[asterisk-users] Queue announcement issues

[Phil Jordan]

I tried sending this to asterisk-bsd a couple of days ago

I've been using Asterisk in various versions on FreeBSD for some time
now, but I've only just got to messing around with ACD.

I find I can't get any in-queue announcements to work, be they either
periodic or queue position announcements. I've read all of the recent
posts on this, and the closest thing to my problem I can find is a
chap on asterisk-users who reported something similar a month ago, but  
worked around it by  not selecting the "r" queue option. That doesn't  
work for me (I wasn't using it anyway) - and I've dropped all queue  
options as a test, still

to no avail.

Before I get round to posting my configs for critique, is this a BSD  
port issue? I see stuff around on the net re the BSD port, to the  
effect that there are some issues with Asterisk applications which are  
related to timers. What exactly is meant by that please? Is that what  
I'm suffering from here or is it something entirely different?


My environment is FreeBSD 5.3.18 (in production use), Asterisk 1.2.9.1
from Ports, no hardware telephony cards (using a wholesale IAX
provider). Calls are being routed in via IAX2 and the agents are on
IAX2 hardphones, not that that latter makes any difference methinks.

Many thanks in advance for any help that can be offered.

Phil




- End forwarded message -




___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] G729 License to Bridge calls through VOIP provider?

[Erik]

Marco Mouta wrote:


By the way could any one tell me wich is the Bandwith with IP over
head for this codec. about 8kb/s?


Let's do some calculations on that:

g729a 20ms results in 20 bytes RTP payload in each packet, in order to traverse 
the OSI model there's some headers that need to be added,
as RTP gets an RTP header, the RTP packet gets an UDP header and the UDP 
datagram gets an IP header:
So add a 12 byte RTP header, 8 byte UDP header and a 20 Byte IP header

This results in:
20 byte RTP payload + 12 byte RTP header + 8 byte UDP header + 20 byte IP 
header=60 byte on the ip layer.
Thats a 40 byte overhead (so 2/3 of the packet is just headers :) and were 
still only on the IP layer now)

So to transmit just 1 RTP packet you are actualy transmitting 60 bytes on the 
IP layer, so in order to get the real used bandwidth we
need to knowhow many packets we are sending and on which medium 
(DSL/ethernet/slip/smokesignals):

20 ms results in 50 packets/s so: 50 packet/s *60 bytes/packet=3000 bytes/s
that's 300*8=24000 bit/s total bandwidth on the IP layer so the overhead is 
24000-8000=16000 bit/s.

The fun starts if you are going to send this over DSL, let's continue the 
calculation:

50 packets of 60 byte IP, add the 2 byte PPPoA header for DSL= 62 bytes per 
packet.
However, DSL operates with 53 bytes ATM cells, in which you can fit 48 bytes payload (and a 5 byte header) so in order to transmit the 62 bytes of 
data you need: 62/48=2 ATM cells.


Why 2 cells you say? Because ATM can't utilize the unused part of cells, so to 
transmit 62 bytes you use the same amount of bandwith (on dsl) as you
would use to transmit 96 (48*2) bytes.

So 1 RTP packet uses 96 bytes on the DSL line, as you already know we have 50 
packets/s so that's 50 packets/s*2 cells=100 Cells/s
100 cells/s * 53 byte = 53000 bytes/s on the DSL line thats 424000 bits/s to 
transmit a 8 kbit/s stream :)

So the total overhead is 424000-8000=416000 bit/s overhead.


If you would use G723 with a 10 ms frequency it gets even worse :)
G723 on 10 ms produces 8 byte RTP payload per packet, so with headers that's 48 
bytes on the IP layer, but now were sending 100 packets/s
so: 48*100=4800 bytes/s --> 38400 bit/s on the ip layer
On DSL this would result in 50 byte packets (pppoa header) with won't fit in 1 
cell, so you would use 2 cells for each IP packet.
100 packets/s * 2 cells = 200 cells/s
that's 200 cells * 53 bytes/cell = 10600 bytes/s on the DSL line
10600*8=84800 bit/s to transmit a 6400 bit/s stream --> 78400 bit/s overhead

If you would use G723 with 20 ms (16 byte RTP payload) you only have 42400 at 
the DSL layer, so by adjusting the sample frequency you could cut the
overhead in half :)


Erik Versaevel
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] odd sound between SIP & IAX clients

[Tim Panton]

On 26 Jul 2006, at 03:04, Joseph Love wrote:

The issue which occurs is that the audio from the SIP client to the  
IAX client will spend most of it's time sounded very robotic, and  
garbled.  It is possible, although very difficult to understand  
someone who is on the SIP phone.


I have asterisk 1.2.10 configured with realtime with both IAX and  
SIP clients.
The SIP clients include a Grandstream gxp2000 hard phone, and  
Counterpath's X-Lite 3 (for windows) softphone.
The IAX clients tested include idefisk (both windows & mac),  
JakenIAX, and LoudHush.
GSM is the preferred codec of both IAX & SIP clients, and is indeed  
the codec being used in all tests.


Audio from the IAX to the SIP client does not experience any  
issues.  SIP to SIP (and presumably, although untested, IAX to IAX)  
communication does not experience any issues.


We also have a T1 card through which many calls have been placed,  
both from the IAX and SIP phones, without any audio issues  
occurring, in either case.


If it weren't for that there have been multiple clients tested to  
verify this robotic sound, I would cough it up to it being a  
incompatability between the particular clients, but this occurs on  
all SIP-IAX communication that has been tried.


I'm running out of options as SIP-IAX intercommunication is kinda  
expected (and necessary for me), and out of good softphones for the  
mac, as most of the mac-compatible softphones are IAX2-based.


Please let me know what additional information is needed to help me  
debug this problem.


We have had reports like this, and it is looking like the iax  
jitterbuffer is the culprit.
Try adding jitterbuffer=no to the general section of iax.conf and see  
if that helps.



Tim Panton

www.mexuar.com



___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Connecting branch offices through IPsec tunnel --latency effects?

[Tim Panton]

On 26 Jul 2006, at 02:00, Rich Adamson wrote:


Dan Austin wrote:

Stephen wrote:

If I connect two offices through an IPsec tunnel, what is the impact

on

latency, and does it noticeably affect calls?
That would depend a lot on the equipment that services the IPSEC  
tunnel

endpoints.

Has anyone out there tried this? What were the effects?

I've run small to mid size offices (20 to 60 people) over IPSEC
tunnels during periods of internal network failures with good  
results.

That includes offices on the opposite side of the world with one-way
latency normally around 100ms, but often up to 160ms.
Using commercial IPSEC endpoints, or OpenSWAN on a decent system only
adds a couple of ms, if that.


I might add that I did a little research for a non-voip project  
relative to what cisco 28xx routers could sustain in terms of ipsec- 
vpn throughput. The cisco doc's report 55 mbs sustained throughput.


On the flip side, the older cisco routers can't sustain 500 kbs  
without adding a hardware encryption board to the router.


So, you are probably very right with the "depends a lot on the  
equipment". ;)


In some cases it seems that VPN might get you less latency!
I hear that some 3g carriers prioritize VPN traffic above VOIP traffic.


Tim Panton

www.mexuar.com



___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Snom 360

[Christian Stredicke]

Welcome to VoIP... Your operator needs to take care about 
QoS when you are doing a download. Alternatively, there are some more-or-less 
tricky and buggy tricks to stop downloads when you are talking; this needs to be 
done on your IAD.
 
See for example http://www.voip-info.org/wiki-QoS.
 
CS

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Dovid 
  BenderSent: Wednesday, July 26, 2006 12:46 PMTo: 
  asterisk-users@lists.digium.comSubject: [asterisk-users] Snom 
  360
  
  Hello List,
  I am trying to configure QoS for the SNOM 360. I 
  plugged the phone in to the internet and then had the customers computer plug 
  in to the phone. Whith default settings when I talked on the phone it was 
  great. As soon as I started a big download the phone call became unclear. I 
  tried messing around with some settings but to no avail. Anyone have any 
  advice ? Thanks.
   
  Dovid
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Just bought a Polycom 501 - Ifeellike myGXP-2000 was better...

[Martin Joseph]

On Jul 25, 2006, at 12:52 PM, Mike wrote:


I didn't want to start a war either.  It was simply an opinion that I
thought was worth expressing after reading all those "GXP-2000 sucks"
messages in the past.

It's still just an opinion, I am certainly not trying to build a  
consensus.


Thanks for all those who helped me get the phone working.


Please do give us another update after you have used the phone for a  
while?



___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Change current working directory to /tmp

[Luki]

Patrick,

I run asterisk in a chroot'ed environment and within it I cd into /tmp
just before starting asterisk. The kernel happily dumps the core files
into that /tmp directory. As far as I can tell, this behavior has not
changed recently and it definitely worked for 1.2.7.1.

You can also force a directory where core files should be dumped with:

mkdir /corefiles
echo /corefiles/core > /proc/sys/kernel/core_pattern

The kernel will then dump all core files for any process into the
/corefiles directory.

--Luki


On 7/25/06, Patrick Cervicek <[EMAIL PROTECTED]> wrote:

To get a core file, I started Asterisk with
cd /tmp
/usr/sbin/asterisk -g -p -U asterisk

Unfortunately, asterisk always changes the cwd (current working
directory) to '/'
I checked that in /proc/.../cwd and with strace. I start asterisk as
User 'asterisk', therefor it is not possible to write core dumps in /.

How can I force asterisk to use /tmp as cwd?

I have
Debian Sarge with Asterisk 1.2.7.1

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] PRI vs "Digital Trunk"

[Barry D. Hassler]

Thanks for the answers folks. I think I now understand that indeed, the "Digital Trunk" is really nothing more than the same analog services, just delivered digitally. no outbound ANI (of especial interest), etc

This is in the Dayton Ohio area, major telco vendor. I'll have to pursue the differences with them further, as the pricing is about double for the PRI vs the "Digital Trunk". I'd like to move 7 analog lines to a digital interface, but just can't cost-justify it in this scenario :-(


On Tue, 2006-07-25 at 15:25 -0400, Barry D. Hassler wrote:


Hi, can someone enlighten me as to the difference between a PRI and a
"Digital Trunk" (other than cost)?

I do understand PRI (B-channel signaling, incoming/outgoing call setup,
D channel for voice/data, etc), but I'm not quite sure how that compares
with what my vendor is calling a "Digital Trunk" (specifically in
contrast to a PRI). The PRI is about twice the cost. 

If this is just a channelized T1 (24 64k voice/data "channels'), would
they each be assigned a specific phone number, or is there further
flexibility in sending/receiving calls, callerid (receive or send), etc?

Feeling ignorant here

Thanks!
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users











Barry D. Hassler
President 

HCST
2332 Grange Hall Road
Beavercreek, Ohio 45431-2345
http://www.hcst.net/


 


[EMAIL PROTECTED] 
+1 937-427-9000    
+1 937-427-8706 FAX    
FWD: 3934279000 (655480) 



HCST*Net Support Issues: please email [EMAIL PROTECTED] 
Billing Issues: Please email [EMAIL PROTECTED]





___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Snom 360

[Dovid Bender]

Hello List,
I am trying to configure QoS for the SNOM 360. I 
plugged the phone in to the internet and then had the customers computer plug in 
to the phone. Whith default settings when I talked on the phone it was great. As 
soon as I started a big download the phone call became unclear. I tried messing 
around with some settings but to no avail. Anyone have any advice ? 
Thanks.
 
Dovid
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Just bought a Polycom 501 - I feellike myGXP-2000 was better...

[Michael Graves]

On Tue, 25 Jul 2006 08:16:58 -0400, Dean Collins wrote:



>I disagree with you entirely which is why I've been such a strong

>supporter for [EMAIL PROTECTED] and now of Trixbox.

>

>There is no reason why this technology shouldn't be installed directly

>by an end user.

>

>It's no more complicated than messenger with LCS etc, it's just MS does

>a better job of hiding the complexity.

>



I don't entirely disagree. I said "a typical end user." There are end users who can and should undertake such matters. As a Linux newby I'm one of those who can do such things with some research and occasional assistance. I am intent upon self reliance, and I don't bash the software for my own inexperience or failings. However, there is a broad range of users out thereconsider my wife for examplewho would have no clue and should not even try. She loves her Aastra 480i CT phone, but can't be bothered even entering numbers into the phones directory...good thing I load that off the server ;-) 



I looked at [EMAIL PROTECTED] for my office but went another direction. It's handy but in some ways limiting. I think that making the effort to learn the setups by hand has had some rewards. I did pay someone a token amount to write my first configs to get me started three years ago. I eventually settled on Astlinux running on embedded hardware as preferable for my application.



I can't speak to the complexity of LCS, but I have noted that a m0n0wall router is a whole lot easier than setting up the equivalent functions in Windows 2003 Server.



Not everything needs to be easy. Just because something is easy doesn't make it good. Some stuff needs to be very goodand the benefits are worth the learning curve.



Michael



> 

>Cheers,

>Dean

>

>

>> -Original Message-

>> From: [EMAIL PROTECTED] [mailto:asterisk-users-

>> [EMAIL PROTECTED]] On Behalf Of Michael Graves

>> Sent: Tuesday, 25 July 2006 12:16 AM

>> To: Asterisk Users Mailing List - Non-Commercial Discussion

>> Subject: RE: [asterisk-users] Just bought a Polycom 501 - I feellike

>myGXP-2000

>> was better...

>> 

>> On Mon, 24 Jul 2006 18:34:24 -0700, shadowym wrote:

>> 

>> >One of the problems with opinions on these forums IMHO is that they

>are

>> >mostly from technical types and not typical end users.

>> 

>> Typical end users would have no business configuring a PBX such as

>Asterisk. A

>> typical end user would leave that to a professional.

>> 

>> OTOH, those who are responsible for PBX systems often answer to

>"typical end

>> users" in a fashion, and so very likely have faced a number of

>opinions about

>> things...well founded and

>> otherwise.

>> 

>> Michael

>> 

>> 

>> 

>> ___

>> --Bandwidth and Colocation provided by Easynews.com --

>> 

>> asterisk-users mailing list

>> To UNSUBSCRIBE or update options visit:

>> http://lists.digium.com/mailman/listinfo/asterisk-users

>___

>--Bandwidth and Colocation provided by Easynews.com --

>

>asterisk-users mailing list

>To UNSUBSCRIBE or update options visit:

> http://lists.digium.com/mailman/listinfo/asterisk-users

>










___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Recommend hard phone which supports IAX2?

[Michael Graves]

On Tue, 25 Jul 2006 19:43:58 +0100, Tim Panton wrote:


>On 25 Jul 2006, at 16:23, Stephen Bosch wrote:

>> Hi:
>>
>> I'm setting up a branch office, but I don't want to trunk from the  
>> main
>> office because I don't want to introduce any more latency. Also, the
>> office will have only a single extension, so I can't justify the  
>> expense
>> of a second Asterisk server for it.
>>
>> SIP is a pain when going through firewalls, and I'm worried about the
>> latency that would come with using an IPsec tunnel between the two
>> sites, so I'm looking for an IAX2 supporting hard phone, and want to
>> hear recommendations from people who have had direct experience  
>> with such.
>>
>> What are the best IAX2 hard phones?

>I've got a couple of IAX hardphones, with PA168, they are useable,
>but only just. They are hard to hang up (which is a design problem)
>and  a pain to get transfer working (which is a software problem).

>Much as I love IAX, I advise you to buy a decent SIP phone
>(SNOM?).

>At home I have a SIP phone and an nslu2 running asterisk, just to act  
>as a
>protocol converter, but any old 486 or PII will do the
>trick.

I echo this sentiment. Except that I'd recommend Astlinux on a WRAP or Soekris 
board. Small, low power,fanless, boots from CF or USB key and able to transcode 
between G.711 and 
G.729a. Astlinux rocks!

Michael



___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] MWI from Octel to Asterisk

[Mike Diehl]

Hi all.

We are in the process of doing some VoIP testing with the intent to eventually 
replace our 5ESS phone switch.

However, during the transition period, we'd like to be able to use our 
existing voicemail system which is Octel.  It's pretty easy to figure out how 
to change the dialplan to send unanswered calls to the Octel VM system, but 
is there any way to retrieve MWI from it from within Asterisk.  We have ISDN 
phones that have a Message Light that we don't want to break.

Any hints would be appreciated.

Mike.
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] odd sound between SIP & IAX clients

[Rich Adamson]

Joseph Love wrote:
The issue which occurs is that the audio from the SIP client to the IAX 
client will spend most of it's time sounded very robotic, and garbled.  
It is possible, although very difficult to understand someone who is on 
the SIP phone.


I have asterisk 1.2.10 configured with realtime with both IAX and SIP 
clients.
The SIP clients include a Grandstream gxp2000 hard phone, and 
Counterpath's X-Lite 3 (for windows) softphone.
The IAX clients tested include idefisk (both windows & mac), JakenIAX, 
and LoudHush.
GSM is the preferred codec of both IAX & SIP clients, and is indeed the 
codec being used in all tests.


Audio from the IAX to the SIP client does not experience any issues.  
SIP to SIP (and presumably, although untested, IAX to IAX) communication 
does not experience any issues.


We also have a T1 card through which many calls have been placed, both 
from the IAX and SIP phones, without any audio issues occurring, in 
either case.


If it weren't for that there have been multiple clients tested to verify 
this robotic sound, I would cough it up to it being a incompatability 
between the particular clients, but this occurs on all SIP-IAX 
communication that has been tried.


I'm running out of options as SIP-IAX intercommunication is kinda 
expected (and necessary for me), and out of good softphones for the mac, 
as most of the mac-compatible softphones are IAX2-based.


Please let me know what additional information is needed to help me 
debug this problem.


Can you try different codecs just to rule out any issues with that? 
E.g., if both devices use ulaw, do you still have the same problem?


I've used both iaxcomm and x-lite to communicate with cisco, polycom, 
grandsteam, etc, without that type of problem.


Is it possible to obtain an ethereal trace of both the iax and sip/rtp 
streams in the same trace?  If so, a couple hundred packets should be 
more then enough to see what's going on.


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Clocking Multiple T1 Cards

[Rich Adamson]

Shaw Terwilliger wrote:

Andrew Kohlsmith wrote:
What I was trying to state was that if you have two data streams that are 
solidly clocked but out of phase, you will not encounter any of these issues.  
If the clock period of either (or both) drifts then yes, you will run into 
trouble.


So it sounds like Asterisk can't synchronize the clocks between the
Digium and Sangoma boards (or any two PCI boards), and this just may be
a limitation of the T1-peripheral-on-PCI architecture.  But it really
shouldn't matter because of the nature of my setup: errors caused by
timing mismatch between the PRI and channel banks won't cause noticeable
quality issues.  Do I have it right?


Yes, for voice there isn't any need to bit-sync the two T1's.  Both the 
digium and sangoma cards gather up bytes (or groups of bits from the 1.5 
mbs stream) and sends them via the pci bus to the drivers, then back 
across the pci bus to the second card. If a "group of bits" is delayed 
or lost in the process (due to significant clocking differences), the 
voice user isn't going to notice. However, as others have noted, it will 
make a difference for modem users (eg, fax).


Saying it a bit different, if the two T1 clocks were significantly out 
of sync, the flow may drop one out of a thousand bytes (or more) per 
voice channel, which the average human won't hear.



___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] odd sound between SIP & IAX clients

[Joseph Love]

The issue which occurs is that the audio from the SIP client to the  
IAX client will spend most of it's time sounded very robotic, and  
garbled.  It is possible, although very difficult to understand  
someone who is on the SIP phone.


I have asterisk 1.2.10 configured with realtime with both IAX and SIP  
clients.
The SIP clients include a Grandstream gxp2000 hard phone, and  
Counterpath's X-Lite 3 (for windows) softphone.
The IAX clients tested include idefisk (both windows & mac),  
JakenIAX, and LoudHush.
GSM is the preferred codec of both IAX & SIP clients, and is indeed  
the codec being used in all tests.


Audio from the IAX to the SIP client does not experience any issues.   
SIP to SIP (and presumably, although untested, IAX to IAX)  
communication does not experience any issues.


We also have a T1 card through which many calls have been placed,  
both from the IAX and SIP phones, without any audio issues occurring,  
in either case.


If it weren't for that there have been multiple clients tested to  
verify this robotic sound, I would cough it up to it being a  
incompatability between the particular clients, but this occurs on  
all SIP-IAX communication that has been tried.


I'm running out of options as SIP-IAX intercommunication is kinda  
expected (and necessary for me), and out of good softphones for the  
mac, as most of the mac-compatible softphones are IAX2-based.


Please let me know what additional information is needed to help me  
debug this problem.


Thanks,
-Joe

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Sangoma Stops Receiving Calls

[Alex Robar]

I'm in Canada, Greater Toronto Area region. I will give a phone call to Sangoma tomorrow morning and post back with their reply.AlexOn 7/25/06, shadowym
 <[EMAIL PROTECTED]> wrote:





What country are you in?
 
Please let us know what Sangoma tells 
you.

  
  
  From: Alex Robar [mailto:[EMAIL PROTECTED]
] 
  Sent: Tuesday, July 25, 2006 2:12 PMTo: Asterisk Users 
  Mailing List - Non-Commercial DiscussionSubject: [asterisk-users] 
  Sangoma Stops Receiving Calls
  Hi all,I have a Sangoma A200 card with hardware echo 
  cancellation. The card has 12 ports (10 of which are active; All FXO). Twice 
  on this particular card I've seen all ports simply stop receiving incoming 
  calls. There is no other indication of this, however. I am able to place 
  outgoing calls just fine, and call other extensions without issue. When 
  someone calls in, the line just rings and rings, with no indication that the 
  card even sees the calls. I'm not even sure where to begin looking into this. 
  Could anyone give me some pointers as to what I might need to be looking for? 
  I'll be giving Sangoma tech support a call, but if anyone has any 
  debugging pointers, they would be much appreciated.Thanks,Alex-- Alex Robar
[EMAIL PROTECTED] 


___--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   
http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar[EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] T.38 call with t38 in original SDP fails

[David Hindmarsh]

Hi All,

I am trying to use some ATAs which support T.38 in the initial invite.

Asterisk(trunk) presently rejects the call as no audio is offered in the
initial SDP.

With ATAs behind NAT the usual method of switching to t38 seems to fail.

This method may resolve the issue of a re-invite to an ATA behind a NAT.

Any ideas would be welcome.

David Hindmarsh

--
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.1.394 / Virus Database: 268.10.4/399 - Release Date: 25/07/2006

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Play sounds to the callee and the caller synchronously when call begins

[snlee]

Hi list,

I use the Dial application with L option and set the variables as
follows:

  LIMIT_PLAYAUDIO_CALLER = yes
  LIMIT_PLAYAUDIO_CALLEE = yes
  LIMIT_CONNECT_FILE = my_sound_filename

When the call begings, it plays my_sound_filename to the caller and
then plays to the callee. How can I make it to play sounds to the
caller and callee synchronously.

thx.

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Sangoma Stops Receiving Calls

[shadowym]

What country are you in?
 
Please let us know what Sangoma tells 
you.

  
  
  From: Alex Robar [mailto:[EMAIL PROTECTED] 
  Sent: Tuesday, July 25, 2006 2:12 PMTo: Asterisk Users 
  Mailing List - Non-Commercial DiscussionSubject: [asterisk-users] 
  Sangoma Stops Receiving Calls
  Hi all,I have a Sangoma A200 card with hardware echo 
  cancellation. The card has 12 ports (10 of which are active; All FXO). Twice 
  on this particular card I've seen all ports simply stop receiving incoming 
  calls. There is no other indication of this, however. I am able to place 
  outgoing calls just fine, and call other extensions without issue. When 
  someone calls in, the line just rings and rings, with no indication that the 
  card even sees the calls. I'm not even sure where to begin looking into this. 
  Could anyone give me some pointers as to what I might need to be looking for? 
  I'll be giving Sangoma tech support a call, but if anyone has any 
  debugging pointers, they would be much appreciated.Thanks,Alex-- Alex Robar[EMAIL PROTECTED] 

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Change current working directory to /tmp

[Patrick Cervicek]

To get a core file, I started Asterisk with
cd /tmp
/usr/sbin/asterisk -g -p -U asterisk

Unfortunately, asterisk always changes the cwd (current working 
directory) to '/'
I checked that in /proc/.../cwd and with strace. I start asterisk as 
User 'asterisk', therefor it is not possible to write core dumps in /.


How can I force asterisk to use /tmp as cwd?

I have
Debian Sarge with Asterisk 1.2.7.1

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] New message

[Rich Adamson]

Eric "ManxPower" Wieling wrote:
Someone connected to the Asterisk console using "asterisk -r" then typed 
"logger reload" then exited the session.


Ira wrote:
This morning I found this message on my Asterisk Console. Does it mean 
I should be concerned about the security of my system?


-- Remote UNIX connection
== Parsing '/etc/asterisk/logger.conf': Found
Asterisk Event Logger Restarted
-- Remote UNIX connection disconnected





You'll also see the exact same log entries "if" you run a cron job to 
log into asterisk and rotate the logs. See it all the time, but we wrote 
the script as well. :)




___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Connecting branch offices through IPsec tunnel --latency effects?

[Rich Adamson]

Dan Austin wrote:

Stephen wrote:

If I connect two offices through an IPsec tunnel, what is the impact

on

latency, and does it noticeably affect calls?

That would depend a lot on the equipment that services the IPSEC tunnel
endpoints.


Has anyone out there tried this? What were the effects?

I've run small to mid size offices (20 to 60 people) over IPSEC
tunnels during periods of internal network failures with good results.
That includes offices on the opposite side of the world with one-way
latency normally around 100ms, but often up to 160ms.

Using commercial IPSEC endpoints, or OpenSWAN on a decent system only
adds a couple of ms, if that.


I might add that I did a little research for a non-voip project relative 
to what cisco 28xx routers could sustain in terms of ipsec-vpn 
throughput. The cisco doc's report 55 mbs sustained throughput.


On the flip side, the older cisco routers can't sustain 500 kbs without 
adding a hardware encryption board to the router.


So, you are probably very right with the "depends a lot on the 
equipment". ;)


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Connecting branch offices through IPsec tunnel -- latency effects?

[Rich Adamson]

Stephen Bosch wrote:

Hi:

If I connect two offices through an IPsec tunnel, what is the impact on
latency, and does it noticeably affect calls?

Has anyone out there tried this? What were the effects?


Yes, I've done it through cisco's vpn, sonicwall vpn, and MS non-ipsec 
tunnels. Works fine "if" the underlying transport is reasonable. Latency 
is generally not an issue; jitter seems to be a hotter issue.



___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Still voice with echo

[Carlos Alberto Bernat Orozco]

First at all, thanks guys for the support!!I've been doing what people told me. To asure that I have DirectX on SJPhone (audio setting option enable DirectX 8.1) and Ican't run fxotune because I don't use this cards (sorry if I'm wrong). I'm just trying to probe my * box with the 
voip-info.orgtutorials. I turn down the mic gain and the pc's has the enough power. I make the echo test on the 2 clients (dialing 500 on *) and it soundsgreat (fluid voice).
This is my [general]sip.conf format: I omitted other parts which were on comments because are examples from the web site[general]context=default;allowguest=no      
;realm=mydomain.tld   bindport=5060bindaddr=0.0.0.0srvlookup=yes   ;domain=mydomain.tld;** Cambio de lineasdisallow=all;allow=g729allow=gsmallow=ulaw
jitterbuffer=yesmaxjitterbuffer=800;allow=ilbc;musicclass=default;language=en;relaxdtmf=yesrtptimeout=60     ;rtpholdtimeout=300;trustrpid = no;sendrpid = yes
;progressinband=never     ;useragent=Asterisk PBX;promiscredir = no        ;usereqphone = no  ;*** Cambio de lineas  DTMFMODE estaba en comentarios 
dtmfmode = rfc2833  ;compactheaders = yes   ;sipdebug = yes         ;subscribecontext = default         
    ;notifyringing = yes    ; Usuario 1 [usuario1]type=friend host=dynamic dtmfmode=rfc2833 username=usuario1
 secret=usuario1; Usuario 2 [usuario2]type=friend host=dynamic dtmfmode=rfc2833 username=usuario2 secret=usuario2And again thanks for the help!
Carlos Bernat
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] SIP and podcasts

[Matt Riddell (NZ)]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

kael wrote:
> I find curious there's no solution to listen to podcasts via SIP servers.

Um, I've just uploaded a total hack to the wiki:

http://www.voip-info.org/wiki/view/PodCast

You will need phpagi, the script, the extensions.conf entry, magpie rss
and a bit of patience.

Drop me a line in you have any problems!

BTW: It is a total hack, I was planning to clean it up, but never got
round to it, seeing as you're looking for it I thought I'd post it anyway.

It works for me, but YMMV!

:)

- --
Cheers,

Matt Riddell
___

http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://freevoip.gedameurope.com (Free Asterisk Voip Community)
http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.2 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFExq5vS6d5vy0jeVcRAvdOAJ9dbiEEp2MSj/SIZki29GSHCSAkaACfV0tZ
hNm2DsK9j1vrAxJ+wg2RYg4=
=IYze
-END PGP SIGNATURE-
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Sangoma Stops Receiving Calls

[Alex Robar]

Flynn,(First off, my mistake... It's an 8 port card with 6 active ports). I have not seen any type of pattern that would give me a hint as to what is going on here. We seemed to be working fine for a month, and then all of a sudden calls weren't coming in. I rebooted the system, calls came in fine for another few months. Then today, I saw the same behavior.
I will attach my .conf files to this message.As for CLI output... Unfortunately, there doesn't seem to be any. Nothing appears on the CLI when a call comes in at all when the system is in this state. 
Thanks for the help,AlexsOn 7/25/06, El Flynn <[EMAIL PROTECTED]> wrote:
Alex Robar wrote:> Hi all,>> I have a Sangoma A200 card with hardware echo cancellation. The card has 12> ports (10 of which are active; All FXO). Twice on this particular card I've> seen all ports simply stop receiving incoming calls. There is no other
> indication of this, however. I am able to place outgoing calls just fine,> and call other extensions without issue. When someone calls in, the line> just rings and rings, with no indication that the card even sees the calls.
> I'm not even sure where to begin looking into this. Could anyone give me> some pointers as to what I might need to be looking for?>Did this just happen, e.g. was your system working fine before? Does it happen
randomly, or have you seen any indication of a pattern of behavior?Perhaps if you could post your zaptel/zapata configs, and maybe some CLI outputwhen this happens, it would be easier for us to help you out.
We've got a client who's been using an A200 with 24 ports over the past 7months, without any problems like what you mentioned.Cheers,Flynn___
--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   
http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar[EMAIL PROTECTED]


zapata.conf
Description: Binary data


zaptel.conf
Description: Binary data
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Caller ID on Transfers

[C F]

WOW, another well documented mistake by Douglas Garstang.

On 7/25/06, Douglas Garstang <[EMAIL PROTECTED]> wrote:

> -Original Message-
> From: C F [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, July 25, 2006 4:46 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Caller ID on Transfers
>
>
> so you are telling me that all the time you have been bitching you
> have been doing an attended transfer.
> anyhow, once you hit transfer, another soft button shows up which says
> blind hit that and dial the number.

Oh Crud. Well, here's what I was doing wrong. I was under the impression that 
you always pressed 'Transfer', entered the new number, and then pressed send. 
At this point, if you pressed 'Transfer' again before the other party picked 
up, then it was an unattended transfer. If you waited until the other party 
picked up, and pressed 'Transfer' again afterwards, then it was an attended 
transfer.

I hadn't realised there was a 'Blind' button (probably called 'Blind Xfer' in 
older software versions). That seems to make a difference and is a step in the 
right direction (BLINDTRANSFER is now set), although I'm still not getting 
RDNIS set.

Doug.


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Caller ID on Transfers

[Watkins, Bradley]

Then you aren't doing a blind transfer in that case.  
You press 'Transfer', 'Blind', and then enter the number.
 
- Brad


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas 
GarstangSent: Tuesday, July 25, 2006 6:27 PMTo: Asterisk 
Users Mailing List - Non-Commercial DiscussionSubject: RE: 
[asterisk-users] Caller ID on Transfers

I 
don't understand how that's possible. When you press the 'transfer' button on 
the polycom, enter a number, and press send, the SIP messaging and setup at that 
point is exactly the same for both an attended and unattanded call. The Polycom 
doesn't give you the second transfer button, to release the call as unattended 
until AFTER the destination number has started to ring.
 
Doug.

  -Original Message-From: Watkins, Bradley 
  [mailto:[EMAIL PROTECTED]Sent: Tuesday, July 25, 2006 
  4:15 PMTo: Asterisk Users Mailing List - Non-Commercial 
  DiscussionSubject: RE: [asterisk-users] Caller ID on 
  Transfers
  I can say with absolute certainty that in our 
  installations using the blind transfer of the Polycom (NOT the Asterisk 
  transfers) will show the original caller ID and not the caller ID of the 
  transferer.  Attended transfers, of course, show the transferer since 
  that is a new call initially.
   
  Regards,
  - Brad
  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Douglas 
  GarstangSent: Tuesday, July 25, 2006 4:05 PMTo: Asterisk 
  Users Mailing List - Non-Commercial DiscussionSubject: RE: 
  [asterisk-users] Caller ID on Transfers
  
  Bruce, I bet your doing Asterisk assisted transfers (blindxfer and 
  atxfer in features.conf), and not using the 'transfer' soft or hard key on the 
  Polycom phones...
  
-Original Message-From: Bruce Reeves 
[mailto:[EMAIL PROTECTED]Sent: Tuesday, July 25, 2006 
1:53 PMTo: Asterisk Users Mailing List - Non-Commercial 
DiscussionSubject: Re: [asterisk-users] Caller ID on 
Transfersthat's odd, our Polycom phones show the 
original caller id on blind transfers but the callerid of the person doing 
the attended transfer, in our case a receptionist. 
On 7/25/06, Doug 
Lytle <[EMAIL PROTECTED]> 
wrote: 
Douglas 
  Garstang wrote:>> talking to).  Blind transfers show 
  the original caller. Doesn't seem to be 
  happening that way with Polycom phones and blind/attended 
  transfers.> Both are showing the original calling party caller id. 
  > ___>I 
  can confirm this.Both Attended and Blind transfers show the same 
  Caller ID (That of theperson doing the 
  transfer).DOug-- Ben Franklin quote:"Those 
  who would give up Essential Liberty to purchase a little Temporary Safety, 
  deserve neither Liberty nor 
  Safety."___ 
  --Bandwidth and Colocation provided by Easynews.com --asterisk-users 
  mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users-- BruceNortex Networks The contents of 
  this e-mail are intended for the named addressee only. It contains information 
  that may be confidential. Unless you are the named addressee or an authorized 
  designee, you may not copy or use it, or disclose it to anyone else. If you 
  received it in error please notify us immediately and then destroy it. 
The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. 
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Caller ID on Transfers

[Douglas Garstang]

> -Original Message-
> From: C F [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, July 25, 2006 4:46 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Caller ID on Transfers
> 
> 
> so you are telling me that all the time you have been bitching you
> have been doing an attended transfer.
> anyhow, once you hit transfer, another soft button shows up which says
> blind hit that and dial the number.

Oh Crud. Well, here's what I was doing wrong. I was under the impression that 
you always pressed 'Transfer', entered the new number, and then pressed send. 
At this point, if you pressed 'Transfer' again before the other party picked 
up, then it was an unattended transfer. If you waited until the other party 
picked up, and pressed 'Transfer' again afterwards, then it was an attended 
transfer.

I hadn't realised there was a 'Blind' button (probably called 'Blind Xfer' in 
older software versions). That seems to make a difference and is a step in the 
right direction (BLINDTRANSFER is now set), although I'm still not getting 
RDNIS set.

Doug.


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Caller ID on Transfers

[Douglas Garstang]

> -Original Message-
> From: C F [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, July 25, 2006 4:34 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Caller ID on Transfers
> 
> 
> what version of asterisk you using? I use the ${BLINDTRANSFER} a lot
> in my dialplans (and I know they work otherwise the users would have
> complained) and it works for me using the polycom blind xfer button
> (the one that shows up only after hitting the transfer button) runing
> polycom sip version 1.5.x

Asterisk version 1.2.9.1 and Polycom SIP Software 1.6.3.
Does it actually say 'Blind xfer' the second time, or 'Transfer'? Mine says 
'Transfer'.

Still don't see how it can work if it looks like an attended transfer to 
Asterisk when hit that Transfer button the first time.

Doug.
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Caller ID on Transfers

[asterisk]

What you describe is an attended transfer.  To do blind transfer on
Polycom, you hit TRANSFER button, then the BLIND _soft_ button, then the
extension.  
My Polycom's show original CID info when doing blind transfers (but not
attended txfr's).

At 05:27 PM 7/25/2006, you wrote:
content-class:
urn:content-classes:message
Content-Type: multipart/alternative;

boundary="_=_NextPart_001_01C6B039.789D6078"
I don't understand
how that's possible. When you press the 'transfer' button on the polycom,
enter a number, and press send, the SIP messaging and setup at that point
is exactly the same for both an attended and unattanded call. The Polycom
doesn't give you the second transfer button, to release the call as
unattended until AFTER the destination number has started to ring.
 
Doug.


-Original Message-

From: Watkins, Bradley
[
mailto:[EMAIL PROTECTED]]

Sent: Tuesday, July 25, 2006 4:15 PM

To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: RE: [asterisk-users] Caller ID on Transfers

I can say with
absolute certainty that in our installations using the blind transfer of
the Polycom (NOT the Asterisk transfers) will show the original caller ID
and not the caller ID of the transferer.  Attended transfers, of
course, show the transferer since that is a new call initially.

 

Regards,

- Brad



From: [EMAIL PROTECTED]
[
mailto:[EMAIL PROTECTED]] On Behalf Of
Douglas Garstang

Sent: Tuesday, July 25, 2006 4:05 PM

To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: RE: [asterisk-users] Caller ID on Transfers


Bruce, I bet
your doing Asterisk assisted transfers (blindxfer and atxfer in
features.conf), and not using the 'transfer' soft or hard key on the
Polycom phones...


-Original Message-

From: Bruce Reeves
[
mailto:[EMAIL PROTECTED]]

Sent: Tuesday, July 25, 2006 1:53 PM

To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: Re: [asterisk-users] Caller ID on Transfers

that's odd, our Polycom phones show the original caller id on blind
transfers but the callerid of the person doing the attended transfer, in
our case a receptionist. 

On 7/25/06, Doug Lytle
<[EMAIL PROTECTED]> wrote:



Douglas Garstang wrote:

>> talking to).  Blind transfers show the original
caller.

>>

>

> Doesn't seem to be happening that way with Polycom phones and
blind/attended transfers.

> Both are showing the original calling party caller id. 

> ___

>

I can confirm this.

Both Attended and Blind transfers show the same Caller ID (That of
the

person doing the transfer).

DOug

-- 

Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little
Temporary Safety, deserve neither Liberty nor Safety."


___ 

--Bandwidth and Colocation provided by
Easynews.com --

asterisk-users mailing list

To UNSUBSCRIBE or update options visit:

  

http://lists.digium.com/mailman/listinfo/asterisk-users




-- 

Bruce

Nortex Networks 


The contents of this e-mail are intended for the named addressee
only. It contains information that may be confidential. Unless you are
the named addressee or an authorized designee, you may not copy or use
it, or disclose it to anyone else. If you received it in error please
notify us immediately and then destroy it. 


___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  

http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Caller ID on Transfers

[C F]

so you are telling me that all the time you have been bitching you
have been doing an attended transfer.
anyhow, once you hit transfer, another soft button shows up which says
blind hit that and dial the number.

On 7/25/06, Douglas Garstang <[EMAIL PROTECTED]> wrote:



I don't understand how that's possible. When you press the 'transfer' button
on the polycom, enter a number, and press send, the SIP messaging and setup
at that point is exactly the same for both an attended and unattanded call.
The Polycom doesn't give you the second transfer button, to release the call
as unattended until AFTER the destination number has started to ring.

Doug.

-Original Message-
From: Watkins, Bradley [mailto:[EMAIL PROTECTED]
Sent: Tuesday, July 25, 2006 4:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Caller ID on Transfers


I can say with absolute certainty that in our installations using the blind
transfer of the Polycom (NOT the Asterisk transfers) will show the original
caller ID and not the caller ID of the transferer.  Attended transfers, of
course, show the transferer since that is a new call initially.

Regards,
- Brad

 
 From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Of Douglas Garstang
Sent: Tuesday, July 25, 2006 4:05 PM

To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Caller ID on Transfers



Bruce, I bet your doing Asterisk assisted transfers (blindxfer and atxfer in
features.conf), and not using the 'transfer' soft or hard key on the Polycom
phones...

-Original Message-
From: Bruce Reeves [mailto:[EMAIL PROTECTED]
Sent: Tuesday, July 25, 2006 1:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Caller ID on Transfers

that's odd, our Polycom phones show the original caller id on blind
transfers but the callerid of the person doing the attended transfer, in our
case a receptionist.


On 7/25/06, Doug Lytle <[EMAIL PROTECTED]> wrote:
> Douglas Garstang wrote:
> >> talking to).  Blind transfers show the original caller.
> >>
> >
> > Doesn't seem to be happening that way with Polycom phones and
blind/attended transfers.
> > Both are showing the original calling party caller id.
> > ___
> >
>
> I can confirm this.
>
> Both Attended and Blind transfers show the same Caller ID (That of the
> person doing the transfer).
>
> DOug
>
> --
>
> Ben Franklin quote:
>
> "Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety."
>
>
> ___

--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:

http://lists.digium.com/mailman/listinfo/asterisk-users



--
Bruce
Nortex Networks The contents of this e-mail are intended for the named
addressee only. It contains information that may be confidential. Unless you
are the named addressee or an authorized designee, you may not copy or use
it, or disclose it to anyone else. If you received it in error please notify
us immediately and then destroy it.
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:

http://lists.digium.com/mailman/listinfo/asterisk-users




___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Caller ID on Transfers

[Douglas Garstang]

> -Original Message-
> From: Anthony Rodgers [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, July 25, 2006 4:21 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Caller ID on Transfers
> 
> 
> The 'o' option to the Dial() command, along with using blind 
> transfers, 
> fixed this problem for us.

Well, for a moment there I thought we had it. Although, as I said earlier, this 
should never work. When you hit transfer, enter a new number and press send, 
the call setup is attended at that point. The caller id has already been sent 
to the new phone before you get the chance to hit the transfer button a second 
time and release the call, or make it unattended.

Anyway, I put 'o' in the dial string, but it made no difference.

exten => 2944093,1,Dial(SIP/2944093,20,tro)
exten => 3254103,1,Dial(SIP/3254103,20,tro)
exten => 9220371,2,Dial(SIP/9220371,20,tro)

Can someone post a relevant sip.conf section so I can compare? Here's mine for 
3254103...

[3254103]
type = friend
context = pbx_betty_start
username = 3254103
secret = foo
accountcode = 3254103
qualify = no
canreinvite = no
host = dynamic
callgroup = 1
pickupgroup = 1
dtmfmode = rfc2833
;nat = no
mailbox = [EMAIL PROTECTED]
allow = g729

Doug.
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Caller ID on Transfers

[C F]

what version of asterisk you using? I use the ${BLINDTRANSFER} a lot
in my dialplans (and I know they work otherwise the users would have
complained) and it works for me using the polycom blind xfer button
(the one that shows up only after hitting the transfer button) runing
polycom sip version 1.5.x

On 7/25/06, Douglas Garstang <[EMAIL PROTECTED]> wrote:

> -Original Message-
> From: C F [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, July 25, 2006 4:06 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Caller ID on Transfers
>
>
> In my experience (although I didn't test this as I type now) using the
> Transfer button on the Polycom if you do a blind will show the
> original CID, and doing an attended will show the transferees CID.
> RDNIS should not come up, but ${BLINDTRANSFER} on a blindtransfer
> should. On a non blind transfer the ${CDR(channel)} or
> ${CDR(dstchannel)} should hold the right channel info. Doing some
> logic on the CDR(var) helps to figure out what happened with the call.
> If the src/dst don't match the channel/dstchannel, then you know a
> xfer occured, also if there are 2 records then you are dealing with an
> attd xfer, while a blind xfer will just have one record that the
> dst/src and dstchannel/channel don't match.
> Hope this help.

That's not the results I am getting. My dial plan has:

exten => 2944093,1,Dial(SIP/2944093,20,tr)
exten => 3254103,1,Dial(SIP/3254103,20,tr)
exten => 9220371,1,Dial(SIP/9220371,20,tr)

When 2944093 dials 3254103, and 3254103 transfers to 9220371, with the Polycom 
transfer soft key, both attended and unattended, the display of 9220371 shows 
what's in sip.conf's caller id for 3254103, not 2944093.

When 2944093 dials 32534103, and 3254103 transfers to 9220371 with an asterisk 
assisted blind transfer(blindxfer), the display of 9220371 shows what's in 
sip.conf for 2944093. When 2944093 dials 32534103, and 3254103 transfers to 
9220371 with an asterisk assisted attended transfer(atxfer), the display of 
9220371 shows what's in sip.conf for 3254103.

I changed my dial plan to:

exten => 2944093,1,Dial(SIP/2944093,20,tr)
exten => 3254103,1,Dial(SIP/3254103,20,tr)
exten => 9220371,1,NoOp(${BLINDTRANSFER})
exten => 9220371,2,Dial(SIP/9220371,20,tr)

When I do an asterisk assisted blind transfer, the BLINDTRANSFER field is 
populated. However, when I do a Polycom blind transfer (hit transfer soft key, 
enter dest num, hit send), the BLINDTRANSFER field is not set. Part of the 
problem may be that 3254103 STILL HAS CONTROL of the call. I have not pressed 
transfer a second time yet to release the call, and Asterisk still think that 
it is attended at this point.

Any way, it's really screwy, because it means that if you use Polycom phones, 
and you use the soft transfer key, your _always_ going to get the caller id of 
the transferer, not the original party.

Douglas.
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Sangoma Stops Receiving Calls

[El Flynn]

Alex Robar wrote:

Hi all,

I have a Sangoma A200 card with hardware echo cancellation. The card has 12
ports (10 of which are active; All FXO). Twice on this particular card I've
seen all ports simply stop receiving incoming calls. There is no other
indication of this, however. I am able to place outgoing calls just fine,
and call other extensions without issue. When someone calls in, the line
just rings and rings, with no indication that the card even sees the calls.
I'm not even sure where to begin looking into this. Could anyone give me
some pointers as to what I might need to be looking for?



Did this just happen, e.g. was your system working fine before? Does it happen 
randomly, or have you seen any indication of a pattern of behavior?


Perhaps if you could post your zaptel/zapata configs, and maybe some CLI output 
when this happens, it would be easier for us to help you out.


We've got a client who's been using an A200 with 24 ports over the past 7 
months, without any problems like what you mentioned.


Cheers,
Flynn


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Caller ID on Transfers

[Douglas Garstang]

I 
don't understand how that's possible. When you press the 'transfer' button on 
the polycom, enter a number, and press send, the SIP messaging and setup at that 
point is exactly the same for both an attended and unattanded call. The Polycom 
doesn't give you the second transfer button, to release the call as unattended 
until AFTER the destination number has started to ring.
 
Doug.

  -Original Message-From: Watkins, Bradley 
  [mailto:[EMAIL PROTECTED]Sent: Tuesday, July 25, 2006 
  4:15 PMTo: Asterisk Users Mailing List - Non-Commercial 
  DiscussionSubject: RE: [asterisk-users] Caller ID on 
  Transfers
  I can say with absolute certainty that in our 
  installations using the blind transfer of the Polycom (NOT the Asterisk 
  transfers) will show the original caller ID and not the caller ID of the 
  transferer.  Attended transfers, of course, show the transferer since 
  that is a new call initially.
   
  Regards,
  - Brad
  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Douglas 
  GarstangSent: Tuesday, July 25, 2006 4:05 PMTo: Asterisk 
  Users Mailing List - Non-Commercial DiscussionSubject: RE: 
  [asterisk-users] Caller ID on Transfers
  
  Bruce, I bet your doing Asterisk assisted transfers (blindxfer and 
  atxfer in features.conf), and not using the 'transfer' soft or hard key on the 
  Polycom phones...
  
-Original Message-From: Bruce Reeves 
[mailto:[EMAIL PROTECTED]Sent: Tuesday, July 25, 2006 
1:53 PMTo: Asterisk Users Mailing List - Non-Commercial 
DiscussionSubject: Re: [asterisk-users] Caller ID on 
Transfersthat's odd, our Polycom phones show the 
original caller id on blind transfers but the callerid of the person doing 
the attended transfer, in our case a receptionist. 
On 7/25/06, Doug 
Lytle <[EMAIL PROTECTED]> 
wrote: 
Douglas 
  Garstang wrote:>> talking to).  Blind transfers show 
  the original caller. Doesn't seem to be 
  happening that way with Polycom phones and blind/attended 
  transfers.> Both are showing the original calling party caller id. 
  > ___>I 
  can confirm this.Both Attended and Blind transfers show the same 
  Caller ID (That of theperson doing the 
  transfer).DOug-- Ben Franklin quote:"Those 
  who would give up Essential Liberty to purchase a little Temporary Safety, 
  deserve neither Liberty nor 
  Safety."___ 
  --Bandwidth and Colocation provided by Easynews.com --asterisk-users 
  mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users-- BruceNortex Networks The contents of 
  this e-mail are intended for the named addressee only. It contains information 
  that may be confidential. Unless you are the named addressee or an authorized 
  designee, you may not copy or use it, or disclose it to anyone else. If you 
  received it in error please notify us immediately and then destroy it. 

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] sounds format

[Mauricio Mantilla]

Hi all, I'm doing an IVR, and I need to record some sounds and use some others that have already been recorded.I've seen several sound formats like gsm, g729, slin, and even wav.What is the best sound format to use for playback in asterisk??
Thanks,Mauricio Mantilla
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] How to send a signal via E1/T1 ISDN to asterisk, to ask the call to be moved.

[Manrique Feoli]

Hi, all
I have an * which receives calls from PSTN and some of them fo to an E1 
where another system is working   (Dialogic Boards).


I need to be able to send a signal to * from the system with the 
Dialogic boards,  preferrably via the E1 so that * knows it has to move 
the call from slot ZAP25 to  SIP/ xxx .


Im thinking to use the manager with a socket connection for this,  but 
would be much cleaner for me if I can send a message via the E1,


has anyone done something similar?
please welcome any ideas for this


PS
(if this sounds familiar,  it's because I'm trying to go arround the 
2b-channel limitation that we discussed earlier on,  where I couldn't 
find a way to tell )


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Caller ID on Transfers

[Douglas Garstang]

> -Original Message-
> From: C F [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, July 25, 2006 4:06 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Caller ID on Transfers
> 
> 
> In my experience (although I didn't test this as I type now) using the
> Transfer button on the Polycom if you do a blind will show the
> original CID, and doing an attended will show the transferees CID.
> RDNIS should not come up, but ${BLINDTRANSFER} on a blindtransfer
> should. On a non blind transfer the ${CDR(channel)} or
> ${CDR(dstchannel)} should hold the right channel info. Doing some
> logic on the CDR(var) helps to figure out what happened with the call.
> If the src/dst don't match the channel/dstchannel, then you know a
> xfer occured, also if there are 2 records then you are dealing with an
> attd xfer, while a blind xfer will just have one record that the
> dst/src and dstchannel/channel don't match.
> Hope this help.

That's not the results I am getting. My dial plan has:

exten => 2944093,1,Dial(SIP/2944093,20,tr)
exten => 3254103,1,Dial(SIP/3254103,20,tr)
exten => 9220371,1,Dial(SIP/9220371,20,tr)

When 2944093 dials 3254103, and 3254103 transfers to 9220371, with the Polycom 
transfer soft key, both attended and unattended, the display of 9220371 shows 
what's in sip.conf's caller id for 3254103, not 2944093.

When 2944093 dials 32534103, and 3254103 transfers to 9220371 with an asterisk 
assisted blind transfer(blindxfer), the display of 9220371 shows what's in 
sip.conf for 2944093. When 2944093 dials 32534103, and 3254103 transfers to 
9220371 with an asterisk assisted attended transfer(atxfer), the display of 
9220371 shows what's in sip.conf for 3254103.

I changed my dial plan to:

exten => 2944093,1,Dial(SIP/2944093,20,tr)
exten => 3254103,1,Dial(SIP/3254103,20,tr)
exten => 9220371,1,NoOp(${BLINDTRANSFER})
exten => 9220371,2,Dial(SIP/9220371,20,tr)

When I do an asterisk assisted blind transfer, the BLINDTRANSFER field is 
populated. However, when I do a Polycom blind transfer (hit transfer soft key, 
enter dest num, hit send), the BLINDTRANSFER field is not set. Part of the 
problem may be that 3254103 STILL HAS CONTROL of the call. I have not pressed 
transfer a second time yet to release the call, and Asterisk still think that 
it is attended at this point. 

Any way, it's really screwy, because it means that if you use Polycom phones, 
and you use the soft transfer key, your _always_ going to get the caller id of 
the transferer, not the original party.

Douglas.
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Caller ID on Transfers

[Anthony Rodgers]

The 'o' option to the Dial() command, along with using blind transfers, 
fixed this problem for us.


A.

On Jul 25, 2006, at 11:25 AM, Douglas Garstang wrote:


I have three phones here with extensions 2944093, 3254103 and 9220371.
 
2944093 calls 3254103. 3254103 transfers 2944093 to 9220371. We want 
the caller id of 2944093 to be presented on the display of 9220371.
However, the caller id of the transferer, 3254103, is appearing. This 
doesn't make any sense.

 
How can we do this?
 
Doug.
 ___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Caller ID on Transfers

[Mojo with Horan & Company, LLC]

This behavior Bruce mentioned is confirmed with our installation 
(krisk.org cfg files)


Bruce Reeves wrote:
that's odd, our Polycom phones show the original caller id on blind 
transfers but the callerid of the person doing the attended transfer, in 
our case a receptionist.


On 7/25/06, * Doug Lytle* <[EMAIL PROTECTED] 
> wrote:


Douglas Garstang wrote:
 >> talking to).  Blind transfers show the original caller.
 >>
 >
 > Doesn't seem to be happening that way with Polycom phones and
blind/attended transfers.
 > Both are showing the original calling party caller id.
 > ___
 >

I can confirm this.

Both Attended and Blind transfers show the same Caller ID (That of the
person doing the transfer).

DOug

-- 


Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little
Temporary Safety, deserve neither Liberty nor Safety."


___
--Bandwidth and Colocation provided by Easynews.com
 --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users




--
Bruce
Nortex Networks !DSPAM:500,44c6780f22891882367086!




___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


!DSPAM:500,44c6780f22891882367086!


--
Mojo <[EMAIL PROTECTED]>
Office Manger, Horan & Company, LLC
(907) 747- x112
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Caller ID on Transfers

[Watkins, Bradley]

I can say with absolute certainty that in our installations 
using the blind transfer of the Polycom (NOT the Asterisk transfers) will show 
the original caller ID and not the caller ID of the transferer.  Attended 
transfers, of course, show the transferer since that is a new call 
initially.
 
Regards,
- Brad


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas 
GarstangSent: Tuesday, July 25, 2006 4:05 PMTo: Asterisk 
Users Mailing List - Non-Commercial DiscussionSubject: RE: 
[asterisk-users] Caller ID on Transfers

Bruce, 
I bet your doing Asterisk assisted transfers (blindxfer and atxfer in 
features.conf), and not using the 'transfer' soft or hard key on the Polycom 
phones...

  -Original Message-From: Bruce Reeves 
  [mailto:[EMAIL PROTECTED]Sent: Tuesday, July 25, 2006 
  1:53 PMTo: Asterisk Users Mailing List - Non-Commercial 
  DiscussionSubject: Re: [asterisk-users] Caller ID on 
  Transfersthat's odd, our Polycom phones show the original 
  caller id on blind transfers but the callerid of the person doing the attended 
  transfer, in our case a receptionist. 
  On 7/25/06, Doug 
  Lytle <[EMAIL PROTECTED]> 
  wrote: 
  Douglas 
Garstang wrote:>> talking to).  Blind transfers show the 
original caller. Doesn't seem to be happening 
that way with Polycom phones and blind/attended transfers.> Both are 
showing the original calling party caller id. > 
___>I can confirm 
this.Both Attended and Blind transfers show the same Caller ID (That 
of theperson doing the transfer).DOug-- Ben 
Franklin quote:"Those who would give up Essential Liberty to 
purchase a little Temporary Safety, deserve neither Liberty nor 
Safety."___ 
--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing 
listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users-- BruceNortex Networks The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. 
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Caller ID on Transfers

[C F]

In my experience (although I didn't test this as I type now) using the
Transfer button on the Polycom if you do a blind will show the
original CID, and doing an attended will show the transferees CID.
RDNIS should not come up, but ${BLINDTRANSFER} on a blindtransfer
should. On a non blind transfer the ${CDR(channel)} or
${CDR(dstchannel)} should hold the right channel info. Doing some
logic on the CDR(var) helps to figure out what happened with the call.
If the src/dst don't match the channel/dstchannel, then you know a
xfer occured, also if there are 2 records then you are dealing with an
attd xfer, while a blind xfer will just have one record that the
dst/src and dstchannel/channel don't match.
Hope this help.

On 7/25/06, Douglas Garstang <[EMAIL PROTECTED]> wrote:



I have three phones here with extensions 2944093, 3254103 and 9220371.

2944093 calls 3254103. 3254103 transfers 2944093 to 9220371. We want the
caller id of 2944093 to be presented on the display of 9220371.
However, the caller id of the transferer, 3254103, is appearing. This
doesn't make any sense.

How can we do this?

Doug.

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:

http://lists.digium.com/mailman/listinfo/asterisk-users




___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Voicemail Forwarding

[C F]

Tell them to create a temp message.

On 7/25/06, Forrest Beck <[EMAIL PROTECTED]> wrote:

Currently I am using this in my dial plan to forward calls to
voicemail when the phone isn't picked up.

exten => 2503,1,Dial(SIP/2503,20)
exten => 2503,2,Voicemail([EMAIL PROTECTED])

Some users prefer that just there name gets read at the prompt instead
of the unavailable message.

So my issue is.  The user creates a unavail message and decides they
want to use there name instead.  The only way I can find to switch it
back to name is to delete unavail.gsm from their voicemail box.

There should be a prompt in the voicemail admin that allows the user
to select what message they want to use (or just a name), and be able
to switch back and forth.

Am I missing something or is there no way to do this?

Thanks.
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Rookie voicemail user question

[William Piper]

On 7/25/06, Randy Paries <[EMAIL PROTECTED]> wrote:

Hello,I just got my Asterisk up and running, and everything is greatWhat i can not seem to find is a doc that describes any of the user commands
Like is there things like, end message or listen to the message i amleaving , or anything like that?ThanksRandy
Google is your friend, learn to use it:
http://www.google.com/search?hl=en&q=asterisk+voicemail+menu
 
bp
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] sip realtime

[William Piper]

You'll need to create a php or perl script to load your sip table into a txt file before doing a reload. You can create your own or just steal the code from AMP and modify it a tad. I believe the file name is retrieve_sip_conf_from_mysql.pl

 
bp
 
On 7/25/06, marek cervenka <[EMAIL PROTECTED]> wrote:
hi,i'm reading a lot docs about asterisk realtimebut i cannot understand how works sip realtime static
i need NAT/qualify for SIP. this is not possible with dynamic realtimei want- save data to sql- asterisk -rx "reload" to read config (sip.conf with sip users) from sqlit is possible?
can you point me to some examples?thanks---Marek Cervenka===___--Bandwidth and Colocation provided by 
Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] PRI vs "Digital Trunk"

[Alexander Lopez]

Cheaper depends on the age of the switch
at the CO you are connecting to. In some parts of Tenn.. I cannot get a PRI but a T1 will be
just fine, alas NO CID, or enhanced features but a T1 nonetheless.

 

I my old CO PRIs had to be FXed (Brought
in from another CO) My CO was a 1A and but not have the SW and/or ability to  provide
ISDN services as the other 5E did. This was a problem as the contract stated
that once ISDN services were available in my CO we would have to change over to
that exchange. We later moved into the area served by the 5E and the change did
not have to happen.

 

 






___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] PRI died and Asterisk crashed

[Steve Kennedy]

The telco I used had a fault with the switch and the PRI (E1) went down.

This seems to have caused Asterisk (1.2.10) to crash. Latest zaptel and
libpri.

Steve

-- 
NetTek Ltd  UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PROTECTED]
Euro Tech News Blog http://eurotechnews.blogspot.com
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Rookie voicemail user question

[Randy Paries]

Hello,
I just got my Asterisk up and running, and everything is great
What i can not seem to find is a doc that describes any of the user commands

Like is there things like, end message or listen to the message i am
leaving , or anything like that?

Thanks
Randy
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] vegastream 50 FXO DTMF Problem

[Issac Simchayof]

Pete,

My config is identical to yours. I think I might have an issue with my Vega.




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter Doyle
Sent: Tuesday, July 25, 2006 4:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] vegastream 50 FXO DTMF Problem

Hi Issac,
I would double-check the dtmfmode=inband thing, just to be 100% sure.
We've never been able to get out of band dtmf working with the vega and
asterisk, so we just use inband and it works well enough.

With our setup, the vega doesn't seem to strip out outgoing inband dtmf
tones, so if asterisk sends them inband, it seems like it should make it
to the PSTN.  That's why I think it'd be valuable to just verify
asterisk was actually sending inband dtmf.  I'm assuming the link
between asterisk and the vega is using an uncompressed codec like
g711a/g711u, otherwise maybe codec compression could be distorting the
dtmf tones, making them unrecognizable to the equipment at the far-end.

Here's the appropriate sections from our sip.conf:
[vega]
type=user
dtmfmode=inband
disallow=all
context=from-pstn
allow=ulaw

[vega-gw]
type=peer
host={ip address here}
dtmfmode=inband
disallow=all
context=from-internal
allow=ulaw

Then in the vega web setup, under media, we set the out of band DTMF to
"0" (off) for g711ulaw (which is all we use internally, and asterisk
translates for other codecs).  

Hope that helps.  It atleast was the solution for us
Thanks
Pete

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Issac
Simchayof
Sent: Tuesday, July 25, 2006 10:43 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] vegastream 50 FXO DTMF Problem

Thanks Pete,

I did try dtmfmode=inband and it did not work for us.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter
Doyle
Sent: Tuesday, July 25, 2006 12:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] vegastream 50 FXO DTMF Problem

Hi Issac,
If I recall correctly, out of band DTMF didn't seem to work for us on
our Vega 50 (atleast not when using the Vega with Asterisk).  We had to
tell Asterisk to use dtmfmode=inband in our sip.conf.  It didn't seem
like we had to change any settings on the Vega, because it was sending
both inband and RFC2833 when RFC2833 was selected (for incoming DTMF).
This is all from memory, so hopefully its correct.

I'm headed to work in a few minutes here, where maybe I can see our
actual config and send it along.
Good Luck!
Pete

P.S.  How has your experience been with the echo cancelers?  Are you
experiencing much echo at all?  That's the one thing we're still
struggling with. Thanks


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Issac
Simchayof
Sent: Tuesday, July 25, 2006 8:48 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] vegastream 50 FXO DTMF Problem

Asterisk is sending the DTMF as we can see in ethereal but the Vega is
not sending them out. We did try the debug before ethereal but the tech
at VegaStream insisted we will need ethereal to troubleshoot this.  



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry
Jones
Sent: Tuesday, July 25, 2006 11:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] vegastream 50 FXO DTMF Problem

How do you mean it does not recognize them? By the routing not working
properly?

Or by not outdialing properly?

No need for ethereal, just turn on sip debugging and it will display the
messages for you, just like * will.


On Jul 25, 2006, at 9:43 AM, Issac Simchayof wrote:

>
> I am having a problem with my VegaStream 50 10 FXO. The unit does not 
> recognize DTMF sent from asterisk on outgoing calls. I have been 
> trying to resolve this with VegaStream support but they have not been 
> very helpful so far. On the last test we ran we used Eathereal to 
> capture traffic and it does look like asterisk does send the DTMF 
> codes to the Vega, but the Vega does not recognize them.
>
> Any Ideas from a VegaStream users?
>
>
>
>
>
> ___
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-u

[asterisk-users] Sangoma Stops Receiving Calls

[Alex Robar]

Hi all,I have a Sangoma A200 card with hardware echo cancellation. The card has 12 ports (10 of which are active; All FXO). Twice on this particular card I've seen all ports simply stop receiving incoming calls. There is no other indication of this, however. I am able to place outgoing calls just fine, and call other extensions without issue. When someone calls in, the line just rings and rings, with no indication that the card even sees the calls. I'm not even sure where to begin looking into this. Could anyone give me some pointers as to what I might need to be looking for?
I'll be giving Sangoma tech support a call, but if anyone has any debugging pointers, they would be much appreciated.Thanks,Alex-- Alex Robar
[EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] ACD Queues Agents logout

[Anthony Rodgers]

Hi Kai,

This is what we do:

[agent-login]
exten => s,1,NoOp(${AgentUser})
exten => 
s,2,AddQueueMember(${AgentContext}|${AgentChannel}|${AgentPenalty})

exten => s,3,Wait(1)
exten => s,4,Playback(agent-loginok)
exten => s,5,Hangup
exten => s,103,RemoveQueueMember(${AgentContext}|${AgentChannel})
exten => s,104,Wait(1)
exten => s,105,Playback(agent-loggedoff)
exten => s,106,Hangup

A.

On Jul 20, 2006, at 6:26 AM, Kai Ober wrote:


Okay, I think i have missed something:

When i use AgentCallbackLogin*(||*007)  the agent is logged in, fine.

But  how do i log OUT.
okay there is a timout,
autologoff=time

but how can an agent explicit log off?



regards

Kai
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] sip realtime

[marek cervenka]

hi,

i'm reading a lot docs about asterisk realtime
but i cannot understand how works sip realtime static

i need NAT/qualify for SIP. this is not possible with dynamic realtime
i want
- save data to sql
- asterisk -rx "reload" to read config (sip.conf with sip users) from sql

it is possible?
can you point me to some examples?
thanks

---
Marek Cervenka
===

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Voicemail Forwarding

[Forrest Beck]

Currently I am using this in my dial plan to forward calls to
voicemail when the phone isn't picked up.

exten => 2503,1,Dial(SIP/2503,20)
exten => 2503,2,Voicemail([EMAIL PROTECTED])

Some users prefer that just there name gets read at the prompt instead
of the unavailable message.

So my issue is.  The user creates a unavail message and decides they
want to use there name instead.  The only way I can find to switch it
back to name is to delete unavail.gsm from their voicemail box.

There should be a prompt in the voicemail admin that allows the user
to select what message they want to use (or just a name), and be able
to switch back and forth.

Am I missing something or is there no way to do this?

Thanks.
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Caller ID on Transfers

[Doug Lytle]

Douglas Garstang wrote:

Doug,

Thanks, but this isn't the same scenario. Can you try transferring from one 
Polycom to another Polycom?
  


Not until the end of the week.  I have an order for more phones.

Doug

--

Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety."


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Caller ID on Transfers

[Douglas Garstang]

> -Original Message-
> From: Joshua Colp [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, July 25, 2006 10:31 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] Caller ID on Transfers
> 
> 
> - Original Message -
> From: Douglas Garstang
> [mailto:[EMAIL PROTECTED]
> To: Asterisk Users Mailing List -
> Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
> Sent:
> Tue, 25 Jul 2006 17:03:00 -0300
> Subject: RE: [asterisk-users] Caller ID on
> Transfers
> 
> > 
> > If the new invite looks like a regular call, how can an AGI 
> script tell that
> > it's a transferred call?
> > 
> > Doug.
> 
> I know of no way for it to.

Well that just sucks.
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Caller ID on Transfers

[Douglas Garstang]

> -Original Message-
> From: Doug Lytle [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, July 25, 2006 2:27 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Caller ID on Transfers
> 
> 
> Douglas Garstang wrote:
> >> DOug
> >> 
> >
> > Hi Doug. Your doing your transfers with the transfer keys 
> on the Polycom, right? So am I. I think a distinction needs 
> to be made here. I get the impression that most people, and 
> certainly the others are using #1 and #2 to do Asterisk 
> assisted transfers. 
> >
> >   
> I've tried it both ways, same results.
> 
> I don't know if me calling from our Definity to the Polycom, 
> putting the 
> call on hold and transfering from the Polycom to my line 2 on the 
> Definity pollutes the results or not.

Doug,

Thanks, but this isn't the same scenario. Can you try transferring from one 
Polycom to another Polycom?

Doug.
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Caller ID on Transfers

[Doug Lytle]

Douglas Garstang wrote:

DOug



Hi Doug. Your doing your transfers with the transfer keys on the Polycom, right? So am I. I think a distinction needs to be made here. I get the impression that most people, and certainly the others are using #1 and #2 to do Asterisk assisted transfers. 

  

I've tried it both ways, same results.

I don't know if me calling from our Definity to the Polycom, putting the 
call on hold and transfering from the Polycom to my line 2 on the 
Definity pollutes the results or not.


Doug



--

Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety."


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Caller ID on Transfers

[Joshua Colp]

- Original Message -
From: Douglas Garstang
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Tue, 25 Jul 2006 17:03:00 -0300
Subject: RE: [asterisk-users] Caller ID on
Transfers

> 
> If the new invite looks like a regular call, how can an AGI script tell that
> it's a transferred call?
> 
> Doug.

I know of no way for it to.

Joshua Colp
Digium
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] PRI vs "Digital Trunk"

[Joe Pukepail]

I don't know about what our LEC is calling a "digital trunk" but verizon tried to offer me something like this for a location that they couldn't offer a PRI, basically it was a just a voice T1 (24 channels), didn't have features like Caller ID, setting outbound caller ID, ANI, etc.  YMMV. 

On 7/25/06, Barry D. Hassler <[EMAIL PROTECTED]> wrote:
Hi, can someone enlighten me as to the difference between a PRI and a"Digital Trunk" (other than cost)?
I do understand PRI (B-channel signaling, incoming/outgoing call setup,D channel for voice/data, etc), but I'm not quite sure how that compareswith what my vendor is calling a "Digital Trunk" (specifically in
contrast to a PRI). The PRI is about twice the cost.If this is just a channelized T1 (24 64k voice/data "channels'), wouldthey each be assigned a specific phone number, or is there furtherflexibility in sending/receiving calls, callerid (receive or send), etc?
Feeling ignorant hereThanks!___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list
To UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] PRI vs "Digital Trunk"

[Michael Collins]

> Hi, can someone enlighten me as to the difference between a PRI and a
> "Digital Trunk" (other than cost)?

Barry,

A "digital trunk" from the telco is most likely as you said - 24
channels of 64k voice.  (I know, technically it's "data" but the "data"
is just digitized voice.)  Think of the digital trunk as the equivalent
of 24 analog phone lines.  They can all be in hunt, some in hunt or
none.  (Depends on the carrier and your specific needs.) They can be
inbound, outbound or both.  (Again, depends on your needs.)  On the
digital trunk, the signaling is all done on the individual channels - if
a call comes in on channel 7 then channel 7 will see a signal from the
telco - frequently this is a "wink" signal.

A PRI is, again as you said, 23 voice channel with one data/signaling
channels, unless you're outside the USA in which case it's 30 and two.
All of the call setup, tear down, etc. info is sent on the D channel(s).
Since the D channel is actual data, and not just touch tones and hook
flashes, the PRI can actually do more.  For example, you can customize
the caller ID info that is sent out, whereas on your digital trunk the
carrier will control what caller ID is sent to the called party.  If you
want some more detailed info on ISDN (both PRI and BRI) then check this
out:
http://www.ralphb.net/ISDN/intro.html

I'm surprised that your carrier is charging more for a PRI.  Here in
California my carriers are dying to sell me PRI's instead of digital
trunks because they feel PRI's are easier to maintain.  (I don't know if
that's true, but I've had more than one carrier tell me that.)

Who is your carrier and where are you located?

-MC


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Caller ID on Transfers

[Douglas Garstang]

Bruce, 
I bet your doing Asterisk assisted transfers (blindxfer and atxfer in 
features.conf), and not using the 'transfer' soft or hard key on the Polycom 
phones...

  -Original Message-From: Bruce Reeves 
  [mailto:[EMAIL PROTECTED]Sent: Tuesday, July 25, 2006 
  1:53 PMTo: Asterisk Users Mailing List - Non-Commercial 
  DiscussionSubject: Re: [asterisk-users] Caller ID on 
  Transfersthat's odd, our Polycom phones show the original 
  caller id on blind transfers but the callerid of the person doing the attended 
  transfer, in our case a receptionist. 
  On 7/25/06, Doug 
  Lytle <[EMAIL PROTECTED]> 
  wrote:
  Douglas 
Garstang wrote:>> talking to).  Blind transfers show the 
original caller. Doesn't seem to be happening 
that way with Polycom phones and blind/attended transfers.> Both are 
showing the original calling party caller id. > 
___>I can confirm 
this.Both Attended and Blind transfers show the same Caller ID (That 
of theperson doing the transfer).DOug-- Ben 
Franklin quote:"Those who would give up Essential Liberty to 
purchase a little Temporary Safety, deserve neither Liberty nor 
Safety."___ 
--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing 
listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users-- BruceNortex Networks 
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] vegastream 50 FXO DTMF Problem

[Peter Doyle]

Hi Issac,
I would double-check the dtmfmode=inband thing, just to be 100% sure.
We've never been able to get out of band dtmf working with the vega and
asterisk, so we just use inband and it works well enough.

With our setup, the vega doesn't seem to strip out outgoing inband dtmf
tones, so if asterisk sends them inband, it seems like it should make it
to the PSTN.  That's why I think it'd be valuable to just verify
asterisk was actually sending inband dtmf.  I'm assuming the link
between asterisk and the vega is using an uncompressed codec like
g711a/g711u, otherwise maybe codec compression could be distorting the
dtmf tones, making them unrecognizable to the equipment at the far-end.

Here's the appropriate sections from our sip.conf:
[vega]
type=user
dtmfmode=inband
disallow=all
context=from-pstn
allow=ulaw

[vega-gw]
type=peer
host={ip address here}
dtmfmode=inband
disallow=all
context=from-internal
allow=ulaw

Then in the vega web setup, under media, we set the out of band DTMF to
"0" (off) for g711ulaw (which is all we use internally, and asterisk
translates for other codecs).  

Hope that helps.  It atleast was the solution for us
Thanks
Pete

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Issac
Simchayof
Sent: Tuesday, July 25, 2006 10:43 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] vegastream 50 FXO DTMF Problem

Thanks Pete,

I did try dtmfmode=inband and it did not work for us.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter
Doyle
Sent: Tuesday, July 25, 2006 12:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] vegastream 50 FXO DTMF Problem

Hi Issac,
If I recall correctly, out of band DTMF didn't seem to work for us on
our Vega 50 (atleast not when using the Vega with Asterisk).  We had to
tell Asterisk to use dtmfmode=inband in our sip.conf.  It didn't seem
like we had to change any settings on the Vega, because it was sending
both inband and RFC2833 when RFC2833 was selected (for incoming DTMF).
This is all from memory, so hopefully its correct.

I'm headed to work in a few minutes here, where maybe I can see our
actual config and send it along.
Good Luck!
Pete

P.S.  How has your experience been with the echo cancelers?  Are you
experiencing much echo at all?  That's the one thing we're still
struggling with. Thanks


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Issac
Simchayof
Sent: Tuesday, July 25, 2006 8:48 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] vegastream 50 FXO DTMF Problem

Asterisk is sending the DTMF as we can see in ethereal but the Vega is
not sending them out. We did try the debug before ethereal but the tech
at VegaStream insisted we will need ethereal to troubleshoot this.  



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry
Jones
Sent: Tuesday, July 25, 2006 11:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] vegastream 50 FXO DTMF Problem

How do you mean it does not recognize them? By the routing not working
properly?

Or by not outdialing properly?

No need for ethereal, just turn on sip debugging and it will display the
messages for you, just like * will.


On Jul 25, 2006, at 9:43 AM, Issac Simchayof wrote:

>
> I am having a problem with my VegaStream 50 10 FXO. The unit does not 
> recognize DTMF sent from asterisk on outgoing calls. I have been 
> trying to resolve this with VegaStream support but they have not been 
> very helpful so far. On the last test we ran we used Eathereal to 
> capture traffic and it does look like asterisk does send the DTMF 
> codes to the Vega, but the Vega does not recognize them.
>
> Any Ideas from a VegaStream users?
>
>
>
>
>
> ___
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk

RE: [asterisk-users] Caller ID on Transfers

[Douglas Garstang]

> -Original Message-
> From: Joshua Colp [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, July 25, 2006 9:47 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] Caller ID on Transfers
> 
> 
> - Original Message -
> From: Douglas Garstang
> [mailto:[EMAIL PROTECTED]
> To: Asterisk Users Mailing List -
> Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
> Sent:
> Tue, 25 Jul 2006 16:31:13 -0300
> Subject: RE: [asterisk-users] Caller ID on
> Transfers
> 
> > 
> > I thought the new SIP invite had a 'Diverted' field or 
> something in it? If
> > that's true, this is really bad because I need some way in 
> my AGI script to
> > determine that it's a transferred call, and not a new call. 
> In the case of a
> > transferred call, we want to set the caller id to the 
> original calling party
> > info, not the transferring party info. I swear that last 
> week when I was
> > doing this, the RDNIS agi variable was being set and I 
> could use that to set
> > the caller id information as needed. However, now it's 
> suddenly stopped
> > working and I don't know why.
> > 
> > 
> > Is this documented somewhere?
> 
> No, the new INVITE does not have that info... I even just 
> tested it from my Polycom IP600, it was a regular normal 
> INVITE. As for documented about the call flow... probably 
> somewhere on the internet, it's a standard SIP REFER transfer 
> with a replaces.

If the new invite looks like a regular call, how can an AGI script tell that 
it's a transferred call?

Doug.
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Caller ID on Transfers

[Douglas Garstang]

> -Original Message-
> From: Doug Lytle [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, July 25, 2006 1:37 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Caller ID on Transfers
> 
> 
> Douglas Garstang wrote:
> >> talking to).  Blind transfers show the original caller.
> >> 
> >
> > Doesn't seem to be happening that way with Polycom phones 
> and blind/attended transfers.
> > Both are showing the original calling party caller id.
> > ___
> >   
> 
> I can confirm this.
> 
> Both Attended and Blind transfers show the same Caller ID 
> (That of the 
> person doing the transfer).
> 
> DOug

Hi Doug. Your doing your transfers with the transfer keys on the Polycom, 
right? So am I. I think a distinction needs to be made here. I get the 
impression that most people, and certainly the others are using #1 and #2 to do 
Asterisk assisted transfers. 

When we initiated transfers with #1, the caller id of the original caller is 
sent to the destination, and when we do #2, the caller id of the transferring 
party is sent to the destination. However, when using the transfer key on the 
polycom phone, and doing both attended and unattended transfers, the caller id 
of the original party is sent to the destination.

We want to use our transfer buttons on the phones!

It appears that asterisk is treating the sequence of SIP messages from polycom 
phones, such that it thinks all transfers are attended transfers, for the 
purposes of caller id 'pass-thru'.

Has anyone else experienced this?

Doug.


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Caller ID on Transfers

[Bruce Reeves]

that's odd, our Polycom phones show the original caller id on blind transfers but the callerid of the person doing the attended transfer, in our case a receptionist. On 7/25/06, 
Doug Lytle <[EMAIL PROTECTED]> wrote:
Douglas Garstang wrote:>> talking to).  Blind transfers show the original caller. Doesn't seem to be happening that way with Polycom phones and blind/attended transfers.> Both are showing the original calling party caller id.
> ___>I can confirm this.Both Attended and Blind transfers show the same Caller ID (That of theperson doing the transfer).DOug--
Ben Franklin quote:"Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."___
--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   
http://lists.digium.com/mailman/listinfo/asterisk-users-- BruceNortex Networks
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Just bought a Polycom 501 - Ifeellike myGXP-2000 was better...

[Mike]

I didn't want to start a war either.  It was simply an opinion that I
thought was worth expressing after reading all those "GXP-2000 sucks"
messages in the past.

It's still just an opinion, I am certainly not trying to build a consensus.

Thanks for all those who helped me get the phone working.  

Mike



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of shadowym
Sent: July 25, 2006 2:48 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Just bought a Polycom 501 - Ifeellike
myGXP-2000 was better...


I don't want this to go Jihad.

End users have EVERY right to have a phone that is easy to use.  That is all
I am saying.  If it is a nightmare to configure but easy to use that is
fine.  The original post suggested it is neither easy to configure nor use. 

> -Original Message-
> From: Michael Graves [mailto:[EMAIL PROTECTED]
> Sent: Monday, July 24, 2006 9:16 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] Just bought a Polycom 501 - I feellike 
> myGXP-2000 was better...
> 
> On Mon, 24 Jul 2006 18:34:24 -0700, shadowym wrote:
> 
> >One of the problems with opinions on these forums IMHO is
> that they are
> >mostly from technical types and not typical end users.
> 
> Typical end users would have no business configuring a PBX such as 
> Asterisk. A typical end user would leave that to a professional.
> 
> OTOH, those who are responsible for PBX systems often answer to 
> "typical end users" in a fashion, and so very likely have faced a 
> number of opinions about things...well founded and otherwise.
> 
> Michael
> 
> 
> 
> 
> 
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] sdp multipart information nortel

[Jerry Geis]

I'm connecting to a nortel switch SIP
that wants to send multipart sdp information.
Have not found a way to turn that off.
Seems like asterisk does not like it and I get a busy signal.

Is this on the 1.4 list of things?
Anyone else run into this?

Anyway around it?

Jerry

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Caller ID on Transfers

[Joshua Colp]

- Original Message -
From: Douglas Garstang
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Tue, 25 Jul 2006 16:31:13 -0300
Subject: RE: [asterisk-users] Caller ID on
Transfers

> 
> I thought the new SIP invite had a 'Diverted' field or something in it? If
> that's true, this is really bad because I need some way in my AGI script to
> determine that it's a transferred call, and not a new call. In the case of a
> transferred call, we want to set the caller id to the original calling party
> info, not the transferring party info. I swear that last week when I was
> doing this, the RDNIS agi variable was being set and I could use that to set
> the caller id information as needed. However, now it's suddenly stopped
> working and I don't know why.
> 
> 
> Is this documented somewhere?

No, the new INVITE does not have that info... I even just tested it from my 
Polycom IP600, it was a regular normal INVITE. As for documented about the call 
flow... probably somewhere on the internet, it's a standard SIP REFER transfer 
with a replaces.

Joshua Colp
Digium
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Caller ID on Transfers

[Doug Lytle]

Douglas Garstang wrote:

talking to).  Blind transfers show the original caller.



Doesn't seem to be happening that way with Polycom phones and blind/attended 
transfers.
Both are showing the original calling party caller id.
___
  


I can confirm this.

Both Attended and Blind transfers show the same Caller ID (That of the 
person doing the transfer).


DOug

--

Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety."


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Polycom_acd_functions SIP trouble

[James Fromm]

Yeah, we tried that.  Tried every combination of variables in sip.conf. 
 Only solution that works is removing the requirement for a secret.


Faris Raouf wrote:

Dovid Bender wrote:

I am sure you prob. know this but in your configs it shows secret 
commented out. Also it with a softphone if it dosent work then, then 
its your configs. Also did you remember to reload asterisk ?

- Original Message - From: "James Fromm" <[EMAIL PROTECTED]>
To: 
Sent: Monday, July 24, 2006 2:24 PM
Subject: [asterisk-users] Polycom_acd_functions SIP trouble


I'm trying to use the latest revision of Bweschke's branch from SVN 
for polycom_acd_functions.  Asterisk builds and runs without error 
but all SIP devices can't register when specifying a secret in 
sip.conf.  The Polycom 601 I'm testing with and a copy of SJphone 
will not register. IAX from Idefisk works without error.




One thing to try is setting type=peer instead of type=friend. I'm a bit 
dazed and confused at the moment, but if I remember correctly Polycom 
phones just don't work with type=friend.


Of course this doesn't explain why SJPhone won't work either so maybe 
I'm totally off-track, but it might be worth giving it a try just the same.


Faris.

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Polycom_acd_functions SIP trouble

[James Fromm]

Exactly.  If I uncomment the secret, no SIP device or softphone will be 
able to register.  I commented the secret so I could continue to 
configure using this revision of the branch.  No SIP device or softphone 
can register as long as a secret is required.


Dovid Bender wrote:
I am sure you prob. know this but in your configs it shows secret 
commented out. Also it with a softphone if it dosent work then, then its 
your configs. Also did you remember to reload asterisk ?

- Original Message - From: "James Fromm" <[EMAIL PROTECTED]>
To: 
Sent: Monday, July 24, 2006 2:24 PM
Subject: [asterisk-users] Polycom_acd_functions SIP trouble


I'm trying to use the latest revision of Bweschke's branch from SVN 
for polycom_acd_functions.  Asterisk builds and runs without error but 
all SIP devices can't register when specifying a secret in sip.conf.  
The Polycom 601 I'm testing with and a copy of SJphone will not 
register. IAX from Idefisk works without error.


The error all SIP devices get is:

Jul 24 10:26:48 NOTICE[31524]: chan_sip.c:14203 
handle_request_register: Registration from 
'' failed for '192.168.0.95' - 
Username/auth name mismatch


Commenting the definition of a secret in sip.conf for the device 
solves this.  Here's the config for one of the devices.


[1003]
type=friend
canreinvite=no
host=dynamic
username=1003
; secret=stuff
context=outbound
callerid="Jimmy" <1003>
[EMAIL PROTECTED]
nat=no

Why won't this revision accept the definition of a secret?  Am I 
missing something simple (stupid)?


Thanks,
Jay

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users




___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] RDNIS and IAX2

[Douglas Garstang]

> -Original Message-
> From: Brian Capouch [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, July 25, 2006 12:53 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] RDNIS and IAX2
> 
> 
> Douglas Garstang wrote:
> > I'll probably get blasted for this. I hope I'm wrong, and 
> then a little blasting is ok. It appears that Asterisk may 
> have let us down again as a 'carrier grade' solution.
> > 
> 
> Did the list software screw up, or did you post this exact same mail 
> yesterday?

I posted the exact same mail yesterday.
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Caller ID on Transfers

[Douglas Garstang]

> -Original Message-
> From: Joshua Colp [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, July 25, 2006 8:54 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] Caller ID on Transfers
> 
> 
> - Original Message -
> From: Douglas Garstang
> [mailto:[EMAIL PROTECTED]
> To: Asterisk Users Mailing List -
> Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
> Sent:
> Tue, 25 Jul 2006 15:37:10 -0300
> Subject: RE: [asterisk-users] Caller ID on
> Transfers
> 
> > > 
> > > What type of transfer? blind or attended?
> > 
> > Does it matter? Both...
> > 
> > Doug.
> 
> Yes, it does matter. An attended SIP transfer is handled much 
> differently then a blind transfer. It starts out as a regular 
> call to another person, Asterisk doesn't know it's actually a 
> transfer. Then when the transfer actually happens the phone 
> says "hey, this call I have up over here to you... it's 
> replacing this other call".

I thought the new SIP invite had a 'Diverted' field or something in it? If 
that's true, this is really bad because I need some way in my AGI script to 
determine that it's a transferred call, and not a new call. In the case of a 
transferred call, we want to set the caller id to the original calling party 
info, not the transferring party info. I swear that last week when I was doing 
this, the RDNIS agi variable was being set and I could use that to set the 
caller id information as needed. However, now it's suddenly stopped working and 
I don't know why.

> 
> Call flow:
> 
> Call #1
> Phone A ---> Asterisk ---> Phone B
> (Phone A performs attended transfer)
> Phone B is put on hold.
> 
> Call #2
> Phone A ---> Asterisk ---> Phone C
> (Phone A transfers Phone B to Phone C)
> 
> Transfer
> "Hey Asterisk, this call with ID 4653456sdfgawe45 I have up 
> to Phone B... it's replacing this call adsf8wet I have up to 
> Phone C -- they should be talking to eachother"
> Phone B ---> Asterisk ---> Phone C
> Phone A disappears out of both calls.
> 
> With a blind transfer the phone can simply say hey channel... 
> this is your new extension and context.

Is this documented somewhere?
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] PRI vs "Digital Trunk"

[Barry D. Hassler]

Hi, can someone enlighten me as to the difference between a PRI and a
"Digital Trunk" (other than cost)?

I do understand PRI (B-channel signaling, incoming/outgoing call setup,
D channel for voice/data, etc), but I'm not quite sure how that compares
with what my vendor is calling a "Digital Trunk" (specifically in
contrast to a PRI). The PRI is about twice the cost. 

If this is just a channelized T1 (24 64k voice/data "channels'), would
they each be assigned a specific phone number, or is there further
flexibility in sending/receiving calls, callerid (receive or send), etc?

Feeling ignorant here

Thanks!
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] SIP and podcasts

[kael]

Alex Robar wrote:
Should be doable, but it would take a bit of scripting. You would have 
to get a program that subscribes to the feeds in Linux (bashpodder does 
this) and downloads the files to a given directory. You would then have 
to run something to convert those mp3s into something Asterisk can use, 
then move the converted files to the appropriate MoH directories. Create 
a dial code (say *703 -> *POD) and have it play MoH for the files in 
that folder.


If you wanted to be really clever about it, you could create an IVR that 
lets you pick from the most recent show, one show back, two shows back, 
etc, etc... And have the script that copies the files over remove the 
oldest file, rename the older files so that they become "show2" and 
"show3", and then rename the newest show to "show1".


Thanks for the suggestion.

I'd need to learn scripting. :-/

I haven't installed Asterisk yet but was thinking there would a hack 
specially for podcasts.


I find curious there's no solution to listen to podcasts via SIP servers.

--
kael
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Caller ID on Transfers

[Douglas Garstang]

> -Original Message-
> From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, July 25, 2006 12:48 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Caller ID on Transfers
> 
> 
> On Tuesday 25 July 2006 14:37, Douglas Garstang wrote:
> > > What type of transfer? blind or attended?
> >
> > Does it matter? Both...
> 
> Yes it does matter.  On any KSU or PBX I have used, attended 
> transfers show 
> the name/extension of the transferer (presumably because it 
> is THEM you are 
> talking to).  Blind transfers show the original caller.

Doesn't seem to be happening that way with Polycom phones and blind/attended 
transfers.
Both are showing the original calling party caller id.
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] SIP and podcasts

[kael]

Marco Mouta wrote:

GABcast has IVR to allow users access podcast from Asterisk


GABcast doesn't seem to allow to subscribe to podcasts but only to 
create them from a softphone, does it ?


--
kael
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE : [asterisk-users] Connecting branch offices through IPsec tunnel --latency effects?

[f6hqz-m]

Hi Stephen,

+99 ms via IPSec FreeSWan
But good protection and no NAT issue.

Francois BERGERET,
France.

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Stephen Bosch
Envoyé : mardi 25 juillet 2006 17:25
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : [asterisk-users] Connecting branch offices through IPsec tunnel
--latency effects?


Hi:

If I connect two offices through an IPsec tunnel, what is the impact on
latency, and does it noticeably affect calls?

Has anyone out there tried this? What were the effects?

Cheers,

-Stephen-
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] RDNIS and IAX2

[Brian Capouch]

Douglas Garstang wrote:

I'll probably get blasted for this. I hope I'm wrong, and then a little 
blasting is ok. It appears that Asterisk may have let us down again as a 
'carrier grade' solution.



Did the list software screw up, or did you post this exact same mail 
yesterday?


B.

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Caller ID on Transfers

[Andrew Kohlsmith]

On Tuesday 25 July 2006 14:37, Douglas Garstang wrote:
> > What type of transfer? blind or attended?
>
> Does it matter? Both...

Yes it does matter.  On any KSU or PBX I have used, attended transfers show 
the name/extension of the transferer (presumably because it is THEM you are 
talking to).  Blind transfers show the original caller.

-A.
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Recommend hard phone which supports IAX2?

[Gonzalo Servat]

On 7/25/06, Stephen Bosch <[EMAIL PROTECTED]> wrote:

Hi:

I'm setting up a branch office, but I don't want to trunk from the main
office because I don't want to introduce any more latency. Also, the
office will have only a single extension, so I can't justify the expense
of a second Asterisk server for it.

SIP is a pain when going through firewalls, and I'm worried about the
latency that would come with using an IPsec tunnel between the two
sites, so I'm looking for an IAX2 supporting hard phone, and want to
hear recommendations from people who have had direct experience with such.

What are the best IAX2 hard phones?


Hi,

You could just get an IAXy and connect an analog phone onto it?

Regards,
Gonzalo.
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] All Extensions Dropped

[Henry F. Camacho Jr.]

I have an Asterisk host connected to a T1 facility, and another Asterisk 
host connected via an IAX trunk in another location.  I have Ring groups 
defined to ring a number of extensions at once.  Intermittently when one 
of these ring groups is triggered, everyone that is on a phone call in 
the remote location is disconnected from their calls.


I think I have narrowed this down a bit to the following log entries on 
the Asterisk host that has the trunks, and is connected to the other 
server VIA IAX trunks


Jul 24 11:33:57 VERBOSE[1205]: -- Accepting call from '9529350635' 
to '7632353001' on channel 0/1, span 1

Jul 24 11:33:57 DEBUG[1205]: Enabled echo cancellation on channel 1
Jul 24 11:33:57 VERBOSE[13405]: -- Executing Dial("Zap/1-1", 
"IAX2/ucfrid/3001") in new stack

Jul 24 11:33:57 VERBOSE[13405]: -- Called ucfrid/3001
Jul 24 11:33:57 VERBOSE[1171]: -- Call accepted by 67.88.180.210 
(format ulaw)

Jul 24 11:33:57 VERBOSE[1171]: -- Format for call is ulaw
Jul 24 11:33:58 VERBOSE[13405]: -- IAX2/ucfrid/1 is ringing
Jul 24 11:33:58 VERBOSE[13405]: -- IAX2/ucfrid/1 is ringing
Jul 24 11:34:05 DEBUG[1171]: Raw Hangup 67.88.180.210:4569, src=2, dst=2
Jul 24 11:34:05 DEBUG[1171]: Raw Hangup 67.88.180.210:4569, src=2, dst=2
Jul 24 11:34:05 DEBUG[1171]: Raw Hangup 67.88.180.210:4569, src=2, dst=2
Jul 24 11:34:05 DEBUG[1171]: Raw Hangup 67.88.180.210:4569, src=2, dst=2
Jul 24 11:34:18 VERBOSE[13405]: -- IAX2/ucfrid/1 stopped sounds
Jul 24 11:34:18 VERBOSE[13405]: -- IAX2/ucfrid/1 answered Zap/1-1
Jul 24 11:34:18 DEBUG[1171]: Ooh, voice format changed to 4
Jul 24 11:34:26 WARNING[1171]: Max retries exceeded to host 
67.88.180.210 on IAX2/[EMAIL PROTECTED]:65065/2 (type = 6,

subclass = 2, ts=958873, seqno=46)
Jul 24 11:34:26 WARNING[1171]: Max retries exceeded to host 
67.88.180.210 on IAX2/[EMAIL PROTECTED]:65065/2 (type = 6,

subclass = 11, ts=958876, seqno=47)
Jul 24 11:34:26 DEBUG[13402]: Didn't get a frame from channel: 
IAX2/[EMAIL PROTECTED]:65065/2
Jul 24 11:34:26 DEBUG[13402]: Bridge stops bridging channels 
IAX2/[EMAIL PROTECTED]:65065/2 and Zap/4-1

Jul 24 11:34:26 DEBUG[13402]: Set option AUDIO MODE, value: ON(1) on Zap/4-1
Jul 24 11:34:26 DEBUG[13402]: Hangup: channel: 4 index = 0, normal = 25, 
callwait = -1, thirdcall = -1
Jul 24 11:34:26 DEBUG[13402]: Not yet hungup...  Calling hangup once 
with icause, and clearing call

Jul 24 11:34:26 DEBUG[13402]: disabled echo cancellation on channel 4
Jul 24 11:34:26 DEBUG[13402]: Set option TDD MODE, value: OFF(0) on Zap/4-1
Jul 24 11:34:26 DEBUG[13402]: Updated conferencing on 4, with 0 
conference users
Jul 24 11:34:26 DEBUG[13402]: Set option AUDIO MODE, value: OFF(0) on 
Zap/4-1

Jul 24 11:34:26 DEBUG[13402]: disabled echo cancellation on channel 4
Jul 24 11:34:26 VERBOSE[13402]: -- Hungup 'Zap/4-1'
Jul 24 11:34:26 DEBUG[13402]: Exiting with DIALSTATUS=ANSWER.
Jul 24 11:34:26 VERBOSE[13402]:   == Spawn extension (longdistance, 
918774263777, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]

.180.210:65065/2'
Jul 24 11:34:26 DEBUG[13402]: We're hanging up 
IAX2/[EMAIL PROTECTED]:65065/2 now...
Jul 24 11:34:26 VERBOSE[13402]: -- Hungup 
'IAX2/[EMAIL PROTECTED]:65065/2'

Jul 24 11:34:30 VERBOSE[1205]: -- Channel 0/1, span 1 got hangup
Jul 24 11:34:30 DEBUG[13405]: Bridge stops because we're zombie or need 
a soft hangup: c0=Zap/1-1, c1=IAX2/ucfrid/1, flags

: No,Yes,No,No

This is extremely intermittent, and otherwise the system works just 
fine.  The above shows a call coming into an a number.  The extension 
was in use, and then the system just seemed to tear down the whole IAX 
trunk.


HFC

:
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Just bought a Polycom 501 - I feellike myGXP-2000 was better...

[shadowym]

I don't want this to go Jihad.

End users have EVERY right to have a phone that is easy to use.  That is all
I am saying.  If it is a nightmare to configure but easy to use that is
fine.  The original post suggested it is neither easy to configure nor use. 

> -Original Message-
> From: Michael Graves [mailto:[EMAIL PROTECTED] 
> Sent: Monday, July 24, 2006 9:16 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] Just bought a Polycom 501 - I 
> feellike myGXP-2000 was better...
> 
> On Mon, 24 Jul 2006 18:34:24 -0700, shadowym wrote:
> 
> >One of the problems with opinions on these forums IMHO is 
> that they are 
> >mostly from technical types and not typical end users.
> 
> Typical end users would have no business configuring a PBX 
> such as Asterisk. A typical end user would leave that to a 
> professional.
> 
> OTOH, those who are responsible for PBX systems often answer 
> to "typical end users" in a fashion, and so very likely have 
> faced a number of opinions about things...well founded and otherwise.
> 
> Michael
> 
> 
> 
> 
> 
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Caller ID on Transfers

[Joshua Colp]

- Original Message -
From: Douglas Garstang
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Tue, 25 Jul 2006 15:37:10 -0300
Subject: RE: [asterisk-users] Caller ID on
Transfers

> > 
> > What type of transfer? blind or attended?
> 
> Does it matter? Both...
> 
> Doug.

Yes, it does matter. An attended SIP transfer is handled much differently then 
a blind transfer. It starts out as a regular call to another person, Asterisk 
doesn't know it's actually a transfer. Then when the transfer actually happens 
the phone says "hey, this call I have up over here to you... it's replacing 
this other call".

Call flow:

Call #1
Phone A ---> Asterisk ---> Phone B
(Phone A performs attended transfer)
Phone B is put on hold.

Call #2
Phone A ---> Asterisk ---> Phone C
(Phone A transfers Phone B to Phone C)

Transfer
"Hey Asterisk, this call with ID 4653456sdfgawe45 I have up to Phone B... it's 
replacing this call adsf8wet I have up to Phone C -- they should be talking to 
eachother"
Phone B ---> Asterisk ---> Phone C
Phone A disappears out of both calls.

With a blind transfer the phone can simply say hey channel... this is your new 
extension and context.

Joshua Colp
Digium
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Recommend hard phone which supports IAX2?

[Tim Panton]

On 25 Jul 2006, at 16:23, Stephen Bosch wrote:


Hi:

I'm setting up a branch office, but I don't want to trunk from the  
main

office because I don't want to introduce any more latency. Also, the
office will have only a single extension, so I can't justify the  
expense

of a second Asterisk server for it.

SIP is a pain when going through firewalls, and I'm worried about the
latency that would come with using an IPsec tunnel between the two
sites, so I'm looking for an IAX2 supporting hard phone, and want to
hear recommendations from people who have had direct experience  
with such.


What are the best IAX2 hard phones?


I've got a couple of IAX hardphones, with PA168, they are useable,
but only just. They are hard to hang up (which is a design problem)
and  a pain to get transfer working (which is a software problem).

Much as I love IAX, I advise you to buy a decent SIP phone
(SNOM?).

At home I have a SIP phone and an nslu2 running asterisk, just to act  
as a

protocol converter, but any old 486 or PII will do the
trick.

Tim.

Tim Panton

www.mexuar.com



___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Caller ID on Transfers

[Douglas Garstang]

> -Original Message-
> From: Joshua Colp [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, July 25, 2006 8:41 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Caller ID on Transfers
> 
> 
> 
> 
> - Original Message -
> From: Douglas Garstang
> [mailto:[EMAIL PROTECTED]
> To: Asterisk Users Mailing List -
> Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
> Sent:
> Tue, 25 Jul 2006 15:25:15 -0300
> Subject: [asterisk-users] Caller ID on
> Transfers
> 
> 
> > I have three phones here with extensions 2944093, 3254103 
> and 9220371.
> >  
> > 2944093 calls 3254103. 3254103 transfers 2944093 to 
> 9220371. We want the
> > caller id of 2944093 to be presented on the display of 9220371.
> > However, the caller id of the transferer, 3254103, is 
> appearing. This
> > doesn't make any sense.
> >  
> > How can we do this?
> >  
> > Doug.
> >  
> 
> What type of transfer? blind or attended?

Does it matter? Both...

Doug.
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Caller ID on Transfers

[Joshua Colp]

- Original Message -
From: Douglas Garstang
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Tue, 25 Jul 2006 15:25:15 -0300
Subject: [asterisk-users] Caller ID on
Transfers


> I have three phones here with extensions 2944093, 3254103 and 9220371.
>  
> 2944093 calls 3254103. 3254103 transfers 2944093 to 9220371. We want the
> caller id of 2944093 to be presented on the display of 9220371.
> However, the caller id of the transferer, 3254103, is appearing. This
> doesn't make any sense.
>  
> How can we do this?
>  
> Doug.
>  

What type of transfer? blind or attended?

Joshua Colp
Digium
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Re: Still voice with echo

[M.Hockings]

Carlos Alberto Bernat Orozco wrote:

Hi group

Thanks Marty for your colaboration. I tried the my voice call with 2 
extensions and SJphone as softphone as you know. For the test I used a 
normal mic plug into the mic port from a laptop and made the call to 
another pc wich has second extension. At first time I believed what you 
told me about the feedback, but it's constant no matter if I put away 
from the speakers. The voice sounds with echo and keeps constants when I 
say :"hello" and sound very bad.


I had similar echo problems using my laptop but it was resolved by a 
combination of using a headset (mic and headphones) and turning down the 
mic gain.  It takes very little pickup from the speakers to the mic to 
cause an objectionable amount of background echo/noise so possibly just 
moving the mic away some is not sufficient.


Also you might make sure that the PC has enough power to deliver the 
sound fast enough (would the echo test in asterisk check this?).  That 
is, if I use an old slow box that does not run a soft phone very well 
you will hear slurring of the sound probably as the jitter buffer tries 
to compensate for the breaks in the data stream?


Mike

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Caller ID on Transfers

[Douglas Garstang]

I have 
three phones here with extensions 2944093, 3254103 and 
9220371.
 
2944093 calls 3254103. 3254103 transfers 2944093 to 9220371. We want the 
caller id of 2944093 to be presented on the display of 
9220371.
However, the caller id of the transferer, 3254103, is appearing. This 
doesn't make any sense.
 
How 
can we do this?
 
Doug.
 
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Clocking Multiple T1 Cards

[Shaw Terwilliger]

Bruce Reeves wrote:
> It will cause issues if you are using fax/modems on the channel bank and
> trying to send out via the PRI. We had a great deal of problems with
> timing sync between 2 spans on a Sangoma A104D until the latest beta
> drivers were released.

No faxes here.  After reading dozens of FAX threads on asterisk-users,
I've just kept a dedicated POTS line for the fax machine.

-- 
Shaw Terwilliger <[EMAIL PROTECTED]>
SourceGear LLC



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] transfers from an E1 using 2b-channel or similar anyone? (QSIG?)

[Manrique Feoli]

Hi Matt,  thanks for your answer,
I guess it is still as you said a while back that you did it using 5ESS

Can you share how you did in 5ESS?  (a sample of the extensions.conf ) 
  and what kind of switch you were connected to?


I'm not sure if the  Alcatel 4400 and the Nortel Meridian 11 supports 
5ESS,  but are willing to find out.


thanks

Manrique


Matthew Fredrickson escribió:

On Jul 25, 2006, at 12:53 PM, Manrique Feoli wrote:


Hi all,
Here is the situation:

A call comes in to an Alcatel PBX and it sends it to an E1 on * ,   
this * either sends the call to a VoIP extension or needs to forward 
it to an extension back on the Alcatel,  but WITHOUT using another 
slot of the E1  (no tromboning or hairpinning).


I've read you can do this with 2b channel transfers implemented on 
5ESS, and also on QSIG.

I know Matthew Fredrickson did it on *  (I think he programmed it for *)

I also know there is quite a bit of people pursuing this same goal,   
which is way important to lower the income barriers for * to enter 
the legacy world.


Has anyone actually done it? I appreciate any input whatsoever,  
and if possible a sample of how to manage it on *.What to put on 
the extensions.conf to perform the transfer and any other files needed,




Unfortunately, I have not implemented the Q.SIG version of 2b channel 
transfer, so for the time being you'll have to stick to hairpinning 
the legs of the call.  The Q.SIG version is a little bit more 
complicated than some of the other versions.


Matthew Fredrickson



___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Clocking Multiple T1 Cards

[Bruce Reeves]

It will cause issues if you are using fax/modems on the channel bank and trying to send out via the PRI. We had a great deal of problems with timing sync between 2 spans on a Sangoma A104D until the latest beta drivers were released.
On 7/25/06, Shaw Terwilliger <[EMAIL PROTECTED]> wrote:
Andrew Kohlsmith wrote:> What I was trying to state was that if you have two data streams that are> solidly clocked but out of phase, you will not encounter any of these issues.> If the clock period of either (or both) drifts then yes, you will run into
> trouble.So it sounds like Asterisk can't synchronize the clocks between theDigium and Sangoma boards (or any two PCI boards), and this just may bea limitation of the T1-peripheral-on-PCI architecture.  But it really
shouldn't matter because of the nature of my setup: errors caused bytiming mismatch between the PRI and channel banks won't cause noticeablequality issues.  Do I have it right?--Shaw Terwilliger <
[EMAIL PROTECTED]>SourceGear LLC___--Bandwidth and Colocation provided by 
Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
-- BruceNortex Networks
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] transfers from an E1 using 2b-channel or similar anyone? (QSIG?)

[Matthew Fredrickson]

On Jul 25, 2006, at 12:53 PM, Manrique Feoli wrote:


Hi all,
Here is the situation:

A call comes in to an Alcatel PBX and it sends it to an E1 on * ,   
this * either sends the call to a VoIP extension or needs to forward 
it to an extension back on the Alcatel,  but WITHOUT using another 
slot of the E1  (no tromboning or hairpinning).


I've read you can do this with 2b channel transfers implemented on 
5ESS, and also on QSIG.
I know Matthew Fredrickson did it on *  (I think he programmed it for 
*)


I also know there is quite a bit of people pursuing this same goal,   
which is way important to lower the income barriers for * to enter the 
legacy world.


Has anyone actually done it? I appreciate any input whatsoever,  
and if possible a sample of how to manage it on *.What to put on 
the extensions.conf to perform the transfer and any other files 
needed,




Unfortunately, I have not implemented the Q.SIG version of 2b channel 
transfer, so for the time being you'll have to stick to hairpinning the 
legs of the call.  The Q.SIG version is a little bit more complicated 
than some of the other versions.


Matthew Fredrickson

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] New message

[Bruce Reeves]

Check your cron jobs, especially since it happened while you were asleep, mine runs at 4:00 am evey day.On 7/25/06, Ira <
[EMAIL PROTECTED]> wrote:Well, I was asleep when it happened and no one else has access to the
machine. Does that mean someone logged in from outside and I shouldbe worried about the security of my machine?Ira>Someone connected to the Asterisk console using "asterisk -r" then
>typed "logger reload" then exited the session.>>Ira wrote:>>This morning I found this message on my Asterisk Console. Does it>>mean I should be concerned about the security of my system?
>>-- Remote UNIX connection>>== Parsing '/etc/asterisk/logger.conf': Found>>Asterisk Event Logger Restarted>>-- Remote UNIX connection disconnected___
--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   
http://lists.digium.com/mailman/listinfo/asterisk-users-- BruceNortex Networks
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] transfers from an E1 using 2b-channel or similar anyone? (QSIG?)

[Manrique Feoli]

Hi all,
Here is the situation:

A call comes in to an Alcatel PBX and it sends it to an E1 on * ,   this 
* either sends the call to a VoIP extension or needs to forward it to an 
extension back on the Alcatel,  but WITHOUT using another slot of the 
E1  (no tromboning or hairpinning).


I've read you can do this with 2b channel transfers implemented on 
5ESS, and also on QSIG.

I know Matthew Fredrickson did it on *  (I think he programmed it for *)

I also know there is quite a bit of people pursuing this same goal,   
which is way important to lower the income barriers for * to enter the 
legacy world.


Has anyone actually done it? I appreciate any input whatsoever,  and 
if possible a sample of how to manage it on *.What to put on the 
extensions.conf to perform the transfer and any other files needed,


thanks



___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] vegastream 50 FXO DTMF Problem

[Issac Simchayof]

Thanks Pete,

I did try dtmfmode=inband and it did not work for us.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter Doyle
Sent: Tuesday, July 25, 2006 12:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] vegastream 50 FXO DTMF Problem

Hi Issac,
If I recall correctly, out of band DTMF didn't seem to work for us on
our Vega 50 (atleast not when using the Vega with Asterisk).  We had to
tell Asterisk to use dtmfmode=inband in our sip.conf.  It didn't seem
like we had to change any settings on the Vega, because it was sending
both inband and RFC2833 when RFC2833 was selected (for incoming DTMF).
This is all from memory, so hopefully its correct.

I'm headed to work in a few minutes here, where maybe I can see our
actual config and send it along.
Good Luck!
Pete

P.S.  How has your experience been with the echo cancelers?  Are you
experiencing much echo at all?  That's the one thing we're still
struggling with. Thanks


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Issac
Simchayof
Sent: Tuesday, July 25, 2006 8:48 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] vegastream 50 FXO DTMF Problem

Asterisk is sending the DTMF as we can see in ethereal but the Vega is
not sending them out. We did try the debug before ethereal but the tech
at VegaStream insisted we will need ethereal to troubleshoot this.  



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry
Jones
Sent: Tuesday, July 25, 2006 11:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] vegastream 50 FXO DTMF Problem

How do you mean it does not recognize them? By the routing not working
properly?

Or by not outdialing properly?

No need for ethereal, just turn on sip debugging and it will display the
messages for you, just like * will.


On Jul 25, 2006, at 9:43 AM, Issac Simchayof wrote:

>
> I am having a problem with my VegaStream 50 10 FXO. The unit does not 
> recognize DTMF sent from asterisk on outgoing calls. I have been 
> trying to resolve this with VegaStream support but they have not been 
> very helpful so far. On the last test we ran we used Eathereal to 
> capture traffic and it does look like asterisk does send the DTMF 
> codes to the Vega, but the Vega does not recognize them.
>
> Any Ideas from a VegaStream users?
>
>
>
>
>
> ___
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] New message

[Ira]

Well, I was asleep when it happened and no one else has access to the 
machine. Does that mean someone logged in from outside and I should 
be worried about the security of my machine?


Ira

Someone connected to the Asterisk console using "asterisk -r" then 
typed "logger reload" then exited the session.


Ira wrote:
This morning I found this message on my Asterisk Console. Does it 
mean I should be concerned about the security of my system?

-- Remote UNIX connection
== Parsing '/etc/asterisk/logger.conf': Found
Asterisk Event Logger Restarted
-- Remote UNIX connection disconnected


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Clocking Multiple T1 Cards

[Shaw Terwilliger]

Andrew Kohlsmith wrote:
> What I was trying to state was that if you have two data streams that are 
> solidly clocked but out of phase, you will not encounter any of these issues. 
>  
> If the clock period of either (or both) drifts then yes, you will run into 
> trouble.

So it sounds like Asterisk can't synchronize the clocks between the
Digium and Sangoma boards (or any two PCI boards), and this just may be
a limitation of the T1-peripheral-on-PCI architecture.  But it really
shouldn't matter because of the nature of my setup: errors caused by
timing mismatch between the PRI and channel banks won't cause noticeable
quality issues.  Do I have it right?

-- 
Shaw Terwilliger <[EMAIL PROTECTED]>
SourceGear LLC



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] netstats like command for sip , Is there one ?

[Mr. James W. Laferriere]

Hello All ,  Is there a command or set of commands that will give the
same data & resources as 'iax2 show netstats' for sip ?
Tia ,  JimL
--
+--+
| James   W.   Laferriere |   SystemTechniques   | Give me VMS |
| NetworkEngineer | 3600 14th Ave SE #20-103 |  Give me Linux  |
| [EMAIL PROTECTED] |  Olympia ,  WA.   98501  |   only  on  AXP |
+--+
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users