage class
> supertone.c:345: syntax error before "while"
> supertone.c:357: syntax error before '&&' token
> supertone.c:357: warning: data definition has no type or storage class
> supertone.c:357: syntax error before '}' token
> supertone.c:357: conflicting declarations of `__result'
> supertone.c:357: `__result' previously declared here
> supertone.c:357: warning: `__result' was declared `extern' and later
> `static'
> supertone.c:357: `x' undeclared here (not in a function)
> supertone.c:357: `__s2' undeclared here (not in a function)
> supertone.c:357: syntax error before "if"
> supertone.c:357: conflicting declarations of `__result'
> supertone.c:357: `__result' previously defined here
> supertone.c:357: warning: data definition has no type or storage class
> supertone.c:357: syntax error before '}' token
> supertone.c:357: warning: data definition has no type or storage class
> supertone.c:357: syntax error before '}' token
> supertone.c:357: conflicting declarations of `__result'
> supertone.c:357: `__result' previously declared here
> supertone.c:357: warning: `__result' was declared `extern' and later
> `static'
> supertone.c:357: `__s1' undeclared here (not in a function)
> supertone.c:357: `set_id' undeclared here (not in a function)
> supertone.c:357: syntax error before "if"
> supertone.c:357: conflicting declarations of `__result'
> supertone.c:357: `__result' previously defined here
> supertone.c:357: warning: data definition has no t

[Asterisk-Users] Recording Queue agents

[David John Walsh]

Hi all

We have a psudo-switchboard in the offices, with receptionists who
login into the switchboard queue, all is well.

I have been asked to look into the possibilty of recording all the
agents conversations.

Legal issues aside:

Can anyone give me any pointers as to how this might be achived.  The
basic rules are:

We only want the conversation, otherwise i'd just monitor all thru the
wait period :)

when the agent transfers it off, the recording needs to stop.

-

Am I opening a can of worms here, or is it as simple as setting a flag
in one of the commands?

All the agents phones are SIP, the queue is an internal extention (so
that all can use the switchboard) so just monitoring the trunks is out
as well (sadly)

Any help (even if its just ideas or sample code) would be greatly apreciated.

We run [EMAIL PROTECTED] - to keep things simple (0.6 in production) (0.9
on my test machine)

thanks

David
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[Asterisk-Users] does ast_app_getdata() reset timeout?

[Xu Wang]

We have an application to call ast_app_getdata() 3 times in order to get
inputs from callers. It seems like that ast_app_getdata() returns
immediately if callers take more time to input first two questions. Here is
the example:

res = ast_app_getdata(chan, "your-account", cd->destination, 7, 0);
..
res = ast_app_getdata(chan, "your-password", cd->destination, 7, 0);
..
res = ast_app_getdata(chan, "your-options", cd->destination, 7, 0);

If 'your-account' takes more time, then 'your-password' returns immediately.

It seems to us that ast_app_getdata() timeout is not reset when returning.
We have 1.0.7 asterisk.

Does anyone know the issue?

thanks!
steven

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Re: [Asterisk-Users] Problems with app_dbodbc.c

[Matt Roth]

Matt,
app_dbodbc.c may be deprecated, but what you described is a common 
problem.  As per this thread:

http://lists.digium.com/pipermail/asterisk-dev/2004-June/004809.html
if you are running make in one of the Asterisk source directory's 
subdirectories (such as /usr/src/asterisk/apps/) you need to edit the 
Makefile to define _GNU_SOURCE.  You can do this by locating the first 
instance of a line containing the string "CFLAGS+=" and adding the 
following string on the next line:

CFLAGS+=-D_GNU_SOURCE
Matthew Roth
http://www.voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Debian
Matthew Boehm wrote:
Matt wrote:
 

Hi,
Has anyone experience with app_dbodbc.c ?  If so.. any thoughts what
the issue might be trying to compile this?
   

   The author of dbodbc just said on the -dev list that this app is
deprecated and you shouldn't use it anymore. Use RealTime instead.
-Matthew
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RE: [Asterisk-Users] Error in starting asterisk

[Nathaniel Angelo A. Torres (247talk)]

Hi, I've tried commenting on echotraining but same error message appears.

[channels]
language=en

context=from-pstn
signalling=em_w
faxdetect=incoming
usecallerid=yes
echocancel=yes
echocancelwhenbridged=no
;echotraining=800
group=1
channel=1-15,17-30

Apr 22 06:02:04 VERBOSE[18915]:   == Parsing '/etc/asterisk/phone.conf': Apr
22 06:02:04 VERBOSE[18915]:   == Parsing '/etc/asterisk/phone.conf': Found
Apr 22 06:02:04 VERBOSE[18915]:   == Registered channel type 'Phone'
(Standard Linux Telephony API Driver)
Apr 22 06:02:04 VERBOSE[18915]:  [chan_zap.so]Apr 22 06:02:04
VERBOSE[18915]:  [chan_zap.so] => (Zapata Telephony)
Apr 22 06:02:04 VERBOSE[18915]:   == Parsing '/etc/asterisk/zapata.conf':
Apr 22 06:02:04 VERBOSE[18915]:   == Parsing '/etc/asterisk/zapata.conf':
Found
Apr 22 06:02:04 WARNING[18915]: parse error: No category context for line 10
of zapata.conf
Apr 22 06:02:04 ERROR[18915]: Unable to load config zapata.conf
Apr 22 06:02:04 WARNING[18915]: chan_zap.so: load_module failed, returning
-1
Apr 22 06:02:04 VERBOSE[18915]:   == Unregistered channel type 'Tor'
Apr 22 06:02:04 VERBOSE[18915]:   == Unregistered channel type 'Zap'
Apr 22 06:02:04 WARNING[18915]: Loading module chan_zap.so failed!

Thanks.

Angelo

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert Webb
Sent: Friday, April 22, 2005 1:43 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Error in starting asterisk


On Fri, 22 Apr 2005 01:26:45 +0800
  "Nathaniel Angelo A. Torres (247talk)" 
<[EMAIL PROTECTED]> wrote:
> Hi, here's the content of my Zapata.conf
> 
> [channels]
> language=en
> 
> context=default
> signalling=em_w
> faxdetect=incoming
> usecallerid=yes
> echocancel=yes
> echocancelwhenbridged=no
> echotraining=800
> group=0
> channel=1-15,17-30
> 
> I really don't know what to load into these values.  I 
>wanted to use E1R2 of
> wcte11xp
> 
> Please help me out. Thanks.
> 
> Angelo
> 
> 





> Nathaniel Angelo A. Torres (247talk) wrote:
> 
>> Hi, I'm receiving this error, please help me solve this.
>>
>> Apr 22 00:12:26 VERBOSE[3735]: == Parsing 
>>'/etc/asterisk/phone.conf': 
>> Apr 22 00:12:26 VERBOSE[3735]: == Parsing 
>>'/etc/asterisk/phone.conf': 
>> Found
>>
>> Apr 22 00:12:26 VERBOSE[3735]: == Registered channel 
>>type 'Phone' 
>> (Standard Linux Telephony API Driver)
>>
>> Apr 22 00:12:26 VERBOSE[3735]: [chan_zap.so]Apr 22 
>>00:12:26 
>> VERBOSE[3735]: [chan_zap.so] => (Zapata Telephony)
>>
>> Apr 22 00:12:26 VERBOSE[3735]: == Parsing 
>>'/etc/asterisk/zapata.conf': 
>> Apr 22 00:12:26 VERBOSE[3735]: == Parsing 
>>'/etc/asterisk/zapata.conf': 
>> Found
>>
>> Apr 22 00:12:26 WARNING[3735]: parse error: No category 
>>context for 
>> line 10 of zapata.conf
>>
>> Apr 22 00:12:26 ERROR[3735]: Unable to load config 
>>zapata.conf
>>
>> Apr 22 00:12:26 WARNING[3735]: chan_zap.so: load_module 
>>failed, 
>> returning -1
>>
>> Apr 22 00:12:26 VERBOSE[3735]: == Unregistered channel 
>>type 'Tor'
>>
>> Apr 22 00:12:26 VERBOSE[3735]: == Unregistered channel 
>>type 'Zap'
>>
>> Apr 22 00:12:26 WARNING[3735]: Loading module 
>>chan_zap.so failed!
>>
>> I'm trying to setup E1 R2 for the digium wcte11xp
>>
>> Thanks.
>>
>> Angelo
>>


Angelo, try commenting out or deleteling the echotraining 
cmmand in your config file. From what I can tell on the 
Wiki, that command is for the X100P card only. And per 
your error, if I am counting correctly, it states that 
there is no category context on line 10. I am assuming 
that line 10 in your config file is the echotraining 
command.

Open your zapata.conf and verisfy that line 10 is indeed 
the echotraining command. If not, please post back waht is 
at line 10.
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RE: [Asterisk-Users] using * for Internet call waiting

[jltaylor]

Would like to see a small "client" for this.
It could be SIP or IAX without all of the phone features.
It would need to provide a URL to the .wav file so it could be played.
Any ideas?
James

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Gary Carr
Sent: Thursday, April 21, 2005 3:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] using * for Internet call waiting


Wondering if it is possible or if something already exist to setup * to
offer Internet Call Waiting. For those that do not know what it is, it's a
small application that runs on a users computer that will pop up a window
letting them know they have a incoming call and who it is from then they can
choose to take the call which will disconnect their dialup modem and ring
their phone or send the call to voice mail.


Thanks,


Gary


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Re: [Asterisk-Users] using * for Internet call waiting

[Mojo with Horan & Company, LLC]

I once tried the pagoo service.  Seems I had to ask the telco for Call 
Forward Busy, and provide them with the toll free number pagoo gave me 
for their service.  When the forwarded call is received by their 
systems, they  would see _my_ callerid information, and thus know to 
contact my computer for the notification purpose.

Also, not sure if this is on track with what you want, but I've used 
jabber_client.pl tied into my dialplan to popup the callerid info of an 
incoming call on my screen..  I could then choose to answer the call or 
let it ring to voicemail.  Seems the jabber client Neos has 
well-designed popups.

links:
http://jabberd.jabberstudio.org/2/
for the jabber_alert.pl script, allows sending jabber msgs from cmd line.
http://www.neosmt.com/
for a jabber client that pops up incoming messages. Note, this is also 
an H.323 client.  Haven't tried it with * yet, but I have been meaning to.

Here's the specific Dialplan line I use:
[inpstn]
exten => s,2,TrySystem(echo "Incoming call from :${CALLERID}" | 
jabber_alert.pl -e [EMAIL PROTECTED] -n [EMAIL PROTECTED] -w 
senders_password)

Because it can sometimes take 2 or 3 seconds to send the jabber message 
on my network, I use TrySystem instead of System, which blocks, waiting 
for the return code from the command I passed.  Because the return code 
is prolly irrelevant, you'd most likely want to use TrySystem too...

hope this helps :)
Moj

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gary Carr
Sent: Thursday, April 21, 2005 4:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] using * for Internet call waiting
Wondering if it is possible or if something already exist to setup * to
offer Internet Call Waiting. For those that do not know what it is, it's
a small application that runs on a users computer that will pop up a
window letting them know they have a incoming call and who it is from
then they can choose to take the call which will disconnect their dialup
modem and ring their phone or send the call to voice mail.
Thanks,
Gary
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RE: [Asterisk-Users] BYOD provider other than broadvoice

[Wiley Siler]

Are you running Ulaw G711?

W
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Musone
Sent: Thursday, April 21, 2005 1:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] BYOD provider other than broadvoice

about 80% of my calls using broadvoice sound _horrible_

all of my customers, vendors, and acquaintences say almost the same
exact thing. "You sound like you're under water".

i've tried every broadvoice proxy, upgraded asterisk till i'm blue in
the face, remoave all firewalls, NATs, setup asterisk on it's own
completely dedicated server..you name it. still "You sound like you're
under water".

This happens whether i call out to people, or people call inbound. I can
hear people on the other end perfectly fine...they almost always hear
horrible jitter.
i'm on two DS3's with actual bandwidth usage of about 5M/sec (i.e. i'm
at like 6% capacity)

My ping times to broadvoice are excellent:

PING proxy.dca.broadvoice.com (147.135.0.128) 56(84) bytes of data.
64 bytes from 147.135.0.128: icmp_seq=1 ttl=51 time=23.1 ms
64 bytes from 147.135.0.128: icmp_seq=2 ttl=51 time=21.0 ms
64 bytes from 147.135.0.128: icmp_seq=3 ttl=51 time=20.7 ms
64 bytes from 147.135.0.128: icmp_seq=4 ttl=51 time=20.9 ms
64 bytes from 147.135.0.128: icmp_seq=5 ttl=51 time=21.1 ms
64 bytes from 147.135.0.128: icmp_seq=6 ttl=51 time=20.3 ms
64 bytes from 147.135.0.128: icmp_seq=7 ttl=51 time=20.9 ms


any SIP<->SIP call _not using broadvoice (i.e. calling from my sipura
device to anoher SIP device across the internet) is perfectly fine..

This has been going on for over a year now...had yet to be better. i'm
about to switch providers also because i simply can't take it anymore.
This is absolutely a broadvoice problem, and since they don't support
asterisk (not that I blame them) i dont have much options. i've been
thinking of just hooking up my Sipura directly to broadvoice to verify
the problem and then "force" them to fix it on the supported device.



-Mark



On 4/21/05, Adam Robins <[EMAIL PROTECTED]> wrote:
> I totally concur.  I switched from Broadvoice to VoicePulse because 
> users were complaining about call quality.  Now, the quality is good 
> -- when it doesn't drop altogether.
> 
> What could be worse than touting your new VoIP system to a client and 
> having it drop the call?
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Wiley 
> Siler
> Sent: Thursday, April 21, 2005 12:32 PM
> To: Trevor Harrison; Asterisk Users Mailing List - Non-Commercial 
> Discussion
> Subject: RE: [Asterisk-Users] BYOD provider other than broadvoice
> 
> Hmmm... Think I would prefer something harder to get provisioned but 
> that doesn't drop calls.
> 
> Your users must be forgiving as hell...  Mine would show up with 
> pitchforks and torches if calls dropped regularly.
> They get twitchy if the calls just vary too much in quality...  8)
> 
> Cheers,
> Wiley
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Trevor 
> Harrison
> Sent: Thursday, April 21, 2005 8:45 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] BYOD provider other than broadvoice
> 
> On 4/21/05, Adam Robins <[EMAIL PROTECTED]> wrote:
> > I drop every 3-4 call with VoicePulse Connect.
> 
> My users are also reporting occasional dropped calls when dialing via 
> VoicePulse Connect.
> 
> But I love the ease of use and setup with their service.
> 
> -Trevor
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this transmission in error, any use, reproduction or dissemination of
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[Asterisk-Users] Provisioning lines 5 and 6 via TFTP

[Robert Goodyear]

Has anyone experienced a problem provisioning lines 5 and 6 of a Cisco 
7960 via a SIPx.CNF over TFTP?

What I'm experiencing is that regardless of the linex_... entries in 
the CNF file, lines 5 and 6 show UNPROVISIONED on the phone console, 
despite the fact that the rest of the line provisioning fields are 
correctly filled on the phone.

As soon as I re-enter the line name on the keypad for line 5 -- and 
JUST the line name because, remember, the rest of the fields are 
already present -- then lines 5 and 6 appear on the phone console.

Weird!
/rg
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[Asterisk-Users] forwarding Sip call to IAX and vice-versa

[Daniel HAIDUC]

hello everyone. i am a new user with asterisk and just playing with it, 
trying to make different configurations.
i have managed to make a call from a sip to another sip phone and 
recently from iax to sip, but not from iax to iax and iax to sip
below are my extensions.conf and the error asterisk is reporting. i 
tryed to keep everything as simple as possible.
thank you for your responses

daniel
[SIP]
exten=> 1000,1,Dial(SIP/cipi,20)
exten=> 1000,2,Answer
exten=> 1000,3,Hangup
exten=> 1001,1,Dial(SIP/nicu,20)
exten=> 1001,2,Answer
exten=> 1001,3,Hangup
exten=> 1002,1,Dial(SIP/bibi,20)
exten=> 1002,2,Answer
exten=> 1002,3,Hangup
exten=> 2000,1,Dial(IAX/dani,20)
exten=> 2000,2,Answer
exten=> 2000,3,Hangup
[IAX]
exten=> _NXXX,1,Dial(IAX/dani,20)
exten=> _NXXX,2,Answer
exten=> _NXXX,3,Hangup
exten=> _,1,Dial(SIP/nicu,20)
exten=> _,2,Answer
exten=> _,3,Hangup
Executing Dial("SIP/nicu-3cbb", "IAX/dani|20") in new stack
Apr 21 20:30:42 WARNING[3497]: channel.c:1901 ast_request: No channel 
type registered for 'IAX'
Apr 21 20:30:42 NOTICE[3497]: app_dial.c:746 dial_exec: Unable to create 
channel of type 'IAX'
== Everyone is busy/congested at this time
-- Executing Answer("SIP/nicu-3cbb", "") in new stack
-- Executing Hangup("SIP/nicu-3cbb", "") in new stack
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Re: [Asterisk-Users] Large Asterisk Setup (~500 Concurrent Calls + Scalability)

[Matt Roth]

Daniel,
I would be interested to hear if anyone knows of a method to completely 
offload the Monitor command from the master server.  It is the missing 
piece of the puzzle to optimizing the digital recording process.

I'm assuming that the CPU usage you are referring to would be incurred 
due to converting the input and output packets from their codec to WAV 
format.  Please correct me if I'm wrong in that assumption.  I see the 
benefits of offloading that CPU usage to another machine, but I'm not 
sure if it's possible or how to implement it if it is. 

One benefit of the design that you overlooked is the offloading of disk 
I/O.  In this application that may not make a big difference, but it 
does allow us to do the mixing and compression to MP3 on a separate 
machine.  Both of those activities are CPU intensive.

You're correct that if we instructed the Monitor command to mix the 
files the mixing would occur on the master server.  I looked at the 
documentation and source (res_monitor.c) of the Monitor command to 
confirm that the default behavior is to NOT mix the files.  The options 
argument must contain the character 'm' for the mixing to occur.  We 
will be executing the Monitor command WITHOUT mixing, then running a 
periodic process on the Digital Recording Client to mix the files and 
compress the result to an MP3.

Thanks for your interest,
Matthew Roth
http://www.voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Debian
Daniel Salama wrote:
But it seems to me that the load of the Monitor command will remain in 
the master server. The only thing you are really offloading is the 
saving of the media stream which should have minimal impact. My 
concern is that, with a busy installation, issuing too many Monitor 
commands may "overload" CPU usage and have a direct effect on the 
quality of the conversations. Are my assumptions wrong?

When you tell the Monitor command to combine the resulting .wav files, 
that will also be executed in the master server, which should add to 
its load. Unless you tell the Monitor command NOT to combine the audio 
files and you run that as a separate "croned" processed in the client 
server which will also execute lame after soxmix.
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RE: [Asterisk-Users] using * for Internet call waiting

[Dan Levine]

I've seen this service done with AOL, I was curious how it was done on
standard phone lines.  Was it something the coordinated with the telco
in some sort of hunt group configuration or something of that nature?


-
Dan Levine
CYTEXONE | Your Technology Specialists
t: 877.CYTEXONE x 810
l: 212.477.0990 x 810
e: [EMAIL PROTECTED]
http://www.cytexone.com

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gary Carr
Sent: Thursday, April 21, 2005 4:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] using * for Internet call waiting

Wondering if it is possible or if something already exist to setup * to
offer Internet Call Waiting. For those that do not know what it is, it's
a small application that runs on a users computer that will pop up a
window letting them know they have a incoming call and who it is from
then they can choose to take the call which will disconnect their dialup
modem and ring their phone or send the call to voice mail.


Thanks,


Gary


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[Asterisk-Users] Queue member persistent stats

[Matthew Boehm]

I've got 5 agents who login/logff via AddQueueMember. Each time they do so,
their stats get reset. Is there anyway to keep these stats across logins?

-Matthew

-- 

Matthew Boehm, IT DirectorCypress Telecommunications
[EMAIL PROTECTED]   3838 N. Sam Houston Parkway E #400
T: 832-200-8640 x3044  Houston, TX 77032

My girlfriend was recently diagnosed with multiple personality disorder;
 When she called yesterday, my CallerID box exploded.


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Re: [Asterisk-Users] using * for Internet call waiting

[Steve Prior]

Gary Carr wrote:
Wondering if it is possible or if something already exist to setup * to 
offer Internet Call Waiting. For those that do not know what it is, it's 
a small application that runs on a users computer that will pop up a 
window letting them know they have a incoming call and who it is from 
then they can choose to take the call which will disconnect their dialup 
modem and ring their phone or send the call to voice mail.
That doesn't really make sense if the * box is in your house because
if the phone line is tied up for a dialup call, then the * box doesn't
have a phone line to receive the call either (unless you had call hunting
in which case you wouldn't need the feature in the first place).  This
sounds like the sort of feature that can only be offered on the
central office side which can know your line is tied up and then know
to email/alert you.
The other scenario is having an * box in a call center that is forwarding
calls to agents and notifies them by TCP/IP if when it tries their
extension and gets a busy signal.  This sounds possible, but I don't
think it's what you meant.
Steve
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[Asterisk-Users] CVS-HEAD: Sip not paying attention to context

[Matthew Boehm]

How can I debug this:

CLI>sip show peer 3047
 * Name   : 3047
  Secret   : 
  MD5Secret: 
  Context  : cytel-fax



CLI>asterisk*CLI> show channels
  Channel  (Context  ExtensionPri )   State  Appl.   Data
SIP/3047-4a2e  (global-outgoing  2819870065   1   )  Up  Dial
ZAP/R1d/2819870065|60


Umm...what?!? How can sip show peer tell me one context, but when the ata
actually makes a call, it goes out a different context?

[cytel-fax]
 exten => _[278]x,1,Dial(SIP/[EMAIL PROTECTED],30)
 exten => _1.,1,Dial(SIP/[EMAIL PROTECTED],30)

 exten => i,1,Playback(pbx-invalid)
 exten => i,2,Hangup()

 exten => t,1,Playback(pbx-invalid)
 exten => t,2,Hangup()

[global-outgoing]
 exten => _[287]X,1,Dial(ZAP/R1d/${EXTEN},60)
 exten => _[287]X,102,GotoIf($[${DIALSTATUS} = BUSY]?b,1:${EXTEN},2)
 exten => _[287]X,2,Dial(SIP/[EMAIL PROTECTED],60)
 exten => _[287]X,103,GotoIf($[${DIALSTATUS} = BUSY]?b,1)
 exten => _[287]X,3,Hangup()

I'm confused.

-- 

Matthew Boehm, IT DirectorCypress Telecommunications
[EMAIL PROTECTED]   3838 N. Sam Houston Parkway E #400
T: 832-200-8640 x3044  Houston, TX 77032

My girlfriend was recently diagnosed with multiple personality disorder;
 When she called yesterday, my CallerID box exploded.


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Re: [Asterisk-Users] Problems with app_dbodbc.c

[Matthew Boehm]

Matt wrote:
> Hi,
> Has anyone experience with app_dbodbc.c ?  If so.. any thoughts what
> the issue might be trying to compile this?

The author of dbodbc just said on the -dev list that this app is
deprecated and you shouldn't use it anymore. Use RealTime instead.

-Matthew

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

[Iassen Hristov]

I suppose you know about the upcoming DS3000P

Supposedly it can handle 28 T1's, so I assume the DSP is done onboard.

Maybe you should give Digium a call.

P.S. Sorry for not threading, but I get the digest and have not figured out
how to inject the original message-id header.

> From: [EMAIL PROTECTED] on behalf of Matt Roth
> Sent: Wed 4/20/2005 1:10 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Large Asterisk Setup (~500 Concurrent Calls
> +Scalability)
> 
> 
> 
> List Members,
> 
> I am involved in the process of designing a large Asterisk setup for a
> call center.  A graphical overview of our tentative design can be found
> here:
> 
> http://home.comcast.net/~mroth01/LargeAsteriskSetup.gif
> 
> Originally, we planned to implement this design by purchasing one
> multi-processor machine and putting multiple quad-span T1 cards
> (Wildcard TE4xxPs) into it.  Through research, it was determined that
> the PCI bus couldn't handle the digital signal processing (DSP) from
> more than one quad-span card.
> The goal of our new design is to offload the DSP to the Asterisk slave
> servers, then route the calls via IAX2 trunks to the Asterisk master
> server.  The Asterisk master server will provide us with a centralized
> point for queuing, digital recording, and music on hold, as well as
> configuration, monitoring, and reporting.  Configuration of the Asterisk
> slave servers would be limited to setting up extensions to terminate the
> incoming T1s and setting up IAX2 trunks to the Asterisk master server. 
> These configurations would be rare, so the slave servers would be
> configured manually on the boxes themselves.



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Re: [Asterisk-Users] One touch voicemail on Cisco 7940/60

[Ron Wellsted]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Morris, Simon wrote:
> Hello,
> 
> I'd like to program my Cisco phones to authenticate themselves to
> voicemail upon hitting the right button on my 7940/60's
> 
> Ideally the voicemail app will detect which extension the call is coming
> from and drop the user straight into the menu.
> 
> Is this possible?
> 
> Many thanks
> 
> 
> ~sm

Yes this is possible.

In your extensions.conf:

exten => _8501,1,Answer()
exten => _8501,2,VoicemailMain(s${CALLERIDNUM})
exten => _8501,3,Hangup()

then program the messages button to dial 8501 either via settings, SIP
Settings, Messages URI (set to 8501) or in the SIPDefault.cnf file

- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
FWD:519961  Gossiptel:9309811
N 52.567623, W 2.137621
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.1 (GNU/Linux)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org

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Re: [Asterisk-Users] Multiple Line config help

[kurt x]

That works.  What I am tyring to do is have two separate DIDs.  One is
4027 and the
other is 94207.  Line 1 = DID 4027 and Line 2 = DID 94027.   Dialing
4027 works to line
1 but dial 94027 gets a 486 busy.

Kurt 

On 4/21/05, Henry Devito <[EMAIL PROTECTED]> wrote:
> Don't you have to configure your dialplan to hunt to the next extensions?
> How else would * know to try 94207 if 4207 is busy?
> - Original Message -
> From: "kurt x" <[EMAIL PROTECTED]>
> To: "Asterisk" 
> Sent: Thursday, April 21, 2005 3:08 PM
> Subject: [Asterisk-Users] Multiple Line config help
> 
> I have aSIPURA 841 that is working on L1 with phone # 4027.  L2 is
> configured for 94027.
> Both numbers register with Asterisk.  When issuing the command "sip show
> peers"
> both numbers have the same IP address but 94027 show its sip port at
> 5061.  Which I expect is right.  When I dial 4027 it works but when I
> dial 94027 I get a "486 busy here" and voice mail picks up.
> 
> config below:
> 
> sip.conf
> [4027]
> type=friend
> host=dynamic
> dtmfmode=rfc2833
> context=home
> callerid=SIPURA-L1
> canreinvite=no
> [EMAIL PROTECTED]
> 
> [94027]
> type=friend
> host=dynamic
> dtmfmode=rfc2833
> context=home
> callerid=SIPURA-L2
> canreinvite=no
> [EMAIL PROTECTED]
> 
> extensions.conf
> exten => _40xx,1,Answer
> exten => _40xx,2,Dial(SIP/${EXTEN},10,t)
> exten => _40xx,3,Voicemail(u${EXTEN})
> exten => _40xx,4,Hangup
> exten => _40xx,103,Voicemail(b${EXTEN})
> exten => _40xx,104,Hangup
> 
> exten => _940xx,1,Answer
> exten => _940xx,2,Dial(SIP/${EXTEN},10,t)
> exten => _940xx,3,Voicemail(u${EXTEN})
> exten => _940xx,4,Hangup
> exten => _940xx,103,Voicemail(b${EXTEN:1})
> exten => _940xx,104,Hangup
> 
> Thanks,
> 
> Kurt
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Re: [Asterisk-Users] BYOD provider other than broadvoice

[Mark Musone]

Exactly :( Hence my yearlong love-hate affair with them...




On 4/21/05, Daniel Dziubanski <[EMAIL PROTECTED]> wrote:
> So the options are?
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Adam Robins
> Sent: Thursday, April 21, 2005 1:29 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion; Trevor Harrison
> Subject: RE: [Asterisk-Users] BYOD provider other than broadvoice
> 
> I totally concur.  I switched from Broadvoice to VoicePulse because
> users were complaining about call quality.  Now, the quality is good --
> when it doesn't drop altogether.
> 
> What could be worse than touting your new VoIP system to a client and
> having it drop the call?
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Wiley
> Siler
> Sent: Thursday, April 21, 2005 12:32 PM
> To: Trevor Harrison; Asterisk Users Mailing List - Non-Commercial
> Discussion
> Subject: RE: [Asterisk-Users] BYOD provider other than broadvoice
> 
> Hmmm... Think I would prefer something harder to get provisioned but
> that doesn't drop calls.
> 
> Your users must be forgiving as hell...  Mine would show up with
> pitchforks and torches if calls dropped regularly.
> They get twitchy if the calls just vary too much in quality...  8)
> 
> Cheers,
> Wiley
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Trevor
> Harrison
> Sent: Thursday, April 21, 2005 8:45 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] BYOD provider other than broadvoice
> 
> On 4/21/05, Adam Robins <[EMAIL PROTECTED]> wrote:
> > I drop every 3-4 call with VoicePulse Connect.
> 
> My users are also reporting occasional dropped calls when dialing via
> VoicePulse Connect.
> 
> But I love the ease of use and setup with their service.
> 
> -Trevor
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> in error, any use, reproduction or dissemination of this transmission is
> strictly prohibited. If you are not the intended recipient, please
> immediately notify the sender by reply email and delete this message and its
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Re: [Asterisk-Users] Call waiting

[[EMAIL PROTECTED]]

dial *70

check out the handbook for all the feature codes

http://asteriskathome.sourceforge.net/handbook


--- Sascha Ferley <[EMAIL PROTECTED]> wrote:
> Hi,
> 
> 
> I am trying to figure out how to setup call waiting
> on a [EMAIL PROTECTED]
> box. We get the call waiting signal from the telco
> and would like to be
> able to switch calls.
> Our setup right now is as following:
> 
> [PSTN] -> [EMAIL PROTECTED] -> [sip to Cisco ATA 188]
> -> Siemens 8825 (Analog)
> 
> When we had the siemens plugged into the PSTN
> directly, we could switch by
> just pressing the line button again. Now this
> doesn't seem to work going
> through the sip channel. Does anyone know how to
> enable this such that the
> switch line signal is propagate back the the PSTN
> and be able to switch
> the call to the call waiting one?
> 
> 
> Please let me know
> 
> Thanks
> Sascha
> 
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Re: [Asterisk-Users] What do I need to get started?

[jonr]

Wilson Pickett wrote:
I want to set up a * box that connects to my telephone line so I can
make calls using VOIP. Do I have to sign up for a service to do this

Yes, read the first doc below.

can I buy a card that goes into my * box that allows me to use VOIP over
my telephone line. If so, what card do I need? I am in the USA if that

Look at Digium.com the TDM400P is a card with 4 sockets for modules to
connect either phone lines or phones. This is the hardware interface
to nomal (home) phone lines and normal (analog) telephone equipment
such as corded or cordless phones.
Start by reading this tutorial:
http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html
The next one adds the POTS or PSTN phones
http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html
This is good to help set up asterisk and explain some conecpts
http://www.automated.it/guidetoasterisk.htm
Reading this would be a good idea:
http://www.asteriskdocs.org
Wilson,
Thanks for the docs and for pointing me to what I need to buy, the 
TDM400P. This is what has been confusing me, I didn't know what I was 
supposed to buy. My message to the list was definitely flame bait for my 
ignorance, thanks for just giving me links and a point in the right 
direction. I will hopefully have more intelligent questions when I post 
back.no guarantees. ;)

Jon
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Re: [Asterisk-Users] One touch voicemail on Cisco 7940/60

[Henry Devito]

Title: One touch voicemail on Cisco 7940/60



Yes it is possible.  Just setup and extension 
that logs into voicemailmain without asking for username or passcode.  Then 
set the messages configuration to that extension in the appropriate config file 
on the 7960.  

  - Original Message - 
  From: 
  Morris, 
  Simon 
  To: asterisk-users@lists.digium.com 
  
  Sent: Thursday, April 21, 2005 2:23 
  PM
  Subject: [Asterisk-Users] One touch 
  voicemail on Cisco 7940/60
  
  Hello,I'd like to program my Cisco phones to 
  authenticate themselves to voicemail upon hitting the right button on my 
  7940/60'sIdeally the voicemail app will detect which extension the 
  call is coming from and drop the user straight into the menu.Is this 
  possible?Many thanks~sm 
  
  

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Re: [Asterisk-Users] Queues configuration

[Henry Devito]

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

Sent: Thursday, April 21, 2005 2:00 PM
Subject: Re: [Asterisk-Users] Queues configuration


Would you happen to have some sample config on how to do this? Is this 
done in queues.conf or in the dial plan? Even pseudocode will help.

Thanks,
Daniel
On Apr 21, 2005, at 2:55 PM, Henry Devito wrote:
3) When callers call into the * box and the agents are busy, they will 
be put on the queue. Now, I wish to be able to tell the callers they are 
in position X in the queue. However, if after, say, 30 seconds no agent 
is available yet, I wish to give the caller the opportunity to listen to 
a IVR where s/he can make a decision whether to continue holding, be 
transfered to another agent group, or leave a message. Is this possible? 
Does anyone have any sample of how to do this?
This is possible.  The person can jump out of queue at anytime or you can 
set an 'overflow' to an IVR menu.

Timeout is set in the extensions.conf file with the queue command
Queue(queuename|options|optionalurl|announceoverride|timeout)
Queues an incoming call in a particular call queue as defined in 
queues.conf.

The option string may contain zero or more of the following characters:
 a.. 't' - allow the called user transfer the calling user
 b.. 'T' - to allow the calling user to transfer the call.
 c.. 'd' - data-quality (modem) call (minimum delay).
 d.. 'H' - allow caller to hang up by hitting *.
 e.. 'n' - no retries on the timeout; will exit this application and go to 
the next step.
 f.. 'r' - ring instead of playing MOH

In addition to transferring the call, a call may be parked and then picked 
up by another user. The optional URL will be sent to the called party if the 
channel supports it, or you can use an external app like XC-AST to launch it 
if your terminal does not support it


Jump out of queue
Menu for the user
You can define a menu for the user, while waiting. For this menu, you can 
only use one-digit extensions. Define the context for the menu in the 
configuration for the queue to enable this option.

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Re: [Asterisk-Users] Broadvoice gateways!

[trixter http://www.0xdecafbad.com]

On Thu, 2005-04-21 at 15:35 -0400, Gerard Marcel wrote:
> How many gateways does broadvoice have?  Does anyone know?  I know
> about sip.broadvoice.com.  Are there other ones?
> 
> 
> TIA, 
> 
To update my list I decided to write a program to brute force every 3
letter combo aaa-zzz for proxy.XXX.broadvoice.com.  I found a couple I
havent seen on broadvoices page  Here is the full list:  Any one marked
never saw before means *I* never saw it before (and I didnt really look
either).  These may be testing, they may be unstable, they may not be
connected to the same auth server, and then again they may work
perfectly for you.  I make no promises or warantees or guarantees other
than when I did this dns query I did exhaustive aaa-zzz (no numbers or
dashes or longer than 3 letters since that didnt fit in their naming
scheme).


proxy.chi.broadvoice.com
proxy.dca.broadvoice.com
proxy.lab.broadvoice.com  -- never saw before
proxy.lax.broadvoice.com
proxy.mia.broadvoice.com
proxy.nyc.broadvoice.com  -- never saw before

-- 
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] BYOD provider other than broadvoice

[Luki]

>> I have broadvoice and they suck lately.
> Can you elaborate?
Yes, what problems are you having with Broadvoice? Did you try one of
their other proxies? Their DCA server works best for me, 2nd choice
would be their LAX server...

> So the options are?
Good question. Doesn't look like VoicePulse is a clear winner here...

--Luki
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RE: [Asterisk-Users] Zap channels busy. Have to soft hangup.

[Gregory Wiktor - ADCom Corp.]

First, you may want to consider that you do not have enough zap
channels.  Can you tell us something about your system?  How many lines
do you have, and are you bridging incoming calls to an extension or
flashing them through a pbx? 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Thomas
Sent: Thursday, April 21, 2005 12:46 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Zap channels busy. Have to soft hangup.

Hey everybody

I am having really bad nightmares about this. Every day now our phone
system has all of it's 4 zap channels full. I have to soft hangup
zap/1-1 and zap/3-1.

voip*CLI> show channels
Channel  (ContextExtensionPri )   State Appl.
Data
Zap/4-1  (defaults1   )  Up Bridged Call 
Zap/3-1
Zap/3-1  (intern-post 9411 1   )  Up Dial 
Zap/g1/411|70
Zap/2-1  (defaults1   )  Up Bridged Call 
Zap/1-1
Zap/1-1  (intern-post 914105702452 1   )  Up Dial 
Zap/g1/14105702452|70
4 active channel(s)

voip*CLI> soft hangup zap/1-1
Requested Hangup on channel 'Zap/1-1'
-- Hungup 'Zap/2-1'
  == Spawn extension (intern-post, 914105702452, 1) exited non-zero on
'Zap/1-1'
-- Hungup 'Zap/1-1'
voip*CLI> soft hangup zap/1-2
zap/1-2 is not a known channel
voip*CLI> show channels
Channel  (ContextExtensionPri )   State Appl.
Data
Zap/4-1  (defaults1   )  Up Bridged Call 
Zap/3-1
Zap/3-1  (intern-post 9411 1   )  Up Dial 
Zap/g1/411|70
2 active channel(s)
voip*CLI> soft hangup Zap/4-1
Requested Hangup on channel 'Zap/4-1'
-- Hungup 'Zap/4-1'
  == Spawn extension (intern-post, 9411, 1) exited non-zero on 'Zap/3-1'
-- Hungup 'Zap/3-1'
voip*CLI> 

Also I have this weird thing where line Zap/3-1 rings and Zap/4-1 also
picks up. Or if I dial an outside number if somebody dials in at the
same time it happens that instead of dialing out I get the person who
just dialed in. Rather confusing for both of us.

Honestly I did dig around a lot and could not find any specifics about
my issue.

Any help highly appreciated. 

-- Thomas
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RE: [Asterisk-Users] Broadvoice gateways!

[Darren Wright]

thru my cavalier T1 I am getting 20ms  ping times to dca.broadvoice.com...I 
switched all my voicepulse and sixtel to BV.
 
 



From: [EMAIL PROTECTED] on behalf of trixter http://www.0xdecafbad.com
Sent: Thu 4/21/2005 4:25 PM
To: Gerard Marcel; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Broadvoice gateways!



On Thu, 2005-04-21 at 15:35 -0400, Gerard Marcel wrote:
> How many gateways does broadvoice have?  Does anyone know?  I know
> about sip.broadvoice.com.  Are there other ones?

sip.broadvoice.com is a generic placeholder (techincally it points to
proxy.mia.broadvoice.com, but if you follow their directions and
edit /etc/hosts to the fastest gateway then its a placeholder name)

At least:
proxy.mia.broadvoice.com (miami)
proxy.lax.broadvoice.com (la)
proxy.chi.broadvoice.com (chicago)
proxy.dca.broadvoice.com (washington dc - prolly baltimore in reality)

the recommended thing is to to:

#!/bin/sh
for i in mia lax chi dca; do
ping -c 10 -q proxy.${i}.broadvoice.com
done

and use the fastest one to where your server is.

--
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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[Asterisk-Users] using * for Internet call waiting

[Gary Carr]

Wondering if it is possible or if something already exist to setup * to 
offer Internet Call Waiting. For those that do not know what it is, it's a 
small application that runs on a users computer that will pop up a window 
letting them know they have a incoming call and who it is from then they can 
choose to take the call which will disconnect their dialup modem and ring 
their phone or send the call to voice mail.

Thanks,
Gary
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RE: [Asterisk-Users] Asterisk and SER

[Alexander Lopez]

SER is a SIP proxy, Asterisk is a PBX, and application server.

SER passes calls from place to place and does not get in the audio path.

SER uses SIP, * is able to transcode, and convert Protocols.

You can build an IVR, VM, and PBX with Asterisk. SER is like a traffic
cop, where * is the car wash, garage, gas station, etc



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Daniel
Salama
Sent: Thursday, April 21, 2005 3:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Asterisk and SER

I'm a little confused between the pros/cons or benefits of one Asterisk 
or SER. I've been using Asterisk for a little bit and I know it's a 
very powerful and scalable platform (in terms of capacity and 
functionality). However, I've read in some posts that some people are 
using SER + * and I read no SER's page that it can even do what * does.

So, if I have an application where I wish to offer telephony services 
(call origination/termination) with "applications" such as prepaid 
services, conferencing, IVRs, "ACD", dialers, etc, would a single 
installation of * be sufficient? By single I mean * alone (it could be 
a cluster) and not with SER. I guess to some degree it may depend on 
the number of clients, but imagine trying to offer a service similar to 
Vonage.

I buy a "simple" SIP phone, bring it home, sign up for service at 
www.beyourownprovider.com, register my phone on the site and voila - 
ready to make and/or receive calls. I can call other members, similar 
to FWD or I can terminate to PSTN.

Now, multiply this scenario to hundreds or thousands of SIP phones 
spread all over the place, in-front and behind NATs.

What would be your approach? Would you still use SER for anything?

Is this the right list to post this question?

Thanks,
Daniel

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Re: [Asterisk-Users] Multiple Line config help

[Henry Devito]

Don't you have to configure your dialplan to hunt to the next extensions? 
How else would * know to try 94207 if 4207 is busy?
- Original Message - 
From: "kurt x" <[EMAIL PROTECTED]>
To: "Asterisk" 
Sent: Thursday, April 21, 2005 3:08 PM
Subject: [Asterisk-Users] Multiple Line config help

I have aSIPURA 841 that is working on L1 with phone # 4027.  L2 is
configured for 94027.
Both numbers register with Asterisk.  When issuing the command "sip show 
peers"
both numbers have the same IP address but 94027 show its sip port at
5061.  Which I expect is right.  When I dial 4027 it works but when I
dial 94027 I get a "486 busy here" and voice mail picks up.

config below:
sip.conf
[4027]
type=friend
host=dynamic
dtmfmode=rfc2833
context=home
callerid=SIPURA-L1
canreinvite=no
[EMAIL PROTECTED]
[94027]
type=friend
host=dynamic
dtmfmode=rfc2833
context=home
callerid=SIPURA-L2
canreinvite=no
[EMAIL PROTECTED]
extensions.conf
exten => _40xx,1,Answer
exten => _40xx,2,Dial(SIP/${EXTEN},10,t)
exten => _40xx,3,Voicemail(u${EXTEN})
exten => _40xx,4,Hangup
exten => _40xx,103,Voicemail(b${EXTEN})
exten => _40xx,104,Hangup
exten => _940xx,1,Answer
exten => _940xx,2,Dial(SIP/${EXTEN},10,t)
exten => _940xx,3,Voicemail(u${EXTEN})
exten => _940xx,4,Hangup
exten => _940xx,103,Voicemail(b${EXTEN:1})
exten => _940xx,104,Hangup
Thanks,
Kurt
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Re: [Asterisk-Users] BYOD provider other than broadvoice

[Curren C. Calhoun]

I've been using Teliax, with the pay as you go plan for outgoing calls.
Call quality is much better than than our other provider.  They charge a
connection fee of .02 per call but as long as the quality is there, that
doesn't bother me one bit.  Plus they only charge the connection fee if you
call a toll-free number, our other provider charged for the entire call.


> From: Daniel Dziubanski <[EMAIL PROTECTED]>
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Date: Thu, 21 Apr 2005 14:58:57 -0400
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> 
> Subject: RE: [Asterisk-Users] BYOD provider other than broadvoice
> 
> So the options are?
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Adam Robins
> Sent: Thursday, April 21, 2005 1:29 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion; Trevor Harrison
> Subject: RE: [Asterisk-Users] BYOD provider other than broadvoice
> 
> I totally concur.  I switched from Broadvoice to VoicePulse because
> users were complaining about call quality.  Now, the quality is good --
> when it doesn't drop altogether.
> 
> What could be worse than touting your new VoIP system to a client and
> having it drop the call?
> 
>  
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Wiley
> Siler
> Sent: Thursday, April 21, 2005 12:32 PM
> To: Trevor Harrison; Asterisk Users Mailing List - Non-Commercial
> Discussion
> Subject: RE: [Asterisk-Users] BYOD provider other than broadvoice
> 
> Hmmm... Think I would prefer something harder to get provisioned but
> that doesn't drop calls.
> 
> Your users must be forgiving as hell...  Mine would show up with
> pitchforks and torches if calls dropped regularly.
> They get twitchy if the calls just vary too much in quality...  8)
> 
> Cheers,
> Wiley
>  
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Trevor
> Harrison
> Sent: Thursday, April 21, 2005 8:45 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] BYOD provider other than broadvoice
> 
> On 4/21/05, Adam Robins <[EMAIL PROTECTED]> wrote:
>> I drop every 3-4 call with VoicePulse Connect.
> 
> My users are also reporting occasional dropped calls when dialing via
> VoicePulse Connect.
> 
> But I love the ease of use and setup with their service.
> 
> -Trevor
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> privileged. This transmission is sent in trust, for the sole purpose of
> delivery to the intended recipient. If you have received this transmission
> in error, any use, reproduction or dissemination of this transmission is
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> immediately notify the sender by reply email and delete this message and its
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[Asterisk-Users] Re: Libunicall Make Error

[Fabio Vasco]

Angelo,
Try to make all packages (spandsp, libsupertone) with ./configure 
--prefix=/usr

In my case this working fine...
Regards,
Fabio Vasco
_
MSN Busca: fácil, rápido, direto ao ponto.  http://search.msn.com.br
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Re: [Asterisk-Users] Broadvoice gateways!

[trixter http://www.0xdecafbad.com]

On Thu, 2005-04-21 at 15:35 -0400, Gerard Marcel wrote:
> How many gateways does broadvoice have?  Does anyone know?  I know
> about sip.broadvoice.com.  Are there other ones?

sip.broadvoice.com is a generic placeholder (techincally it points to
proxy.mia.broadvoice.com, but if you follow their directions and
edit /etc/hosts to the fastest gateway then its a placeholder name)

At least:
proxy.mia.broadvoice.com (miami)
proxy.lax.broadvoice.com (la)
proxy.chi.broadvoice.com (chicago)
proxy.dca.broadvoice.com (washington dc - prolly baltimore in reality)

the recommended thing is to to:

#!/bin/sh
for i in mia lax chi dca; do
ping -c 10 -q proxy.${i}.broadvoice.com
done

and use the fastest one to where your server is.

-- 
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] Re: Starting with Asterisk-SIP

[jltaylor]

Don't get many hugs around here...

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of ruben
cuevas rumin
Sent: Thursday, April 21, 2005 2:28 PM
To: Moises Silva
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: Starting with Asterisk-SIP


Hi Mosies,

Thanks for your help, now I have a SIP server using asterisk and I can
communicate my two SIP clients with asterisk in the middle :).

This is the first step, but I have to work a lot of yet, so I think I
will disturbe you and the other people in the list (I'm sorry).

Thank you very much for your help.

"Un saludo y un abrazo" ;). (It's an spanish expresion)

   Rubén.

On 4/20/05, Moises Silva <[EMAIL PROTECTED]> wrote:
> Hi again Ruben. Well, it would be good idea to put here what do you
> have in your extensions.conf. Actually i have only "includes" in this
> file, several statements like this:
>
> #include /var/lib/pavoz/extengeneral.iss
>
> So, its easier its administration, but for a simple test you can do
> this in extensions.conf:
>
> [testdialplan]
> exten => _.,1,Dial(SIP/${EXTEN},40,r)
> exten => _.,2,Hangup();
>
> then, in sip.conf:
> [general]
> port=5060
> bindaddr=0.0.0.0
> localnet=192.168.1.0/24 ; here you need your net config net_addr/mask
> tos=lowdelay
> tos=184
> defaultexpirey=120
> disallow=all
> allow=ilbc
> allow=alaw
> allow=ulaw
> defaultcontext=incoming_iss
>
> [15]
> type=friend
> secret=adminpass
> host=dynamic
> nat=no
> dtmfmode=info
> canreinvite=yes
> qualify=yes
> context=testdialplan
>
> [12]
> type=friend
> secret=adminpass
> host=dynamic
> nat=no
> dtmfmode=info
> canreinvite=yes
> qualify=yes
> context=testdialplan
>
> So, you need 2 sip phones (can use kphone) with SIP username 12 and
> 15, using password 'adminpass'.
>
> This is what will happend:
>
> - When you start kphone's, or any other SIP phone, the phones will,
> they will try to make a SIP register with the server that you specify,
> so you have to configure the phones yo try a register in the Asterisk
> Box IP. Asterisk will receive its request for registry and will check
> that the username and secret exists in the file sip.conf, if exists,
> will save the registry and then Asterisk and the phones will be
> connected. Now, when you dial from any sip user, the number will be
> sent to Asterisk, and asterisk will try to find a match in the dialed
> pattern in the context that the SIP entry specifies (in this case the
> parameter context=testdialplan), so , for example, if you dial "12"
> from sip user "15", the 12 will match in the pattern "_.", because the
> dot match anything, you can be more specific an put in extensions.conf
> "_XX", instead of "_.", and Asterisk will only match when you dial a
> number of 2 digits length, and that digits are 0-9 (the X means 0-9).
> You can read more about this in:
>
> http://voip-info.org/wiki-Asterisk+config+extensions.conf
>
> Once the pattern is matched, Asterisk will attempt to execute the
> commands that are there, in this case a Dial() command, that say "Open
> a Channel type SIP, and try to dial to the ${EXTENSION}", ${EXTENSION}
> is a special var, you can read more about asterisk vars in:
>
> http://voip-info.org/wiki-Asterisk+variables
>
> So it will try to dial to a SIP user with the dialed extensión.
>
> So that all, it should work for a small test.
>
> I have studied in Universidad de Guadalajara, in Guadalajara, México.
> Any other people from México here :-)
>
> Good Look!
>
>
> On 4/19/05, ruben cuevas rumin <[EMAIL PROTECTED]> wrote:
> > Hi Moises,
> >
> > Thanks for the reply, and thanks Dana too.
> >
> > I  know that I can to communicate two SIPs phones without Asterisk in
> > the middle. But this isn't my final objective, This is the first step
> > in my project, it mean, I firstly want make works a simple testbed
> > (the one I described in the previous mail), and then step by step
> > configure more difficult testbed.
> > So if you, please, could help me to configure this simple test, I'm
> > will be happy :).
> > I think my problem is the dial plan in the extensions.conf.
> >
> > Ah, I'm studing electronics and comunnication eng, in the University
> > Carlos III of Madrid. Congratulations for your graduation, I hope end
> > in September of this year.
> > Which University do you have study?
> >
> > Best Regards and thank you for your help.
> >
> > On 4/19/05, Moises Silva <[EMAIL PROTECTED]> wrote:
> > > Hi Ruben. You can make a direct IP call. If the 2 sip phones can ping
> > > each other (that is, both are reachable in the network), then in
> > > kphone select the option File > New Call, then type
> > > sip:[EMAIL PROTECTED] , the 'number' is the number wich is configured
> > > in kphone, sipdeviceip will be the IP of the machine that is running
> > > the kphone application. Note that this kind of call does not have
> > > nothing to do with Asterisk, the phones are using sip protocol wi

Re: [Asterisk-Users] BYOD provider other than broadvoice

[Mark Musone]

about 80% of my calls using broadvoice sound _horrible_

all of my customers, vendors, and acquaintences say almost the same
exact thing. "You sound like you're under water".

i've tried every broadvoice proxy, upgraded asterisk till i'm blue in
the face, remoave all firewalls, NATs, setup asterisk on it's own
completely dedicated server..you name it. still "You sound like you're
under water".

This happens whether i call out to people, or people call inbound. I
can hear people on the other end perfectly fine...they almost always
hear horrible jitter.
i'm on two DS3's with actual bandwidth usage of about 5M/sec (i.e. i'm
at like 6% capacity)

My ping times to broadvoice are excellent:

PING proxy.dca.broadvoice.com (147.135.0.128) 56(84) bytes of data.
64 bytes from 147.135.0.128: icmp_seq=1 ttl=51 time=23.1 ms
64 bytes from 147.135.0.128: icmp_seq=2 ttl=51 time=21.0 ms
64 bytes from 147.135.0.128: icmp_seq=3 ttl=51 time=20.7 ms
64 bytes from 147.135.0.128: icmp_seq=4 ttl=51 time=20.9 ms
64 bytes from 147.135.0.128: icmp_seq=5 ttl=51 time=21.1 ms
64 bytes from 147.135.0.128: icmp_seq=6 ttl=51 time=20.3 ms
64 bytes from 147.135.0.128: icmp_seq=7 ttl=51 time=20.9 ms


any SIP<->SIP call _not using broadvoice (i.e. calling from my sipura
device to anoher SIP device across the internet) is perfectly fine..

This has been going on for over a year now...had yet to be better. i'm
about to switch providers also because i simply can't take it anymore.
This is absolutely a broadvoice problem, and since they don't support
asterisk (not that I blame them) i dont have much options. i've been
thinking of just hooking up my Sipura directly to broadvoice to verify
the problem and then "force" them to fix it on the supported device.



-Mark



On 4/21/05, Adam Robins <[EMAIL PROTECTED]> wrote:
> I totally concur.  I switched from Broadvoice to VoicePulse because
> users were complaining about call quality.  Now, the quality is good --
> when it doesn't drop altogether.
> 
> What could be worse than touting your new VoIP system to a client and
> having it drop the call?
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Wiley
> Siler
> Sent: Thursday, April 21, 2005 12:32 PM
> To: Trevor Harrison; Asterisk Users Mailing List - Non-Commercial
> Discussion
> Subject: RE: [Asterisk-Users] BYOD provider other than broadvoice
> 
> Hmmm... Think I would prefer something harder to get provisioned but
> that doesn't drop calls.
> 
> Your users must be forgiving as hell...  Mine would show up with
> pitchforks and torches if calls dropped regularly.
> They get twitchy if the calls just vary too much in quality...  8)
> 
> Cheers,
> Wiley
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Trevor
> Harrison
> Sent: Thursday, April 21, 2005 8:45 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] BYOD provider other than broadvoice
> 
> On 4/21/05, Adam Robins <[EMAIL PROTECTED]> wrote:
> > I drop every 3-4 call with VoicePulse Connect.
> 
> My users are also reporting occasional dropped calls when dialing via
> VoicePulse Connect.
> 
> But I love the ease of use and setup with their service.
> 
> -Trevor
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> are intended solely for addressee. The information may also be legally 
> privileged. This transmission is sent in trust, for the sole purpose of 
> delivery to the intended recipient. If you have received this transmission in 
> error, any use, reproduction or dissemination of this transmission is 
> strictly prohibited. If you are not the intended recipient, please 
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RE: [Asterisk-Users] One touch voicemail on Cisco 7940/60

[Alexander Lopez]

Title: One touch voicemail on Cisco 7940/60









Program
your messages key to ‘voicemail; with:

 

messages_uri:  "voicemail"

 

in your SIP(MACADDRESS).cfg config file…

 

 

 

And in extensions.conf:

 

Exten => voicemail,1,Wait(1)  ;   Wait
a minute to make sure audio is up

Exten => voicemail,2, VoiceMailMain([EMAIL PROTECTED]) ; Change default to whatever your vm context is.

 

-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Morris, Simon
Sent: Thursday, April 21, 2005
3:23 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] One
touch voicemail on Cisco 7940/60

 

Hello,

I'd like to program my Cisco phones to authenticate themselves to voicemail
upon hitting the right button on my 7940/60's

Ideally the voicemail app will detect which extension the call is coming from
and drop the user straight into the menu.

Is this possible?

Many thanks


~sm 






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[Asterisk-Users] Multiple Line config help

[kurt x]

I have aSIPURA 841 that is working on L1 with phone # 4027.  L2 is
configured for 94027.
Both numbers register with Asterisk.  When issuing the command "sip show peers"
both numbers have the same IP address but 94027 show its sip port at
5061.  Which I expect is right.  When I dial 4027 it works but when I
dial 94027 I get a "486 busy here" and voice mail picks up.

config below:

sip.conf
[4027]
type=friend
host=dynamic
dtmfmode=rfc2833
context=home
callerid=SIPURA-L1
canreinvite=no
[EMAIL PROTECTED]

[94027]
type=friend
host=dynamic
dtmfmode=rfc2833
context=home
callerid=SIPURA-L2
canreinvite=no
[EMAIL PROTECTED]

extensions.conf
exten => _40xx,1,Answer
exten => _40xx,2,Dial(SIP/${EXTEN},10,t)
exten => _40xx,3,Voicemail(u${EXTEN})
exten => _40xx,4,Hangup
exten => _40xx,103,Voicemail(b${EXTEN})
exten => _40xx,104,Hangup

exten => _940xx,1,Answer
exten => _940xx,2,Dial(SIP/${EXTEN},10,t)
exten => _940xx,3,Voicemail(u${EXTEN})
exten => _940xx,4,Hangup
exten => _940xx,103,Voicemail(b${EXTEN:1})
exten => _940xx,104,Hangup

Thanks,

Kurt
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[Asterisk-Users] leastrecent queue option

[Dana Olson]

For agent queues, I checked this option out:

; leastrecent - ring interface which was least recently called by this queue

When I use this, if I have 3 agents logged into the queue, and I pump
in 3 calls to the queue, I would expect that one call would go to each
agent, but instead, it doesn't work this way. All calls ring on 3
extensions of one phone only. If I take the call on the phone, then
hang it up, any further calls will go to the next phone.

The problem here, to me, is that when the agent's phone rings, the
pointer to which agent was least recently called doesn't get reset to
the next agent, as one might expect it to... At the very least, I
would think that it would reset at the pickup of the call, if not at
the ring. Am I wrong, or does anyone know if this is fixed in CVS?

Even if it is, I'm not planning on using a development version of
Asterisk in production, so CVS is not an option.

I will continue testing various queue strategies until I find one that
is useable, but any insight into this particular behaviour is
appreciated.

--
Dana
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Re: [Asterisk-Users] 802.1p , precedence and TOS

[Steve Blair]

Eugenio De Vena wrote:
well tos is already set and working ( I debugged the packet in and out ) and
they work
but to and precedence work at layer 3 i.e. in the ip header so they are
considered by routers
and layer3 switches. I want the 802.1p switch to pass voice traffic
priorized on the lan not the wan.
The way to do this is at level 2 ( mac level ) . The only language switch
understand, because they do not
go at IP level ( at least 3com 3300 ... )
 

Ah sorry. You did say 3Com 3300. Yes you are screwed. If you'd consider 
a different
switch perhaps you could gain some additional control over packet 
forwarding. On the
Extreme Summit switches you can mark, remark at the ingress port the 
802.1p COS
by examining the IP precedence /DSCP. This allows an IP phone (for 
example) to
set IP precedence and the ingress switch to add the 802.1p COS tag on 
the way
into the network. Just a thought.


Eugenio
- Original Message - 
From: "Steve Blair" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Thursday, April 21, 2005 6:56 PM
Subject: Re: [Asterisk-Users] 802.1p , precedence and TOS

 

 Why wouldn't you just use tos= or tos=lowdelay
depending upon the config file in question? Remember there is an overlap
between IP precedence bits and some DSCPs for backward compatability.
Honor that overlap and you can use DiffServ processing logic even if your
device can only set an IP prec. value.
Eugenio De Vena wrote:
   

well I thought that with diffserv it could be done, I will double check
 

and
 

let you know,
thanks for you hint.
Eugenio
- Original Message - 
From: "SCollins" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Thursday, April 21, 2005 1:49 PM
Subject: Re: [Asterisk-Users] 802.1p , precedence and TOS


 

I Don't know if this is a solution that is not better suited for your
Linux Distro, using vconfig and a 802.1q/1p aware NIC, on the * Server,
   

to
 

create and prioitize the Ingress and Egress Voice VLAN traffic.
Would you not agree?
Sean



   

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--
ISC Network Engineering
The University of Pennsylvania
3401 Walnut Street, Suite 221A
Philadelphia, PA 19104
voice: 215-573-8396
  215-746-8001
fax: 215-898-9348
sip:[EMAIL PROTECTED]

   

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[Asterisk-Users] Asterisk and SER

[Daniel Salama]

I'm a little confused between the pros/cons or benefits of one Asterisk 
or SER. I've been using Asterisk for a little bit and I know it's a 
very powerful and scalable platform (in terms of capacity and 
functionality). However, I've read in some posts that some people are 
using SER + * and I read no SER's page that it can even do what * does.

So, if I have an application where I wish to offer telephony services 
(call origination/termination) with "applications" such as prepaid 
services, conferencing, IVRs, "ACD", dialers, etc, would a single 
installation of * be sufficient? By single I mean * alone (it could be 
a cluster) and not with SER. I guess to some degree it may depend on 
the number of clients, but imagine trying to offer a service similar to 
Vonage.

I buy a "simple" SIP phone, bring it home, sign up for service at 
www.beyourownprovider.com, register my phone on the site and voila - 
ready to make and/or receive calls. I can call other members, similar 
to FWD or I can terminate to PSTN.

Now, multiply this scenario to hundreds or thousands of SIP phones 
spread all over the place, in-front and behind NATs.

What would be your approach? Would you still use SER for anything?
Is this the right list to post this question?
Thanks,
Daniel
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[Asterisk-Users] One touch voicemail on Cisco 7940/60

[Morris, Simon]

Title: One touch voicemail on Cisco 7940/60






Hello,

I'd like to program my Cisco phones to authenticate themselves to voicemail upon hitting the right button on my 7940/60's

Ideally the voicemail app will detect which extension the call is coming from and drop the user straight into the menu.

Is this possible?

Many thanks


~sm



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Re: [Asterisk-Users] ZAP - outgoing call using different D-Channel eachtime ?

[Matthew Boehm]

Walter Klomp wrote:
> Hi Fellow Asteriskians,
>
> Currently I am having an Asterisk installation piggy-backed behind an
> "intelligent" voice-switch that does the LCR, over E1's using Digium
> TE410p
>
> The problem is that Asterisk always starts channel 1 if the channel
> is free, and only switches to channel 2 if channel 1 is in use...
>
> Can the behaviour be changed so that outgoing calls using ZAP
> channels are on a rotary basis, like 1..30 and then back to 1 for
> each outgoing call?  It has namely happened that channel 1 on the
> other side "hung" in the "Free" state which severely impacted my
> operations as you can imagine...
>
> Hope somebody can shed some light on this. I can't seem to find the
> option in any of the zap conf files and I have been trying to meddle
> with the chan_zap.c source file to no avail...

Don't you mean B-Channel? You only need 1 D-channel.

http://www.voip-info.org/wiki-Asterisk+Zap+channels

I use Dial(ZAP/R1d/2815551212,30) on all our zap calls and this will go in
order on the span.

-Matthew

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Re: [Asterisk-Users] Re: Starting with Asterisk-SIP

[ruben cuevas rumin]

Hi Mosies,

Thanks for your help, now I have a SIP server using asterisk and I can
communicate my two SIP clients with asterisk in the middle :).

This is the first step, but I have to work a lot of yet, so I think I
will disturbe you and the other people in the list (I'm sorry).

Thank you very much for your help.

"Un saludo y un abrazo" ;). (It's an spanish expresion)
  
   Rubén.

On 4/20/05, Moises Silva <[EMAIL PROTECTED]> wrote:
> Hi again Ruben. Well, it would be good idea to put here what do you
> have in your extensions.conf. Actually i have only "includes" in this
> file, several statements like this:
> 
> #include /var/lib/pavoz/extengeneral.iss
> 
> So, its easier its administration, but for a simple test you can do
> this in extensions.conf:
> 
> [testdialplan]
> exten => _.,1,Dial(SIP/${EXTEN},40,r)
> exten => _.,2,Hangup();
> 
> then, in sip.conf:
> [general]
> port=5060
> bindaddr=0.0.0.0
> localnet=192.168.1.0/24 ; here you need your net config net_addr/mask
> tos=lowdelay
> tos=184
> defaultexpirey=120
> disallow=all
> allow=ilbc
> allow=alaw
> allow=ulaw
> defaultcontext=incoming_iss
> 
> [15]
> type=friend
> secret=adminpass
> host=dynamic
> nat=no
> dtmfmode=info
> canreinvite=yes
> qualify=yes
> context=testdialplan
> 
> [12]
> type=friend
> secret=adminpass
> host=dynamic
> nat=no
> dtmfmode=info
> canreinvite=yes
> qualify=yes
> context=testdialplan
> 
> So, you need 2 sip phones (can use kphone) with SIP username 12 and
> 15, using password 'adminpass'.
> 
> This is what will happend:
> 
> - When you start kphone's, or any other SIP phone, the phones will,
> they will try to make a SIP register with the server that you specify,
> so you have to configure the phones yo try a register in the Asterisk
> Box IP. Asterisk will receive its request for registry and will check
> that the username and secret exists in the file sip.conf, if exists,
> will save the registry and then Asterisk and the phones will be
> connected. Now, when you dial from any sip user, the number will be
> sent to Asterisk, and asterisk will try to find a match in the dialed
> pattern in the context that the SIP entry specifies (in this case the
> parameter context=testdialplan), so , for example, if you dial "12"
> from sip user "15", the 12 will match in the pattern "_.", because the
> dot match anything, you can be more specific an put in extensions.conf
> "_XX", instead of "_.", and Asterisk will only match when you dial a
> number of 2 digits length, and that digits are 0-9 (the X means 0-9).
> You can read more about this in:
> 
> http://voip-info.org/wiki-Asterisk+config+extensions.conf
> 
> Once the pattern is matched, Asterisk will attempt to execute the
> commands that are there, in this case a Dial() command, that say "Open
> a Channel type SIP, and try to dial to the ${EXTENSION}", ${EXTENSION}
> is a special var, you can read more about asterisk vars in:
> 
> http://voip-info.org/wiki-Asterisk+variables
> 
> So it will try to dial to a SIP user with the dialed extensión.
> 
> So that all, it should work for a small test.
> 
> I have studied in Universidad de Guadalajara, in Guadalajara, México.
> Any other people from México here :-)
> 
> Good Look!
> 
> 
> On 4/19/05, ruben cuevas rumin <[EMAIL PROTECTED]> wrote:
> > Hi Moises,
> >
> > Thanks for the reply, and thanks Dana too.
> >
> > I  know that I can to communicate two SIPs phones without Asterisk in
> > the middle. But this isn't my final objective, This is the first step
> > in my project, it mean, I firstly want make works a simple testbed
> > (the one I described in the previous mail), and then step by step
> > configure more difficult testbed.
> > So if you, please, could help me to configure this simple test, I'm
> > will be happy :).
> > I think my problem is the dial plan in the extensions.conf.
> >
> > Ah, I'm studing electronics and comunnication eng, in the University
> > Carlos III of Madrid. Congratulations for your graduation, I hope end
> > in September of this year.
> > Which University do you have study?
> >
> > Best Regards and thank you for your help.
> >
> > On 4/19/05, Moises Silva <[EMAIL PROTECTED]> wrote:
> > > Hi Ruben. You can make a direct IP call. If the 2 sip phones can ping
> > > each other (that is, both are reachable in the network), then in
> > > kphone select the option File > New Call, then type
> > > sip:[EMAIL PROTECTED] , the 'number' is the number wich is configured
> > > in kphone, sipdeviceip will be the IP of the machine that is running
> > > the kphone application. Note that this kind of call does not have
> > > nothing to do with Asterisk, the phones are using sip protocol without
> > > asterisk in the middle. When kphone makes a register to asterisk, then
> > > you dont need to specify sip:[EMAIL PROTECTED] you only dial a
> > > number and the number is immediatly sent to asterisk wich routes the
> > > call where the dialplan says.
> > >

Re: [Asterisk-Users] BYOD provider other than broadvoice

[Brian Capouch]

Wiley Siler wrote:
Multiple providers...
I am currently using one for outgoing exclusively due to the low latency
and excellent call quality
You mind saying who that is?
Thx.
B.
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[Asterisk-Users] Broadvoice gateways!

[Gerard Marcel]

How many gateways does broadvoice have?  Does anyone know?  I know
about sip.broadvoice.com.  Are there other ones?


TIA, 


GM
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RE: [Asterisk-Users] Line Noise UPDATE - If you've got line noise, read this

[Paul]

Hm...well, here's something interesting. On my previous box, neither of
the cards were sharing IRQs with anything.now, both cards are on 11,
along with many other things. This could very well be a problem. As far as
the network goes, there is very little traffic and the switch is full duplex
100 megabit. Bandwidth is only a factor on he local lan, since asterisk
dials out through a FXO card(X100P) then it doesn't go across my broadband
connection. Let me know your thoughts.sorry for the verbose output.


Paul


[EMAIL PROTECTED] /sbin]$ ./lspci -v
00:00.0 Host bridge: VIA Technologies, Inc. VT8377 [KT400 AGP] Host Bridge
(rev 80)
Subsystem: Elitegroup Computer Systems: Unknown device 1884
Flags: bus master, 66Mhz, medium devsel, latency 8
Memory at d000 (32-bit, prefetchable) [size=128M]
Capabilities: 

00:01.0 PCI bridge: VIA Technologies, Inc.: Unknown device b198 (prog-if 00
[Normal decode])
Flags: bus master, 66Mhz, medium devsel, latency 0
Bus: primary=00, secondary=01, subordinate=01, sec-latency=0
I/O behind bridge: 9000-9fff
Memory behind bridge: e800-e9ff
Prefetchable memory behind bridge: d800-e7ff
Capabilities: 

00:09.0 Communication controller: Tiger Jet Network Inc. Intel 537
Subsystem: Intel Corp.: Unknown device 0003
Flags: bus master, medium devsel, latency 32, IRQ 11
I/O ports at a000 [size=256]
Memory at eb002000 (32-bit, non-prefetchable) [size=4K]
Capabilities: 

00:0a.0 Communication controller: Tiger Jet Network Inc. Intel 537
Subsystem: Intel Corp.: Unknown device 0003
Flags: bus master, medium devsel, latency 32, IRQ 11
I/O ports at a400 [size=256]
Memory at eb00 (32-bit, non-prefetchable) [size=4K]
Capabilities: 

00:0b.0 Ethernet controller: Linksys Network Everywhere Fast Ethernet 10/100
model NC100 (rev 11)
Subsystem: Linksys: Unknown device 0570
Flags: bus master, medium devsel, latency 32, IRQ 5
I/O ports at a800 [size=256]
Memory at eb001000 (32-bit, non-prefetchable) [size=1K]
Expansion ROM at  [disabled] [size=128K]
Capabilities: 

00:0f.0 RAID bus controller: VIA Technologies, Inc.: Unknown device 3149
(rev 80)
Subsystem: Elitegroup Computer Systems: Unknown device 1884
Flags: bus master, medium devsel, latency 32, IRQ 11
I/O ports at ac00 [size=8]
I/O ports at b000 [size=4]
I/O ports at b400 [size=8]
I/O ports at b800 [size=4]
I/O ports at bc00 [size=16]
I/O ports at c000 [size=256]
Capabilities: 

00:0f.1 IDE interface: VIA Technologies, Inc. VT82C586/B/686A/B PIPC Bus
Master IDE (rev 06) (prog-if 8a [Master SecP PriP])
Subsystem: VIA Technologies, Inc. VT8235 Bus Master ATA133/100/66/33
IDE
Flags: bus master, medium devsel, latency 32
I/O ports at c400 [size=16]
Capabilities: 

00:10.0 USB Controller: VIA Technologies, Inc. USB (rev 81) (prog-if 00
[UHCI])
Subsystem: Elitegroup Computer Systems: Unknown device 1884
Flags: bus master, medium devsel, latency 32, IRQ 10
I/O ports at c800 [size=32]
Capabilities: 

00:10.1 USB Controller: VIA Technologies, Inc. USB (rev 81) (prog-if 00
[UHCI])
Subsystem: Elitegroup Computer Systems: Unknown device 1884
Flags: bus master, medium devsel, latency 32, IRQ 10
I/O ports at cc00 [size=32]
Capabilities: 

00:10.2 USB Controller: VIA Technologies, Inc. USB (rev 81) (prog-if 00
[UHCI])
Subsystem: Elitegroup Computer Systems: Unknown device 1884
Flags: bus master, medium devsel, latency 32, IRQ 11
I/O ports at d000 [size=32]
Capabilities: 

00:10.3 USB Controller: VIA Technologies, Inc. USB (rev 81) (prog-if 00
[UHCI])
Subsystem: Elitegroup Computer Systems: Unknown device 1884
Flags: bus master, medium devsel, latency 32, IRQ 11
I/O ports at d400 [size=32]
Capabilities: 

00:10.4 USB Controller: VIA Technologies, Inc. USB 2.0 (rev 86) (prog-if 20
[EHCI])
Subsystem: Elitegroup Computer Systems: Unknown device 1884
Flags: bus master, medium devsel, latency 32, IRQ 11
Memory at eb003000 (32-bit, non-prefetchable) [size=256]
Capabilities: 

00:11.0 ISA bridge: VIA Technologies, Inc.: Unknown device 3227
Subsystem: Elitegroup Computer Systems: Unknown device 1884
Flags: bus master, stepping, medium devsel, latency 0
Capabilities: 

00:11.5 Multimedia audio controller: VIA Technologies, Inc. VT8233 AC97
Audio Controller (rev 60)
Subsystem: Elitegroup Computer Systems: Unknown device 1884
Flags: medium devsel, IRQ 11
I/O ports at d800 [size=256]
Capabilities: 

00:12.0 Ethernet controller: VIA Technologies, Inc. VT6102 [Rhine-II] (rev
78)
Subsystem: VIA Technologies, Inc.

RE: [Asterisk-Users] BYOD provider other than broadvoice

[Wiley Siler]

Multiple providers...

I am currently using one for outgoing exclusively due to the low latency
and excellent call quality

My incoming DID provider and I are still working out a couple of little
things.

Having a couple of accoutns works nicely though...

W
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Daniel
Dziubanski
Sent: Thursday, April 21, 2005 11:59 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] BYOD provider other than broadvoice

So the options are?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
Robins
Sent: Thursday, April 21, 2005 1:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Trevor
Harrison
Subject: RE: [Asterisk-Users] BYOD provider other than broadvoice

I totally concur.  I switched from Broadvoice to VoicePulse because
users were complaining about call quality.  Now, the quality is good --
when it doesn't drop altogether.

What could be worse than touting your new VoIP system to a client and
having it drop the call?

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
Siler
Sent: Thursday, April 21, 2005 12:32 PM
To: Trevor Harrison; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] BYOD provider other than broadvoice

Hmmm... Think I would prefer something harder to get provisioned but
that doesn't drop calls.

Your users must be forgiving as hell...  Mine would show up with
pitchforks and torches if calls dropped regularly.
They get twitchy if the calls just vary too much in quality...  8)

Cheers,
Wiley
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Trevor
Harrison
Sent: Thursday, April 21, 2005 8:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] BYOD provider other than broadvoice

On 4/21/05, Adam Robins <[EMAIL PROTECTED]> wrote:
> I drop every 3-4 call with VoicePulse Connect.

My users are also reporting occasional dropped calls when dialing via
VoicePulse Connect.

But I love the ease of use and setup with their service.

-Trevor
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Re: [Asterisk-Users] BYOD provider other than broadvoice

[Andrew Kohlsmith]

On April 21, 2005 12:32 pm, Wiley Siler wrote:
> Hmmm... Think I would prefer something harder to get provisioned but
> that doesn't drop calls.

Contact shido6 on IRC, nufone works great for me; I push through about 5000 
min/mo and the only technical problems I've had with them in the past year 
have been my own.  :-)

-A.
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[Asterisk-Users] Problems with app_dbodbc.c

[Matt]

Hi,
Has anyone experience with app_dbodbc.c ?  If so.. any thoughts what
the issue might be trying to compile this?

I'm installing it on a RedHat Enterprise 3.0 server.

[EMAIL PROTECTED] apps]# make
cc -fPIC   -c -o app_dbodbc.o app_dbodbc.c
In file included from /usr/include/asterisk/linkedlists.h:4,
 from /usr/include/asterisk/chanvars.h:17,
 from /usr/include/asterisk/channel.h:19,
 from /usr/include/asterisk/file.h:17,
 from app_dbodbc.c:30:
/usr/include/asterisk/lock.h: In function `ast_mutex_init':
/usr/include/asterisk/lock.h:302: `PTHREAD_MUTEX_RECURSIVE' undeclared
(first use in this function)
/usr/include/asterisk/lock.h:302: (Each undeclared identifier is
reported only once
/usr/include/asterisk/lock.h:302: for each function it appears in.)
app_dbodbc.c: In function `lookupblacklist_exec':
app_dbodbc.c:134: structure has no member named `cid'
app_dbodbc.c:136: structure has no member named `cid'
app_dbodbc.c:139: structure has no member named `cid'
app_dbodbc.c:143: structure has no member named `cid'
app_dbodbc.c:144: structure has no member named `cid'
app_dbodbc.c:147: structure has no member named `cid'
app_dbodbc.c:152: structure has no member named `cid'
app_dbodbc.c: In function `lookupcidname_exec':
app_dbodbc.c:165: structure has no member named `cid'
app_dbodbc.c:166: structure has no member named `cid'
app_dbodbc.c:167: warning: passing arg 3 of `ast_set_callerid' makes
integer from pointer without a cast
app_dbodbc.c:167: too many arguments to function `ast_set_callerid'
app_dbodbc.c: In function `odbcget_exec':
app_dbodbc.c:355: structure has no member named `cid'
make: *** [app_dbodbc.o] Error 1
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Re: [Asterisk-Users] Large Asterisk Setup (~500 Concurrent Calls + Scalability)

[Daniel Salama]

On Apr 21, 2005, at 12:12 PM, Matt Roth wrote:
Our idea is to mount a network drive on a remote machine and save the 
files created by the Monitor command directly to it.  Using this 
setup, the activities for each call would be separated by machine as 
follows:

---Asterisk Master Server---
* The Monitor command is executed out of the dialplan.
* The input and output packets are saved to the mounted drive of the 
Digital Recording Client.
But it seems to me that the load of the Monitor command will remain in 
the master server. The only thing you are really offloading is the 
saving of the media stream which should have minimal impact. My concern 
is that, with a busy installation, issuing too many Monitor commands 
may "overload" CPU usage and have a direct effect on the quality of the 
conversations. Are my assumptions wrong?

---Digital Recording Client---
* soxmix is executed to mix the audio together to produce a single WAV 
file.
* lame is executed to convert the WAV file produced by soxmix into an 
MP3.
* This may be an unnecessary step if soxmix can produce an MP3 
directly.
* The MP3 files are indexed to be easily retrievable by various 
criteria.
   * Directory/file naming conventions, database storage, and custom 
code will be used here.
* An FTP server will be running to allow access to the MP3s.
   * Custom code on the managers' machines will be used to access the 
database for selecting the MP3s to transfer.
* Archiving is done at off-peak hours.
When you tell the Monitor command to combine the resulting .wav files, 
that will also be executed in the master server, which should add to 
its load. Unless you tell the Monitor command NOT to combine the audio 
files and you run that as a separate "croned" processed in the client 
server which will also execute lame after soxmix.

Comments?
- Daniel
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RE: [Asterisk-Users] queue - transfer calls

[Juan Manuel Coronado Zúñiga]

El lun, 18-04-2005 a las 10:18 -0400, Ariel Batista escribió:
>  
>  1. Attended transfer: If the agent transfers the call to someone
> in the management team, the call is immediately transferred,
> and the agent is not able to talk to the manager before. Is
> there a way to allow an agent to talk to the management befora
> actually transferring, so that he can explain the issue in
> advance
> 
> In stead of transferring to the next level support have your agents
> park the call to lets say 700 it should give you something like 701
> then call the next agent tell them what the problem is and to pickup
> exten 701.
> 

The attended transfer works fine but you have to implement it on * from
the CVS. It's still not available in the stable release.

See: http://www.voip-info.org/wiki-Asterisk+config+features.conf


Regards,

-- 
Juan Manuel Coronado Z.
Avatar Ltda. 
Parquesoft
Calle 4 # 8-30, Office 402
Telephone:(+57 2) 822 1214 - ext 300
Popayán, Colombia.
http://www.avatar.com.co



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[Asterisk-Users] adding a thrid asterisk server

[MDM]

To All
I am a new to Asterisk and dialplans have me stumped
I just inherited 2 Asterisk server conected as IAX peers.
Now from what i can tell when Asterisk Server (ask-chic) needs to make a 
call to an extension
which resides on the other server (ask-MAIN) it goes over a IAX channel.

Now i am trying to add that third asterisk server to mix (ask-SD) and i 
figured i would do it in baby steps.
The first thing i did was configure two local SIP client so they could 
call each other and leave voicemail.

Then i tried to  add (ask-SD) to (ask-Main) as a IAX peer. Just like 
(ask-chic).
To test i tried dialing an (ask-SD) ext from a phone off the (ask-MAIN) 
server.

The call did not go through. However i watched it from the CLI> and 
captured the following output. What could be wrong. i am so stumped.

'[EMAIL PROTECTED]' in 15000 ms
-- Accepting call from '' to '7101' on channel 0/23, span 2
-- Executing Dial("Zap/47-1", "IAX2/ask-SD/7101") in new stack
-- Called telx-NY17S/7101
Apr 21 13:52:12 WARNING[147465]: chan_iax2.c:5495 socket_read: Call 
rejected by ask-SD: No such context/extension
-- IAX2/telx-NY17S/2 is circuit-busy
-- Hungup 'IAX2/telx-NY17S/2'
  == Everyone is busy/congested at this time
-- Executing Congestion("Zap/47-1", "") in new stac

here is server (ask-sd) iax.conf and extension.conf files
iax.conf:
[EMAIL PROTECTED] asterisk]# cat iax.conf
[general]
allow=all
jitterbuffer=no
tos=lowdelay
[guest]
type=user
context=guest
callerid="Gust  User"
; BMS-ask-Main-asterisk - Incoming -
;
[ask-mail]
type=user
secret=ask-mail
context=from-ask-main
disallow=all
allow=ulaw
; bmc-asl-main - Outgoing
;
[telx-nyc]
type=peer
username=ask-sd   ; our username
secret=ask-sd; our password
host=192.168.11.30  ; host to connect to
;qualify=yes
;trunk=yes  ; use trunking
[EMAIL PROTECTED] asterisk]#
Extension.conf
[EMAIL PROTECTED] asterisk]#  cat extensions.conf
[general]
static=yes
writeprotect=yes
[bogon-calls]
exten => _.,1,Congestion
[from-sip]
exten => 7101,1,Dial(SIP/7101,20)
exten => 7101,2,Voicemail(u7101)
exten => 7101,102,Voicemail(b7101)
exten => 7101,103,Hangup
exten => 7102,1,Dial(SIP/7102,20)
exten => 7102,2,Voicemail(u7102)
exten => 7102,102,Voicemail(b7102)
exten => 7102,103,Hangup
exten => 7199,1,VoicemailMain(${CALLERIDNUM})
[macro-telx-nyc]
exten => s,1,Noop()
exten => s,2,Dial(IAX2/ask-mail/${ARG1})
[outgoing]
;ingnorepat => 9
exten => _9NXXNXX,1,Noop()
exten => _9NXXNXX,2,Macro(ask-main,${EXTEN})
exten => _9NXXNXX,3,Playback(invalid)
exten => _9NXXNXX.4,Hangup
[EMAIL PROTECTED] asterisk]#
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Re: [Asterisk-Users] Bristuff and Belgium

[Kristof Hardy]

David Masure wrote:
Does anyone has any experience with bristuff in Belgium... ???
Yes.
version 1.0.6).  It works although I received a lot of messages from 
zaphfc telling me it didn't receive the correct number of frames for 
both lines.  Sometimes too much, sometimes too few...  Probably a buffer 
I have not received any problems of that kind 'yet', could you show some 
logging? What card are you using?

Maybe let know what version of 'other' software you are using? and what 
distribution do you use?

cheers.
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Re: [Asterisk-Users] Queues configuration

[Daniel Salama]

Would you happen to have some sample config on how to do this? Is this 
done in queues.conf or in the dial plan? Even pseudocode will help.

Thanks,
Daniel
On Apr 21, 2005, at 2:55 PM, Henry Devito wrote:
3) When callers call into the * box and the agents are busy, they 
will be put on the queue. Now, I wish to be able to tell the callers 
they are in position X in the queue. However, if after, say, 30 
seconds no agent is available yet, I wish to give the caller the 
opportunity to listen to a IVR where s/he can make a decision whether 
to continue holding, be transfered to another agent group, or leave a 
message. Is this possible? Does anyone have any sample of how to do 
this?
This is possible.  The person can jump out of queue at anytime or you 
can set an 'overflow' to an IVR menu.

Daniel Salama
[EMAIL PROTECTED]
Voice: (954) 655-8051
Fax  : (954) 252-3988

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[Asterisk-Users] Re: chan_unicall.c compile error

[Fabio Vasco]

Hector,
This is my Linux Fedora Core 3 version info
[EMAIL PROTECTED] proc]# cat version
Linux version 2.6.9-1.667 ([EMAIL PROTECTED]) (gcc version 
3.4.2 20041017 (Red Hat 3.4.2-6.fc3)) #1 Tue Nov 2 14:41:25 EST 2004

I am get Asterisk with the "cvs -r stable", i supose the version is 1.0.6.
Thanks for your help.
Fabio
_
MSN Busca: fácil, rápido, direto ao ponto.  http://search.msn.com.br
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RE: [Asterisk-Users] BYOD provider other than broadvoice

[Daniel Dziubanski]

So the options are?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam Robins
Sent: Thursday, April 21, 2005 1:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Trevor Harrison
Subject: RE: [Asterisk-Users] BYOD provider other than broadvoice

I totally concur.  I switched from Broadvoice to VoicePulse because
users were complaining about call quality.  Now, the quality is good --
when it doesn't drop altogether.

What could be worse than touting your new VoIP system to a client and
having it drop the call?

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
Siler
Sent: Thursday, April 21, 2005 12:32 PM
To: Trevor Harrison; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] BYOD provider other than broadvoice

Hmmm... Think I would prefer something harder to get provisioned but
that doesn't drop calls.

Your users must be forgiving as hell...  Mine would show up with
pitchforks and torches if calls dropped regularly.
They get twitchy if the calls just vary too much in quality...  8)

Cheers,
Wiley
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Trevor
Harrison
Sent: Thursday, April 21, 2005 8:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] BYOD provider other than broadvoice

On 4/21/05, Adam Robins <[EMAIL PROTECTED]> wrote:
> I drop every 3-4 call with VoicePulse Connect.

My users are also reporting occasional dropped calls when dialing via
VoicePulse Connect.

But I love the ease of use and setup with their service.

-Trevor
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Re: [Asterisk-Users] 802.1p , precedence and TOS

[Eugenio De Vena]

well tos is already set and working ( I debugged the packet in and out ) and
they work
but to and precedence work at layer 3 i.e. in the ip header so they are
considered by routers
and layer3 switches. I want the 802.1p switch to pass voice traffic
priorized on the lan not the wan.
The way to do this is at level 2 ( mac level ) . The only language switch
understand, because they do not
go at IP level ( at least 3com 3300 ... )
Eugenio

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

Sent: Thursday, April 21, 2005 6:56 PM
Subject: Re: [Asterisk-Users] 802.1p , precedence and TOS


>
>   Why wouldn't you just use tos= or tos=lowdelay
> depending upon the config file in question? Remember there is an overlap
> between IP precedence bits and some DSCPs for backward compatability.
> Honor that overlap and you can use DiffServ processing logic even if your
> device can only set an IP prec. value.
>
> Eugenio De Vena wrote:
>
> >well I thought that with diffserv it could be done, I will double check
and
> >let you know,
> >thanks for you hint.
> >Eugenio
> >
> >- Original Message - 
> >From: "SCollins" <[EMAIL PROTECTED]>
> >To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> >
> >Sent: Thursday, April 21, 2005 1:49 PM
> >Subject: Re: [Asterisk-Users] 802.1p , precedence and TOS
> >
> >
> >
> >
> >>I Don't know if this is a solution that is not better suited for your
> >>Linux Distro, using vconfig and a 802.1q/1p aware NIC, on the * Server,
to
> >>create and prioitize the Ingress and Egress Voice VLAN traffic.
> >>
> >>Would you not agree?
> >>
> >>Sean
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >
> >___
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> >
>
> -- 
>
> ISC Network Engineering
> The University of Pennsylvania
> 3401 Walnut Street, Suite 221A
> Philadelphia, PA 19104
>
>
> voice: 215-573-8396
>
>215-746-8001
>
> fax: 215-898-9348
>
> sip:[EMAIL PROTECTED]
>
>
>

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