Re: SV: [Asterisk-Users] periodic-announce not working

[Michiel van Baak]

That depends. Has the channel going into Queue been "answered" yet
before you send it in with r ?


I tried both with and without Answer()
Makes no difference

Michiel
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Re: [Asterisk-Users] problem - DSL line and Digium card

[Martin Joseph]

On Jun 22, 2006, at 7:06 PM, Eric Hartley wrote:


I was wondering if anyone has seen a similar problem...

I have a DSL line that doubles as a voice line to my Asterisk box.  
When the Digium card answers that line, the DSL modem is disconnected 
until the line releases.  The phone line is split at the DSL modem 
then run to an FXO port.


I'm using one of the 4 port analog Digium cards (3 FXO/1 FXS).

Do you have a line filter on the FXO port?

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Re: [Asterisk-Users] Routing inboud from ISDN to second * server.

[Armin Schindler]

On Fri, 23 Jun 2006, Thomas Laurids Pedersen wrote:
> Hi All,
> 
> I have setup 2 asterisk servers using AAH 2.8. I have configured a IAX2
> trunk between the 2 servers using the guide on dumbme. Trunk is not using
> register string and no authentication.
> 
> In my dial plan I have 7XX numers on server B and 6xx numbers on Server A.
> 
> Calls from my SNOM phones are ok between the extensions on the 2 servers.
> 
> In server A I have a eicon 4 port BRI card connected. Calls from outside to
> extensions on server A works fine. (Inbound routes are configured for
> "local" extensions).
> 
> Now how do I configure so that extensions on server B can be called from
> the ISDN connections ? like this ISDN -> ServerA   ->(IAX2)-> ServerB ->
> extension
> 
> Server A does not have direct network access to the SNOM phone.
> 
> >From my reading I think that I have to make an inbound route or asterisk
> will not treat the call at all. This is also what I see if I debug capi.
> 
> I have tried to make an inbound route and custom extension for all 7XX
> extentions and then entering the dial string IAX2/astB/7255 for extention
> 725. Also tried the dial string IAX2/[EMAIL PROTECTED] with no luck.
> 
> How do I do this ?

How does your extensions.conf look like?
I don't understand the problem. There is no difference between
isdn/capi calls or SNOM calls here, except if you have other ISDN numbers
you want to 'map' to the SNOM phones. The Dial() commands should look like 
the same as for the other astA <-> astB connections.
Maybe you can give a real example.

Armin

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[Asterisk-Users] GXP 2000 - BLF and Hold/Hangup Answering

[Corporate IT Solutions - Michael Dunne]

I have a network of GXP 2000 phones and would like to know if there is a
way to configure the phones so that if there is one person talking, and
another call comes in then they can hold/hangup that call and take the
incoming call.

At the moment, when a call comes in and the phone is offhook, then that
phone is completely unavailable for that ring session, any call coming
in after that call will of course ring.

Is this limited to the GXP series or does the SNOM phones fix this, etc.

Any advice is appreciated of course.
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[Asterisk-Users] Asterisk-1.2.9.1 e MOH

[Josué Conti]

Hi All
Somebody knows as resolv the error below? Already I compiled asterisk-addons-1.2.3, but exactly thus it reports this error, could help me?

 
-- Executing WaitMusicOnHold("SIP/3205-d9ef", "30") in new stack    -- Started music on hold, class 'default', on SIP/3205-d9efJun 23 02:14:21 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame 49443303 
Jun 23 02:14:21 WARNING[24960]: common.c:134 decode_header: Layer 1 not supported!Jun 23 02:14:21 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame e4e45a8aJun 23 02:14:21 WARNING[24960]: 
layer3.c:966 III_dequantize_sample: mpg123: Can't rewind stream by 1758 bits!Jun 23 02:14:21 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame 49443303Jun 23 02:14:21 WARNING[24960]: common.c
 :134 decode_header: Layer 1 not supported!Jun 23 02:14:21 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame e4e45a8aJun 23 02:14:21 WARNING[24960]: layer3.c:966 III_dequantize_sample: mpg123: Can't rewind stream by 1758 bits! 
Jun 23 02:14:21 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame Jun 23 02:14:21 WARNING[24960]: common.c:134 decode_header: Layer 1 not supported!Jun 23 02:14:22 WARNING[24960]: 
interface.c:215 decodeMP3: Junk at the beginning of frame 0b00eaffJun 23 02:14:22 WARNING[24960]: common.c:134 decode_header: Layer 1 not supported!Jun 23 02:14:22 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame ebfff3ff 
Jun 23 02:14:22 WARNING[24960]: common.c:134 decode_header: Layer 1 not supported!Jun 23 02:14:22 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame 33003700Jun 23 02:14:22 WARNING[24960]: 
common.c:134 decode_header: Layer 1 not supported!Jun 23 02:14:22 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame 1f001800Jun 23 02:14:22 WARNING[24960]: common.c:134 decode_header: Layer 1 not supported! 
Jun 23 02:14:22 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame e2ffd8ffJun 23 02:14:22 WARNING[24960]: common.c:134 decode_header: Layer 1 not supported!Jun 23 02:14:22 WARNING[24960]: 
interface.c:215 decodeMP3: Junk at the beginning of frame dcffddffJun 23 02:14:22 WARNING[24960]: common.c:134 decode_header: Layer 1 not supported!Jun 23 02:14:22 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame d6ffc6ff 
Jun 23 02:14:22 WARNING[24960]: common.c:134 decode_header: Layer 1 not supported!Jun 23 02:14:22 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame 1f002300Jun 23 02:14:22 WARNING[24960]: 
common.c:134 decode_header: Layer 1 not supported!Jun 23 02:14:22 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame cfffc0ffJun 23 02:14:22 WARNING[24960]: common.c:134 decode_header: Layer 1 not supported! 
Jun 23 02:14:22 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame 20001a00Jun 23 02:14:22 WARNING[24960]: common.c:134 decode_header: Layer 1 not supported!Jun 23 02:14:22 WARNING[24960]: 
interface.c:215 decodeMP3: Junk at the beginning of frame f7ff1300Jun 23 02:14:22 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame e1ffe7ffJun 23 02:14:22 WARNING[24960]: common.c:134 decode_header: Layer 1 not supported! 
Jun 23 02:14:22 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame daffd4ffJun 23 02:14:22 WARNING[24960]: common.c:134 decode_header: Layer 1 not supported!Jun 23 02:14:22 WARNING[24960]: 
interface.c:215 decodeMP3: Junk at the beginning of frame 2a003d00Jun 23 02:14:22 WARNING[24960]: common.c:134 decode_header: Layer 1 not supported!Jun 23 02:14:22 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame dbffd8ff 
Jun 23 02:14:22 WARNING[24960]: common.c:134 decode_header: Layer 1 not supported!Jun 23 02:14:22 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame c6ffc5ffJun 23 02:14:22 WARNING[24960]: 
common.c:134 decode_header: Layer 1 not supported!Jun 23 02:14:22 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame ddfff1ffJun 23 02:14:22 WARNING[24960]: common.c:134 decode_header: Layer 1 not supported! 
Jun 23 02:14:22 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame e3ffecffJun 23 02:14:22 WARNING[24960]: common.c:134 decode_header: Layer 1 not supported!Jun 23 02:14:22 WARNING[24960]: 
interface.c:215 decodeMP3: Junk at the beginning of frame d7ffecffJun 23 02:14:22 WARNING[24960]: common.c:134 decode_header: Layer 1 not supported!Jun 23 02:14:22 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame d8ffd3ff 
Jun 23 02:14:22 WARNING[24960]: common.c:134 decode_header: Layer 1 not supported!Jun 23 02:14:22 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame 0e000500Jun 23 02:14:22 WARNING[24960]: 
common.c:134 decode_header: Layer 1 not supported!Jun 23 02:14:22 WARNING[24960]: inter

Re: [Asterisk-Users] iax2 registration problems

[Bartosz Wegrzyn - asterisk]

this helped thank you

> Bartosz Wegrzyn - asterisk wrote:
>> On the asterisk1 I got this:
>>
>> register => username:[EMAIL PROTECTED]
>>
>>
> Try changing this to :
>
> register => incommingiax2:[EMAIL PROTECTED]
>
> There is a good reason I say this, but I'm too tired to remember what it
> is.
>> [eop]
>> username=username
>> secret=secret
>> type=peer
>> host=ipaddress1
>> auth=md5
>>
>>
>> on the second box I got this
>>
>> this host is ipaddress2
>>
>> [incommingiax2]
>> username=username
>> type=user
>> secret=secret
>> host=dynamic
>> context=from-internal-custom
>> auth=md5
>>
>
>
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Subject: [Asterisk-Users] Passing DID to external number?

[Philippe Lindheimer]

For the DID's the easiest way for you to trasmit the incoming DID is to create custom extensions for the external numbers that access the external trunk directly. (e.g. they should NOT go to Loca/. or it will not retain the orignal CID in freepbx- which is effectively how it is being sent when you put x# in the ring list).As far as why it is only ringing one of your external numbers, I can only guess (with the limited information) that you may only have a single outbound channel available on your trunk so the second number is getting rejected? (Look in the log and see if it tells you that or if not provides additonal clues).p<[EMAIL PROTECTED]><[EMAIL PROTECTED]>From: "Brian McCarey" <[EMAIL PROTECTED]>To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
 Date: Thu, 22 Jun 2006 18:40:25 +0100Subject: [Asterisk-Users] Passing DID to external number? Hi,   We run a small  switchboard using Asterisk and Free PBX.   We have two main  extensions and two ring groups. The first ring group rings the two internal  extensions. If the internal extensions do not pick up the call after 15 seconds  then
 the second ring group kicks in which should ring the two internal  extensions plus two external numbers.   Firstly, how do I  pass the DID number of an incoming call to the external number so that the  external number sees the incoming number and not the voip dial out  number?   Secondly, when the  second ring group kicks in only one of the external numbers dials when both  internal extensions and both external numbers should ring according to the ring  group setting. Any ideals what's going wrong?   Kind  regards   Brian.  UK 
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Re: [Asterisk-Users] Echo and crackle

[Eric \"ManxPower\" Wieling]

Mojo with Horan & Company, LLC wrote:
I will agree that switching to the TDM card significantly helped my echo 
and sound quality, I would take a second to point out that interrupt 
sharing on your * server might cause crackling-like noises.  Try


lspci -vb
 and
cat /proc/interrupts

to see if you discern any hardware using the same irq the x101p is.


lspci does not show the IRQs *after* ACPI is enabled.  /proc/interrupts 
does.



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Re: [Asterisk-Users] Soekris net4801 and IAXy dhcp issue

[Eric \"ManxPower\" Wieling]

Juan Luis Moyano wrote:
Hi all, I have a Soekris net4801-50 board with OpenBSD 3.9 where I've 
configured a dhcp server and tested it with a regular PC connected 
directly via a crossover cable with success. The problem comes when I 
try to connect my IAXy device instead of the PC. I can see with 'tcpdump 
-nettti sis1' that the IAXy isn't sending any packets to the dhcp 
server. I thought my IAXy was bad but then I configured a second dhcp 
server with the exact same config file and the IAXy worked right out. So 
I don't have a clue of what could be happening. Please shed me some 
light on this issue. Thanks in advance.


I've suspected for a while that the IAXy does not use DHCP, but uses a 
similar, older protocol called BOOTP.  It could not hurt to try and 
enable BOOTP on your DHCP server (ISC DHCP server supports this).

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Re: [Asterisk-Users] when I press "transfer" -> blind -> 700 . The user is not able to hear what extension the call was parked on

[Eric \"ManxPower\" Wieling]

Matt wrote:

Interesting, I have #2 setup to do blind transgfers, and if I do a
#270 it tells me the number "seven one" and then hangs up on me and
the user is left on park 71.


Maybe Asterisk knows that doing a blind transfer to park a call is a 
silly and pointless thing to do and does a supervised transfer instead?



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Re: [Asterisk-Users] when I press "transfer" -> blind -> 700 . The user is not able to hear what extension the call was parked on

[Eric \"ManxPower\" Wieling]

sdgesa gaeharth wrote:

I have blindxfer  => #1 set  in features.Doesn't this means #1 is the same as  
transfer -> blind, correct? Both are blind transfers..
  
  Is so, why when I transfer using #1 do I hear what extension the call was parked at but not transfer -> blind?


#1 is, for whatever reason, doing a supervised transfer.

You do NOT get to hear the called party in a blind transfer.  If you 
hear the called party when you do a transfer then it is a supervised 
transfer, not a blind transfer.


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[Asterisk-Users] Quantum Voice Asterisk?

[Kenny Kant]

Are there any subscribers to Quantum voice on here who have gotten 
Asterisk to work with their service?  I found a small how to for 
[EMAIL PROTECTED] but have still not gotten it to work.  I am registering 
with their sip fine but am having a hard time passing calls to them.



Kenny


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[Asterisk-Users] problem - DSL line and Digium card

[Eric Hartley]

I was wondering if anyone has seen a similar problem...

I have a DSL line that doubles as a voice line to my Asterisk box.  When 
the Digium card answers that line, the DSL modem is disconnected until 
the line releases.  The phone line is split at the DSL modem then run to 
an FXO port.


I'm using one of the 4 port analog Digium cards (3 FXO/1 FXS).

Thanks for any suggestions.

Eric Hartley
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[Asterisk-Users] RTA, jitter, MOS et al over the internet

[Curt Shaffer]

I have been in the process of trying to troubleshoot a phone
system that is doing IAX trunking to a provider. The average RTA is 75ms with
spikes from time to time and jitter from time to time as well. My question is
this; How much can one trust this types of samples when going over the
internet? I mean who knows who is doing what kind of ICMP rate limiting or
dropping ICMP all together? What is a good measurement or troubleshooting step
for intermittent bad quality when dealing with links that you have no control
over or is that even relevant? Here is our setup:

 

All outbound calls are going out POTS unless that line is
congested. All inbound (even from the POTS as it is forwarded directly to an IP
DID) are over the IAX trunk. We are not seeing any issues on outbound calls. On
inbound, however, we are getting intermittent choppy voice, echo and cutting
out. This is heard by the person coming in, i.e when someone is calling they
hear these symptoms of the users on the asterisk server. From our side the
issue doesn’t seem to exist or it is so much less that it is really
irrelevant. 

 

As I have mentioned I have seen spikes of ping times and
times of jitter but this is recognized by tools utilizing ICMP so I don’t
know how much I can trust them. 

 

Thanks for the help!

 

Curt






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[Asterisk-Users] Cisco IP Phones - FYI

[Rich Adamson]

Just attended the Cisco Networkers conference and sat through a 
presentation addressing in part the Cisco IP phones. All was presented 
openly without the need for disclosures, etc.


The following is simply an FYI from the presentation, FWIW.

Existing IP Phones (7905/7912/7940/7960):
 Transport: UDP
 Call Signalling: rfc2833
 Security: Digest Auth
 Feature parity to SCCP: None

Sip Enhanced Phones (7911/7941/7961/7971/7970):
 Transport: UDP/TCP/TLS
 Call Setup: KPML and Dial Rules (KPML sends each dialed digit in a
   packet)
 Call Signaling: KPML, rfc2833
 Security: CAPF/CTL/TLS
 Feature parity to SCCP: Almost the same (will be later)

Notes:
1. All cisco phones with dual rj45's pass BPU packets (spanning tree) 
through internal switch ports. The switch is a non-mac-address-learning 
switch.
2. Cisco Discovery Protocol (CDP) is implemented in the phones and is 
used to discover the VLAN number (for rtp traffic), and to pass PoE 
power requirements to the closet switch. Per 802.3af, 3 power levels are 
possible. The 7960 requires 6.3 watts. If the switch doesn't support 
CDP, the switch will have to assume the highest power consumption.
3. Future releases of cisco's enhanced sip will provide the same 
features as the SCCP (skinny) phone images.
4. Early models of the 7912 phone had a hardware problem with QoS that 
can't be fixed. Later 7912's the problem was fixed.


Rich


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[Asterisk-Users] Voip* 300 minutes limit, credit expires

[Ronald Wiplinger]

Betamax makes our life more and more difficulty, hehehehehehe.

I found (today) that the free calls are limited to 300 minutes per week. 
It is good to know what "excess" use means!

That gives now also a challenge in the dialplan

Let's assume we have 5 accounts, each one has 300 minutes.
We use a variable as provider and get the right value of the not 
outmaxed provider into this variable.

How can we do that?

exten => _9011Z.,103,Dial(SIP/00${EXTEN:[EMAIL PROTECTED],30)
should be replaced with ${voipdiscount} 


and we need before a statement that finds the content of this variable.

Q: Does anybody know how to download the "recently used" statement?

I am interested how Voip* will react to the recently law change in 
Germany, where for mobile phone operator (and I assume that this law can 
be used for Voip* as well) a prepaid value may not expire anymore


Now lets look at Voip* pricing:
12 US$  per month for 300 x 4 minutes, with the expiration within 3 
months (13 weeks) ==> 1200/(13*300) = 0.3 cents in the BEST case!!! If 
you use more than 300, than you have to pay "at least" whatever that 
means in real numbers  1.2 cents.


It is getting more and more complicated, and that for pennies!!!
Unfortunately you cannot reach anybody there. I would like to have only 
ONE account and pay more for multiple use than this kind of tricks!


just some thoughts!

Ronald
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Re: [Asterisk-Users] SE Michigan asterisk users group

[btb]

On Jun 22, 2006, at 16.27, BerkHolz, Steven wrote:


I am thinking of getting an asterisk user group together for either SE
Michigan or just Metro-Detroit.

How much interest in asterisk in Michigan is there on this list?

I am already on the board of glimasoutheast, with is a group for
technology professionals. (very broad range)
It is a spin-off from Automation Alley, which is SE Michigan's version
of Silicone Valley.


there's at least one of us, albeit in sw michigan.  kind of a long  
drive to detroit.  ;)

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Re: [Asterisk-Users] Don't use CDRTool From AG-projescts

[Steve Totaro]

Alex Robar wrote:

Harry,

Fisrt ag-projects talk about is product like a gpl
software however they don't provide at least some
documentation for non commercial users .


Are they required to release docs? It's a nice bonus, but there's 
nothing forcing them to. If you want docs, you could always be helpful 
and write some yourself.


try to call them !!
i'll offer you some money .

You can not Call them for some advices ... 



And why should you be able to? Again, it's nice to have phone support, 
but in no way is it required. ViciDial doesn't have phone support, but 
people use it. FreePBX doesn't offer phone support, but it's pretty 
popular.


It's really a bad product don't waste your time to
setup it.
this enterprise must be  fogotten it's ag-projects .
it's not a reliable society ... more and more


Sounds to me like complaints from someone who couldn't figure out how 
to get the product setup to me.


If you have valid complaints then by all means, post it to the list as 
a warning... But even then, there's no need to openly bash a product 
or company you don't like. This isn't the forum for such things.


Alex

--
Alex Robar
[EMAIL PROTECTED]  

From their "README"

Disclaimer -- The information provided by CDRTool documentation 
is not always enough to successfully complete the installation and 
deployment of CDRTool. Most of the configuration tasks are related to 
setting up components outside CDRTool environment. Configuration of 
components like MySQL server, FreeRadius server, Cisco gateways, 
MediaProxy or SIP Express Router require a project management approach 
in place of a step by step installation procedure.


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Re: [Asterisk-Users] Snom 360 Passsword Issue

[Edward de Zeeuw]

I'll take a look first thing tomorrow and let you know what I find.  Thanks!
Edward

Colin Anderson wrote:
> In the Snom web management page under Advanced make sure "Challenge response
> on phone" is turned to OFF. This is a stupid feature to have on by default
> from the factory. 
>
> -Original Message-
> From: Edward de Zeeuw [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, June 21, 2006 11:54 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Snom 360 Passsword Issue
>
>
> I have multiple (20+) Snom 360 phones communicating with asterisk
> 1.2.7.1.  Almost regularly (daily) and in some cases ongoing 9every 10
> minutes the phones ask for password and id the account they are seeking
> the password for.  If I hit the X key the phone continues operating
> normally.  Has anyone else come across a similar issue?
>
> Edward
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[Asterisk-Users] Motherboard Selection For TE110P & TDM400P

[Joshua West]

Hey there,

What recommendations does anybody have as to motherboards known to work
well with the Digium TE110P T1 interface cards and the TDM400P analog
POTS interface cards?  We currently make use of the TE110P for our T1
(esf w/ b8zs - not a PRI) and have experienced many issues caused by our
PBX's motherboard.  Additionally, we built a server to become our new
PBX, but that machine runs into even worse problems.  It's been
determined through work with Digium that the Intel server class of
chipsets (7200's, 7300's, 7500's) don't do well with the demanding
amount of IRQ's their cards require.

Digium has suggested we use a motherboard more suited for desktop and
gaming use, as they are (supposedly) much better at handling demanding
hardware.  They specifically recommended motherboards with chipsets such
as Intel 865, 945, etc - but absolutely avoid all from the Intel 915 series.

So, those of you with TE110P and TDM400P cards, what motherboard are you
using in your PBX?  How's it working?  No issues at all?  'zttest'
scores?  Any crackling or echo?  Noticed any differences when the system
is under high load?  Perhaps even you've gone so far as to run
'patlooptest' with a T1 loopback cable connected; care to share your
results?

Just for reference, motherboard's we've had issues with:
- Abit IC7-G
- Supermicro P8SCT

Thanks.

-- 
Joshua West
Linux Infrastructure Engineer
Boston Engineering Corporation
http://www.boston-engineering.com


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Re: [Asterisk-Users] Soekris net4801 and IAXy dhcp issue

[Leo Ann Boon]

See my comments in-line.
Juan Luis Moyano wrote:

Hi all, I have a Soekris net4801-50 board with OpenBSD 3.9 where I've 
configured a dhcp server and tested it with a regular PC connected 
directly via a crossover cable with success. The problem comes when I 
try to connect my IAXy device instead of the PC. I can see with 
'tcpdump -nettti sis1' that the IAXy isn't sending any packets to the 
dhcp server. I thought my IAXy was bad but then I configured a second 
dhcp server with the exact same config file and the IAXy worked right 
out. So I don't have a clue of what could be happening. Please shed me 
some light on this issue. Thanks in advance.


Juan Luis Moyano

#cat /etc/dhcpd.conf
shared-network LOCAL-NET {

   option  domain-name "b-fon.com.ar";
   option  domain-name-servers 10.32.2.254, 200.69.193.1, 
200.69.193.2;


   subnet 10.32.2.0 netmask 255.255.255.0 {
   option routers 10.32.2.254;
   range 10.32.2.32 10.32.2.64;


Have you tried:
  range dynamic-bootp 10.32.2.32 10.32.2.64;

IIRC, IAXY requires bootp.

Leo


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[Asterisk-Users] Re: Quality monitoring

[Michael Wallette]

I haven't tried the scripts yet, but the book "*VoIP Hacks : Tips & 
Tools for Internet Telephony (Hacks)" 
(http://www.amazon.com/exec/obidos/tg/detail/-/0596101333/qid=1151018442/sr=2-1/ref=pd_bbs_b_2_1/103-6123898-6212653?v=glance&s=books)


has a collection of perl scripts for monitoring network quality, 
including, IIRC, jitter. I don't know how well the scripts will work 
with Nagios, but it might be worth checking out.


--Mike


*

Message: 13
Date: Thu, 22 Jun 2006 12:53:50 -0400
From: "Curt Shaffer" <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] Quality monitoring
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"

Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset="us-ascii"

Does anyone out there have a recommendation for tools that will monitor the
quality of VoIP systems? I am looking for jitter and MOS monitoring. I have
a custom Nagios plugin that is alerting me if the jitter jumps out of a 20ms
but I am looking for a little more detail. I would not be against writing
something in Perl for Nagios to do but I don't really know where to start on
measuring jitter other than with ICMP pulls and really don't know where to
start with doing MOS. 

 


Any ideas?

 


Thanks

 


Curt


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Re: [Asterisk-Users] Re: Avaya phone 4610sw message waiting indicator and other settings

[Tom Lynn]

Nope.  Let me know if you do.  I've suspended my efforts until I see a new version of firmware available on the Avaya web site.
On 6/21/06, Erick Perez <[EMAIL PROTECTED]> wrote:
Thanks for your comments Tom. Indeed the MWI and the programmablebuttons are the only things that do not work for me. Besides that, the
phone is great and the audio quality is superb.Did you managed somehow to make the MWI work?Will keep searching the net, the 4602 page is somehow poor on the documentation.On 6/21/06, Tom Lynn <
[EMAIL PROTECTED]> wrote:> Well, I wouldn't say nobody.  I do and I've corresponded with a few people> that do. There's a page on voip-info.org
 dedicated to the Avaya 4602> telephone and SIP (I'm hoping I'm not the only reader of that page).  When> I've used my Avaya phone in conference (FWD CoffeeHouse), I've had people> sincerely compliment me on the quality of sound with my phone.
>> But..>> Avaya has a few things working against it within the context of Asterisk:>> * MWI just doesn't work (If you insist on trying it, get ready for your> phone to lose it's registration with * every hour or so)
> * Dial strings beginning with * character appear to go nowhere with these> phones> * They're perceived as rather expen$ive> * As a company, they're simply not focused on * since it doesn't help sell
> any of their other product.  They prefer selling things that drive> maintenance contract revenue and, let's face it, the phone is the commodity> appliance that connects to *.  Even within the enterprise space, very few
> carry maintenance on their telephone sets anymore.>> Funny anectdote:  Avaya loves showing Cisco 79xx phones with a SIP load> registered to their PBX systems with a Powered By Avaya background.  They
> claim that, unlike Cisco, they will accept third party SIP clients> registering to their system.  However, they really don't provide any kind of> support for their phones used with a system other than their own.  My Mom
> used to call that the Pot calling the Kettle Black.>> Good phone, great sound, just no support and a bit wonky on the features.>> My 2 cents.>
>> On 6/21/06, Erick Perez < [EMAIL PROTECTED]> wrote:> >> nobody uses avaya phones with asterisk?>> On 6/20/06, Erick Perez < 
[EMAIL PROTECTED]> wrote:> > Hi, I setup my tftp to send SIP configurations (the bin files) to the> > avaya phone. When it finished loading and rebooting it asked for the> > extension and the password and the asterisk ip address. I had to input
> > that manually and is now working perfectly with asterisk.> >> > what is the format of the text files to make this phone load the> > asterisk ip, extension number, codec used, password as well as to
> > configure message waiting indicator and maybe modify some of the> > buttons (such as just pressing one of the available programmable>  > buttons to access voicemail). I have 10 more of these phones and i
> > want to do provisioning automatically.> >> > in the 46xxsettings.txt file there are no such parameters> >> >> > thanks,> >> >> > --
> >> > > Erick Perez> > Panama Sistemas> > Integradores de Telefonia IP y Soluciones Para Centros de Datos> > Panama, Republica de Panama
> > Cel Panama. +(507) 6694-4780> >> >  --> 
> Erick Perez> Panama Sistemas> Integradores de Telefonia IP y Soluciones Para Centros de Datos> Panama, Republica de Panama> Cel Panama. +(507) 6694-4780> 
> ___> --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>>>--Erick PerezPanama SistemasIntegradores de Telefonia IP y Soluciones Para Centros de Datos
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[Asterisk-Users] TE405P Dropping Calls. !! Got I-frame while link state 0

[Brian Stuart]

Hello we have an * setup using a TE405P with a crossover to a 
Dialogic PRI card. The machine with the Dialogic card is an automated 
call generator. The asterisk machine is being used as a test platform 
to answer the incoming calls play an audio file then record the 
incoming audio. This set up has been working nicely when the calls 
were coming in on a SIP channel.
However when call come in on the PRI it seems that * randomly drops 
all channels. Some times every 30 seconds sometimes every 10 minutes. 
The * CLI> will get the errors shown below. I have seen similar 
postings about this error but found no explanation or solution.


The asterisk machine has no shared interrupts.

Could this be a cabling issue?

Write to 36 failed: Unknown error 500
Short write: 0/15 (Broken pipe)
-- Executing Hangup("Zap/2-1", "") in new stack
!! Got I-frame while link state 0
-- Hungup 'Zap/2-1'
!! Got S-frame while link down
!! Frame got rejected!
  == Primary D-Channel on span 1 up
-- Executing Hangup("Zap/1-1", "") in new stack



Brian Stuart
Edulink Systems, Inc.
1(888)338-7177 ext. 219
http://www.edulinksys.com/ 



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Re: [Asterisk-Users] Don't use CDRTool From AG-projescts

[Alex Robar]

Harry, Fisrt ag-projects talk about is product like a gplsoftware however they don't provide at least some
documentation for non commercial users .Are they required to release docs? It's a nice bonus, but there's nothing forcing them to. If you want docs, you could always be helpful and write some yourself.
try to call them !!i'll offer you some money .You can not Call them for some advices ...
And why should you be able to? Again, it's nice to have phone support, but in no way is it required. ViciDial doesn't have phone support, but people use it. FreePBX doesn't offer phone support, but it's pretty popular.
It's really a bad product don't waste your time tosetup it.this enterprise must be  fogotten it's ag-projects .
it's not a reliable society ... more and moreSounds to me like complaints from someone who couldn't figure out how to get the product setup to me.If you have valid complaints then by all means, post it to the list as a warning... But even then, there's no need to openly bash a product or company you don't like. This isn't the forum for such things.
Alex-- Alex Robar[EMAIL PROTECTED]
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Re: [Asterisk-Users] Don't use CDRTool From AG-projescts

[Jean-Michel Hiver]

[EMAIL PROTECTED] a écrit :


hello to all,

I advice you to not use CDRtool from ag-projects :
Fisrt ag-projects talk about is product like a gpl
software however they don't provide at least some
documentation for non commercial users .
 


And they don't have to.

I have not dealt with them before, nor do I have any connections with 
them, yet I find your libellous accusations completely unacceptable.


Regards,
Jean-Michel.

--
Jean-Michel Hiver - http://ykoz.net/
Découvrez la Réunion des Technologies IP & Telecom
TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

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[Asterisk-Users] Routing inboud from ISDN to second * server.

[Thomas Laurids Pedersen]

Hi All,

I have setup 2 asterisk servers using AAH 2.8. I have configured a IAX2
trunk between the 2 servers using the guide on dumbme. Trunk is not using
register string and no authentication.

In my dial plan I have 7XX numers on server B and 6xx numbers on Server A.

Calls from my SNOM phones are ok between the extensions on the 2 servers.

In server A I have a eicon 4 port BRI card connected. Calls from outside to
extensions on server A works fine. (Inbound routes are configured for
"local" extensions).

Now how do I configure so that extensions on server B can be called from
the ISDN connections ? like this ISDN -> ServerA   ->(IAX2)-> ServerB ->
extension

Server A does not have direct network access to the SNOM phone.

>From my reading I think that I have to make an inbound route or asterisk
will not treat the call at all. This is also what I see if I debug capi.

I have tried to make an inbound route and custom extension for all 7XX
extentions and then entering the dial string IAX2/astB/7255 for extention
725. Also tried the dial string IAX2/[EMAIL PROTECTED] with no luck.

How do I do this ?

Pls help anyone.

Regards

Thomas

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[Asterisk-Users] PCI or MiniPCI Hardware DSP for G.729, G.723.1 and/or GSM

[Roland Zagler]

Hi to all,

we are searching for a hardware based DSP solution for use
with Asterisk based on PCI or MiniPCI to reduce main processor
load and to use embedded boards with Digium E1/T1 cards like
TE410P.

does anyone know about any manufactorer of those cards or someone
who is able to develop/build such cards?

Specifications:
PCI or MiniPCI
up to 120 concurrent transcodings
Codecs: G.729/G.729A or G.723.1 or GSM or combinations of them

Thank you in advance,
Roland Zagler
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[Asterisk-Users] Asterisk Users Group

[David Evennou \(Data Masters, Inc.\)]

Hi Steven,
 
Depending on the location, I may be 
interested.
Thanks,David
 
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Re: *** Spam *** [Asterisk-Users] Don't use CDRTool From AG-projescts

[Steve Totaro]

Harry has been on my blacklist for some time now.

Christian Stredicke wrote:

This post cannot be left without comment. People who don't know you or Adrian 
might get a wrong impression.

I know Adrian quite well and know that he is one of the real experts in this industry and he and his stuff does not deserve such a treatment. 


I would recommend that you change your attitude. It seems like you did not get 
what you want (for free) and you complain like a small child. An apology would 
be appropriate.

CS 

  

-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
[EMAIL PROTECTED]

Sent: Thursday, June 22, 2006 3:16 PM
To: Asterisk-users@lists.digium.com
Cc: [EMAIL PROTECTED]
Subject: *** Spam *** [Asterisk-Users] Don't use CDRTool From 
AG-projescts


hello to all,

I advice you to not use CDRtool from ag-projects :
Fisrt ag-projects talk about is product like a gpl software 
however they don't provide at least some documentation for 
non commercial users .


try to call them !!
i'll offer you some money .

You can not Call them for some advices ...

It's really a bad product don't waste your time to setup it. 
this enterprise must be  fogotten it's ag-projects .
it's not a reliable society ... more and more 

projects around open(ser) asterisk and more are offered good 
unliked projects cdrtool please do not use ag-projects products !



Harry is not Harry Potter !

Regards

  












  


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RE: [Asterisk-Users] Thoughts on building a Voicemail only Asteriskserver?

[William Boehlke]

A 2u server with a single processor and 1GB of memory will 
more than suffice. RAID for the voicemail store. 
 
Considering the application, we'll be happy to set it up 
for you once you have the hardware. Contact me off list.
 
William Boehlke
Signate
 
 
 


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Christopher 
AloiSent: Thursday, June 22, 2006 2:17 PMTo: Asterisk 
Users Mailing List - Non-Commercial DiscussionSubject: 
[Asterisk-Users] Thoughts on building a Voicemail only 
Asteriskserver?
Hello List -I've done some reading on voip-info regarding 
hardware requirements for an Asterisk server; but I haven't been able to find 
anyone doing what we plan to, so I am hoping you can assist.We are 
looking to provide a voice mail only Asterisk solution for approx. 100 homeless 
people, a customer of ours is planning to provide the service. The 
Asterisk service will reside in our data center which will provide the 
TDM->SIP GW so the Asterisk will receive all it's calls via SIP.A 
rough overview of what I think we will need:- A non-redundant server 
running Asterisk -- -- The Asterisk build will have a very simple dial 
plan-- -- -- Two inbound DID's (one for checking vmail and one for leaving 
voice mail for an extension)-- -- -- A management interface for the voice 
mail boxes, so I will need to run Apache - A disk array (either local 
RAID or external NAS) to house the voice mail storage.-- -- The voice mail 
system will allow 30MB of storage per user, so 30MBx100users=3GB-- -- I'd 
like the 3GB of storage to be in either in the RAID or dumped onto an NFS or NAS 
Does anyone have any recommendations on a server that might fit the bill 
above? Or experience running a similar application?Just looking for 
some thoughts on RAM, Processor speed, Disc etc...Thanks in advance. 
-Chris
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[Asterisk-Users] Troncal SIP

[Aldo Alexander Leyva Alvarado]

helloI have the following configuration of one troncal SIP,in order to remove calls through a supplier from voice ,the supplier is in the USA , they suggest to use the codecs g729 , which we got and installed it , I want to know  the way we could optimize the comunication with is listening in a dificult way,not very clear , which opcion of configuration could use in troncales SIP in order to get the better quality.
this is my present configuration:[voip_sip]disallow=allallow=g729qualify=yestos=184;allow=gsm;allow=ulaw;allow=alaw;allow=all;allow=ulaw;allow=alaw;disallow=all
host=
XXX.XXX.XXX.XXXtype=peerThanks for your answers.Aldo Leyva
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[Asterisk-Users] Thoughts on building a Voicemail only Asterisk server?

[Christopher Aloi]

Hello List -I've done some reading on voip-info regarding hardware requirements for an Asterisk server; but I haven't been able to find anyone doing what we plan to, so I am hoping you can assist.We are looking to provide a voice mail only Asterisk solution for approx. 100 homeless people, a customer of ours is planning to provide the service.
The Asterisk service will reside in our data center which will provide the TDM->SIP GW so the Asterisk will receive all it's calls via SIP.A rough overview of what I think we will need:- A non-redundant server running Asterisk
-- -- The Asterisk build will have a very simple dial plan-- -- -- Two inbound DID's (one for checking vmail and one for leaving voice mail for an extension)-- -- -- A management interface for the voice mail boxes, so I will need to run Apache
- A disk array (either local RAID or external NAS) to house the voice mail storage.-- -- The voice mail system will allow 30MB of storage per user, so 30MBx100users=3GB-- -- I'd like the 3GB of storage to be in either in the RAID or dumped onto an NFS or NAS
Does anyone have any recommendations on a server that might fit the bill above? Or experience running a similar application?Just looking for some thoughts on RAM, Processor speed, Disc etc...Thanks in advance.
-Chris
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Re: [Asterisk-Users] iax2 registration problems

[Thomas Kenyon]

Bartosz Wegrzyn - asterisk wrote:
> On the asterisk1 I got this:
>
> register => username:[EMAIL PROTECTED]
>
>   
Try changing this to :

register => incommingiax2:[EMAIL PROTECTED]

There is a good reason I say this, but I'm too tired to remember what it is.
> [eop]
> username=username
> secret=secret
> type=peer
> host=ipaddress1
> auth=md5
>
>
> on the second box I got this
>
> this host is ipaddress2
>
> [incommingiax2]
> username=username
> type=user
> secret=secret
> host=dynamic
> context=from-internal-custom
> auth=md5
>   


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Re: [Asterisk-Users] when I press "transfer" -> blind -> 700 . The user is not able to hear what extension the call was parked on

[Matt]

Interesting, I have #2 setup to do blind transgfers, and if I do a
#270 it tells me the number "seven one" and then hangs up on me and
the user is left on park 71.

On 6/22/06, sdgesa gaeharth <[EMAIL PROTECTED]> wrote:

I have blindxfer => #1 set  in features.Doesn't this means #1 is the same as
transfer -> blind, correct? Both are blind transfers..

 Is so, why when I transfer using #1 do I hear what extension the call was
parked at but not transfer -> blind?



James Texter <[EMAIL PROTECTED]> wrote:
 This is the way blind transfers work.  The transferring party doesn't get
to hear anything.  For call parking, you have no choice but to use
supervised transfer if you want the user to hear the parking space.  If it
worked before, it must have been dumb luck with the timing.


 On 6/22/06 10:24 AM, "sdgesa gaeharth" <[EMAIL PROTECTED]> wrote:


I am using Polycom 501s with asterisk 1.2.4.

   When transfering to call parking wih "#1" -> 700 the user is able to
hear asterisk tell him what extension the call was parked on.   However,
when I press "transfer" -> blind -> 700 . The user is  not able to hear what
extension the call was parked on. It seems like  the polycom is hanging up
before asterisk is able to finish telling the  user the extension. I can not
tell if this is a problem with the phone,  asterisk, transfers or call
parking. This problem just started  happening a few weeks ago.  Before then
, blind transfer worked  fine. It must be a config issue somewhere

   using "#1" -> 700:
   -- Started music on hold, class 'default', on channel 'Zap/1-1'
 == Parked Zap/1-1 on 701. Will timeout back to extension [incoming] s,
1 in 45 seconds
   -- Added extension '701' priority 1 to parkedcalls
   -- Playing 'digits/7' (language 'en')
-- Executing ParkedCall("SIP/1000-300e", "701") in new stack
   -- Stopped music on hold on Zap/1-1
   -- Channel SIP/1000-300e connected to parked call 701
   -- Hungup 'Zap/1-1'

   using "transfer" -> blind -> 700
   -- Started music on hold, class 'default', on channel 'Zap/1-1'
 == Parked Zap/1-1 on 701. Will timeout back to extension [incoming] s,
1 in 45 seconds
   -- Added extension '701' priority 1 to parkedcalls
   -- Playing 'digits/7' (language 'en')
   -- Playing 'digits/0' (language 'en')
   -- Playing 'digits/1' (language 'en')
 == Spawn extension (local-access, 97037551131, 1) exited non-zero on
'SIP/1000-d779'
   -- Executing ParkedCall("SIP/1000-5f5a", "701") in new stack
   -- Stopped music on hold on Zap/1-1
   -- Channel SIP/1000-5f5a connected to parked call 701
   -- Started music on hold, class 'default', on channel 'Zap/1-1'
   -- Stopped music on hold on Zap/1-1


   Even though the user can not hear the extension the call was parked on,
the call can be retrieved by guessing. Which I am assumming means the  call
was successfully parked.

   Digit map:

[2-9]11|0T|011xxx.T|[0-1][2-9]x|[2-9]x|[2-9]xxxT|1xxxT

   extensions.conf:
   [general]
   static=yes
   writeprotect=no
   autofallthrough=yes
   clearglobalvars=no
   priorityjumping=no

   [globals]
   ATTENDANT=SIP/1006&SIP/1002&SIP/1011&SIP/1009
   OUTBOUNDTRUNK=ZAP/g1

   [meetme-ext]
   exten => 600,1,MeetMe(1234|Mp|98765)

   [extentions]
   include => parkedcalls
   include => meetme-ext
   include => direct-to-voicemail
   exten => _10XX,1,Dial(SIP/${EXTEN},20,t)
   exten => _10XX,n,Answer
   exten => _10XX,n,VoiceMail([EMAIL PROTECTED])
   exten => _10XX,n,Hangup()

   [voicemail]
   exten => _910XX,1,Wait(1)
   exten => _910XX,n,VoiceMailMain(${EXTEN:[EMAIL PROTECTED])

   [direct-to-voicemail]
   exten => _810XX,1,VoiceMail(u${EXTEN:[EMAIL PROTECTED])
   exten => _810XX,n,Hangup()

   [local]
   include => extentions
   include => voicemail

   [incoming]
   exten => s,1,Answer
   exten => s,n,Wait(2)
   exten => s,n,Set(TIMEOUT(response)=15)
   exten => s,n,Background(intro)
   exten => s,n,WaitExten()
   exten => s,n,Playback(vm-goodbye)
   exten => s,n,Hangup()
   exten => 0,1,Dial(${ATTENDANT},20,t)
   exten => 0,n,Playback(vm-nobodyavail)
   exten => 0,n,Hangup()
   exten => 1,1,Directory(voicemail,extentions,f)
   exten => 2,1,Directory(voicemail,extentions)
   include => meetme-ext
   include => extentions
   exten => i,1,Playback(pbx-invalid)
   exten => i,2,Goto(incoming,s,1)
   exten => t,1,Playback(vm-goodbye)
   exten => t,2,Hangup()

   [outbound]
   ignorepat => 9
   include => parkedcalls
   exten =>
_9XX,1,Dial(${OUTBOUNDTRUNK}/ww${EXTEN:1},30,T)
   exten => _9XX,2,Congestion()
   exten => _9XX,102,Congestion()
   exten => _91900NXX,1,Congestion()
   exten => _91976NXX,1,Congestion()
   exten =>
_91[123456789]XXNXX,1,Dial(${OUTBOUNDTRUNK}/ww${EXTEN:1},30,T)
   exten => _91[123456789]XXNXX,2,Congestion()
   exten => _91[123456789]XXNXX,102,Congestion()
   exten => 9911,1,Dial(${OUTBOUNDTRUNK}/ww911)
   exten => 9411,1,Dial(${OUTBOUN

Re: [Asterisk-Users] Re: Can I enter an extension to dial whilevoicemail is playing?

[John Klimek]

Ahhh!  That fixed it!!!

However, it seems like I need to keep the Answer() in there.  This
causes incoming callers to here a stuttered ringing in the beginning.
Is there a way to fix/remove this?  I'm guessing there isn't because
Asterisk needs to answer to monitor the line for number presses...

However, I'm concerned this stuttered ringing will cause people to
call in to think we have a problem with our phones or something...


On 6/22/06, Tim Sharp <[EMAIL PROTECTED]> wrote:

The options are not seperated by commas.
 exten => s,1,Dial(SIP/50,23,r,d)
should be
 exten => s,1,Dial(SIP/50,23,rd)

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John Klimek
Sent: Thursday, June 22, 2006 2:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: Can I enter an extension to dial
whilevoicemail is playing?


Any idea why it wouldn't work in my dial plan?

On 6/22/06, Peter Antonacci <[EMAIL PROTECTED]> wrote:
> d: This flag trumps the 'H' flag and intercepts any dtmf while waiting for
> the call to be answered and returns that value on the spot. This allows you
> to dial a 1-digit exit extension while waiting for the call to be answered -
> see also
>
>
> On 6/22/06, John Klimek <[EMAIL PROTECTED]> wrote:
> > Anybody have any more information on this Dial() "d" option for incoming
> calls?
> >
> > On 6/19/06, John Klimek < [EMAIL PROTECTED]> wrote:
> > > Thanks for the information...
> > >
> > > After doing some reading it looks like I can use the "d" option with
> > > the Dial() command to be able to enter a 1-digit extension while the
> > > other extension is ringing, but this doesn't seem to be working for me
> > > either...
> > >
> > > Here is my new config:
> > >
> > > exten => s,1,Dial(SIP/50,23,r,d)
> > > exten => s,2,VoiceMail( [EMAIL PROTECTED])
> > > exten => s,3,Playback(vm-goodbye)
> > > exten => s,4,Hangup
> > >
> > > exten => 1,1,SayDigits(1)
> > > exten => 2,1,SayDigits(2)
> > > exten => 10,1,SayDigits(10)
> > >
> > > However, when my phone is ringing (eg. extension 50), I try entering
> > > "1" or "2" (to be forwarded via the Dial "d" option), but it doesn't
> > > do anything.
> > >
> > > What am I doing wrong?
> > >
> > > I like your solution above, but if I use that I'll need to wait 23
> > > seconds for Dial() to timeout before I can do anything.  I'd like to
> > > be immediately able to enter an extension (if possible, which maybe
> > > it's not...)
> > >
> > > On 6/19/06, Leah Newmark <[EMAIL PROTECTED]> wrote:
> > > > Using the Background command, you will be able to play the voicemail
> > > > while still being allowed to enter digits.
> > > >
> > > > exten => s,1,Wait(2)
> > > > exten =>
> 108,2,Background(voicemail/default/108/unavail)
> > > >
> > > >
> > > > exten => s,1,Dial(SIP/50,23,r)
> > > > exten =>
> s,2,Background(/voicemail/default/50/unavail) ;or whatever
> the
> > > > soundfile is called
> > > > exten => s,3,Voicemail(s50) ;s will skip the greeting and just go to
> the
> > > > beep
> > > > exten => s,4,Playback(vm-goodbye)
> > > > exten => s,5,Hangup
> > > >
> > > > You can then put
> > > > exten => 1, Dial(sip/me)
> > > > exten => 2, Dial(sip/her)
> > > > or whatever your dial statements look like.
> > > >
> > > > Leah Newmark
> > > > Capalon VoIP
> > > >
> > > >
> > > > [EMAIL PROTECTED] wrote:
> > > >
> > > > Message: 9
> > > > Date: Mon, 19 Jun 2006 14:18:22 -0400
> > > > From: "John Klimek" <[EMAIL PROTECTED]>
> > > > Subject: [Asterisk-Users] Can I enter an extension to dial while
> > > > voicemail   is playing?
> > > > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> > > > 
> > > > Message-ID:
> > > >
> <[EMAIL PROTECTED]>
> > > > Content-Type: text/plain; charset=ISO-8859-1; format=flowed
> > > >
> > > > I have a very, very simple Asterisk setup in my house.  I have a
> > > > Sipura 3000 with a PSTN line connected and one analog phone connected.
> > > >
> > > > The [incoming] context looks like this:
> > > >
> > > > exten => s,1,Dial(SIP/50,23,r)
> > > > exten => s,2,VoiceMail([EMAIL PROTECTED])
> > > > exten => s,3,Playback(vm-goodbye)
> > > > exten => s,4,Hangup
> > > >
> > > > As you can see, when somebody calls in if I don't answer in 23 seconds
> > > > then they are forwarded to my voicemail.
> > > >
> > > > How can I make it so I can call an enter extensions either while the
> > > > phone is ringing or while the voicemail message is playing?  I want
> > > > the system to be as seemless as possible so the wife is happy =)
> > > >
> > > > Right now it works great because my Sipura 3000 forwards to call to
> > > > Asterisk and Asterisk rings my analog phone, but the incoming caller
> > > > hears a steady dial-tone the whole time.  I wouldn't want that to
> > > > change.  (so the caller isn't wondering what is going on)
> > > >
> > > > Any help is appriciated  :)
> > > >
> > > > ___
> > > > --Bandwidth and 

RE: [Asterisk-Users] uniden uip 200 phones lockup but rare - anyo ne seen this

[Nathan C. Smith]

I have several too and I also see this problem on occasion.  Like you say,
it is fairly rare and I can't pinpoint a cause or tell if it is a symptom of
something else.  

I think I wrote to tech support about it but never heard anything.  I'm
wondering how long they will continue to support the phone.

-Nate

> -Original Message-
> From: Jerry Geis [mailto:[EMAIL PROTECTED] 
> Sent: Wednesday, June 21, 2006 3:28 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] uniden uip 200 phones lockup but 
> rare - anyone seen this
> 
> 
> has anyone had any problems with uip200 phones locking up?
> I have around 10 of them and once in a great while (perhaps 
> once a month) the phone locks up.
> 
> Anyone else seen this problem?
> 
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[Asterisk-Users] Dell PowerEdge 1650

[Sean Cook]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
 
Anyone have a 1650 running successfully in production mode with 2-4
PRI's?  I want to make sure I don't have a motherboard compatibility
problem before I buy one of these.  We are going to be using a Digium
TE210P to start off with and probably moving to the TE411P down the
road aways?

thanks,

Sean
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.3 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
 
iD8DBQFEmwOb1Kolm8VQlAURAtBOAJsFdirbMhilGmcKd07+tORXxwZLKgCfQrjs
3hoZGLFllzW6xLrpRuuxjMk=
=3bOb
-END PGP SIGNATURE-

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Re: [Asterisk-Users] sip to h323 ... direct RTP?

[Jeremy McNamara]

Kevin P. Fleming wrote:

I believe this is incorrect; all the RTP-using channel drivers supply 
'ast_rtp_bridge' as their native bridge method, so assuming they also implement 
the 'set_rtp_peer' method, then an RTP native bridge between dissimilar 
channels should work fine. If the channel driver(s) also support sending the 
RTP peer address to the endpoint (as chan_sip does with reinvite), then a 
direct media path should also be possible.




The problem is 're-inviting' in H.323-jive is very much a non-trivial task.


Jeremy McNamara
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[Asterisk-Users] Sip error messages

[Neil Bullock]

Please can anyone advise what these messages mean?

Jun 22 21:38:07 ERROR[2785]: chan_sip.c:11323 sipsock_read: We could NOT
get the channel lock for SIP/213.xxx.5.xxx-0816e1b8!
Jun 22 21:38:07 ERROR[2785]: chan_sip.c:11324 sipsock_read: SIP MESSAGE
JUST IGNORED: ACK
Jun 22 21:38:07 ERROR[2785]: chan_sip.c:11325 sipsock_read: BAD! BAD! BAD!

Asterisk 1.2.9.1

and most importantly whether I should worry about them

Cheers,

Neil



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[Asterisk-Users] SE Michigan asterisk users group

[BerkHolz, Steven]

I am thinking of getting an asterisk user group together for either SE
Michigan or just Metro-Detroit.

How much interest in asterisk in Michigan is there on this list?

I am already on the board of glimasoutheast, with is a group for
technology professionals. (very broad range)
It is a spin-off from Automation Alley, which is SE Michigan's version
of Silicone Valley.

-- 
Steven

http://www.glimasoutheast.org 

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RE: [Asterisk-Users] freepbx centos 4 install script?

[Michael Collins]

> Has anyone created a script that will download and install all of the
> freepbx prerequisites in the INSTALL file automatically on a Centos 4
box?
> 
In a manner of speaking the trixbox guys have.  Have you ever seen that
(or Asterisk @ Home)?  There is a script, install.sh, that installs a
bunch of stuff.  The FreePBX pre-reqs are mixed in with everything else,
but if you see the "yum -y install" line you'll see a ton of RPMs that
get installed, some of which are your pre-reqs.  Later on in the script
it has the actual installation - make, make install, etc.

Check it out: trixbox.org

-MC
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Re: [Asterisk-Users] Don't use CDRTool From AG-projescts

[Jens Vagelpohl]

On 22 Jun 2006, at 22:11, Christian Stredicke wrote:

This post cannot be left without comment. People who don't know you  
or Adrian might get a wrong impression.


Honestly, I think it can. That post tells you everything you need to  
know about the camplaining party ;)


jens

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[Asterisk-Users] How to set overlap dial timeout in bristuff zaptel?

[Benoit Panizzon]

Hi all

There seam to be a very short timeout waiting for digits being dialed. (about 
6 seconds).

Is there a way to increase that time? I have a phone with integrated address 
book and my fingers are just not fast enough to open the menue, select an 
entry and hit 'dial'.

-Benoit-
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[Asterisk-Users] iax2 registration problems

[Bartosz Wegrzyn - asterisk]

On the asterisk1 I got this:

register => username:[EMAIL PROTECTED]

[eop]
username=username
secret=secret
type=peer
host=ipaddress1
auth=md5


on the second box I got this

this host is ipaddress2

[incommingiax2]
username=username
type=user
secret=secret
host=dynamic
context=from-internal-custom
auth=md5


on first host 1 am getting:

Jun 22 14:42:10 NOTICE[2398]: chan_iax2.c:7411 socket_read: Registration
of 'proxy1' rejected: 'Registration Refused' from: 'ipaddress2'

if I enable debuging I got this:

Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ
Timestamp: 4ms SCall: 6 DCall: 0 [ipaddress2:4569]
USERNAME : username
REFRESH : 60

Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGREJ
Timestamp: 00017ms SCall: 1 DCall: 6 [ipaddress2:4569]
CAUSE : Registration Refused
CAUSE CODE : 29

Jun 22 14:43:50 NOTICE[2398]: chan_iax2.c:7411 socket_read: Registration
of 'username' rejected: 'Registration Refused' from: 'ipaddress2'
Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK
Timestamp: 00017ms SCall: 6 DCall: 1 [ipaddress2:4569]


Why the box is refusing the registration.
That box is on DMZ vlan with local ip
On the pix firewall public ip address2 is mapped to that box ip

debug on the second box

Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ
Timestamp: 7ms SCall: 2 DCall: 0 [ipaddress1:4569]
USERNAME : username REFRESH : 60

Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGREJ
Timestamp: 1ms SCall: 3 DCall: 2 [ipaddress1:4569]
CAUSE : Registration Refused
CAUSE CODE : 29

Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK
Timestamp: 1ms SCall: 2 DCall: 3 [ipaddress1:4569]


any ideas what could be wrong?

Thanks

Bart
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RE: *** Spam *** [Asterisk-Users] Don't use CDRTool From AG-projescts

[Christian Stredicke]

This post cannot be left without comment. People who don't know you or Adrian 
might get a wrong impression.

I know Adrian quite well and know that he is one of the real experts in this 
industry and he and his stuff does not deserve such a treatment. 

I would recommend that you change your attitude. It seems like you did not get 
what you want (for free) and you complain like a small child. An apology would 
be appropriate.

CS 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> [EMAIL PROTECTED]
> Sent: Thursday, June 22, 2006 3:16 PM
> To: Asterisk-users@lists.digium.com
> Cc: [EMAIL PROTECTED]
> Subject: *** Spam *** [Asterisk-Users] Don't use CDRTool From 
> AG-projescts
> 
> hello to all,
> 
> I advice you to not use CDRtool from ag-projects :
> Fisrt ag-projects talk about is product like a gpl software 
> however they don't provide at least some documentation for 
> non commercial users .
> 
> try to call them !!
> i'll offer you some money .
> 
> You can not Call them for some advices ...
> 
> It's really a bad product don't waste your time to setup it. 
> this enterprise must be  fogotten it's ag-projects .
> it's not a reliable society ... more and more 
> 
> projects around open(ser) asterisk and more are offered good 
> unliked projects cdrtool please do not use ag-projects products !
> 
> 
> Harry is not Harry Potter !
> 
> Regards
> 
>   
> 
> 
> 
> 
> 
> 
> 
>   
> 
>   
>   
> __
> _
> Yahoo! Mail réinvente le mail ! Découvrez le nouveau Yahoo! 
> Mail et son interface révolutionnaire.
> http://fr.mail.yahoo.com
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>http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
> 
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Re: [Asterisk-Users] Showing Current Calls

[Joshua Colp]

The first line has (None) as the location because a PBX is not running on it
as it was created by the channel below it using the Dial application. As for
the BridgedCall(SIP/2944079-e7f2) part that's to indicate it's bridged to
that channel. Anything could have been put in that space since no PBX is
running on it, and thus no application... But BridgedCall(channel name) was
done so you would know what it is bridged to.


On 6/22/06 4:50 PM, "Douglas Garstang" <[EMAIL PROTECTED]> wrote:

> Using this as an example:
> 
> hestia*CLI> show channels
> Channel  Location State   Application(Data)
> SIP/2944093-f9e2 (None)   Up
> BridgedCall(SIP/2944079-e7f2)
> SIP/2944079-e7f2 [EMAIL PROTECTED]:2  Up  Dial(SIP/2944093|36|tr)
> 
> Why does the first line show bridged call, while the second does not?
> Why is the Location for the first line (None)?
> 
> 
>> -Original Message-
>> From: C F [mailto:[EMAIL PROTECTED]
>> Sent: Thursday, June 22, 2006 1:23 PM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [Asterisk-Users] Showing Current Calls
>> 
>> 
>> Whats wrong with show channels?
>> 
>> On 6/22/06, Douglas Garstang <[EMAIL PROTECTED]> wrote:
>>> 
>>> 
>>> Can someone recommend the best way to view current calls in
>> progress on the
>>> Asterisk console?
>>> Neither the 'show channels' or 'sip show channels' commands
>> are easy to
>>> read.
>>> 
>>> hestia*CLI> show channels
>>> Channel  Location State   Application(Data)
>>> 
>>> SIP/2944093-f9e2 (None)   Up  Bridged
>>> Call(SIP/2944079-e7f2)
>>> SIP/2944079-e7f2 [EMAIL PROTECTED]:2  Up
>> Dial(SIP/2944093|36|tr)
>>> 
>>> 2 active channels
>>> 1 active call
>>> 
>>> hestia*CLI>
>>> hestia*CLI> sip show channels
>>> Peer User/ANRCall ID  Seq (Tx/Rx)  Form
>>  Hold Last
>>> Message
>>> xxx.yyy.128.115  (None)  e77bba33-cc  00101/02261  unkn
>>  No   Rx:
>>> REGISTER
>>> xxx.yyy.128.110  (None)  739f4603-e8  00101/00778  unkn
>>  No   Rx:
>>> REGISTER
>>> xxx.yyy.128.86   (None)  56caad3a-eb  00101/01046  unkn
>>  No   Rx:
>>> REGISTER
>>> xxx.yyy.128.115  (None)  91ea0410-60  00101/02262  unkn
>>  No   Rx:
>>> REGISTER
>>> xxx.yyy.128.86   (None)  488801e-105  00101/01046  unkn
>>  No   Rx:
>>> REGISTER
>>> xxx.yyy.128.86   (None)  c3b27274-ef  00101/01194  unkn
>>  No   Rx:
>>> REGISTER
>>> xxx.yyy.128.77   2944093 2405f1ef74d  00102/0  ulaw
>>  No   Tx:
>>> ACK
>>> xxx.yyy.128.83   2944079 cf1722ef-cc  00101/2  ulaw
>>  No   Rx:
>>> ACK
>>> 
>>> Doug.
>>> 
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>>> 
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>>> 
>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>> 
>>> 
>>> 
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-- 
Joshua Colp
Software Developer
Digium
P - 256-428-6066
C - 506-878-0147
[EMAIL PROTECTED]


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Re: [Asterisk-Users] How to configure asterisk to emulate FXO signaling ?

[Tzafrir Cohen]

On Thu, Jun 22, 2006 at 11:38:42AM -0700, Carlos Munoz wrote:
> Tzafrir Cohen wrote:
> 
> >On Wed, Jun 21, 2006 at 05:21:17PM -0700, Carlos Munoz wrote:
> > 
> >
> >>Tzafrir Cohen wrote:
> >>
> >>   
> >>
> >>>On Wed, Jun 21, 2006 at 03:46:15PM -0700, Carlos Munoz wrote:
> >>>
> >>>
> >>> 
> >>>
> I'm unable to configure asterisk to provide dial tone, busy tone, 
> detect dtmf digits, etc to an analog phone connected to a FXS port. 
> Unfortunately, this particular  hardware does not provide FXO 
> signaling. Incoming calls work perfect, the phone rings and voice flows 
> on both directions. However, I can't place outgoing calls, there is no 
> dial tone and dialing out makes no difference.
> 
> Does anyone know how to configure asterisk to emulate FXO signaling or 
> point me in the right direction ?
>  
> 
>    
> 
> >>>What hardware is it, exactly?
> >>>
> >>>
> >>>
> >>> 
> >>>
> >>It is a small SOHO router from LinkSys that has 2 FXS ports, an ethernet 
> >>switch with 4 ports and a wireless interface. It came with Linux 2.6.14 
> >>installed and a VOIP application which uses 40% CPU when a call is up. 
> >>I'm trying to replace this VOIP application with asterisk configured as 
> >>TA (basically asterisk is a VOIP phone). I think the phone hardware is 
> >>based on legarity slics.
> >>   
> >>
> >
> >Did you actually get Asterisk running on it?
> >
> >Are there free drivers for the wireless innterface (so you won't depend
> >on the specific kernel version)? 
> >
> >You can't simply use the existing zaptel drivers, as the adapters don't
> >even sit on a PCI bus. If the slics are indeed the same ProSlics used by
> >the TDM400P, it could help. But some driver rewriting would probably be
> >needed.
> >
> >Are there any existing free drivers that use those adapters?
> >
> > 
> >
> Yes, I got asterisk to come up. I can receive calls with no problems. It 
> is outgoing calls that doesn't work. I think the VOIP application that 
> came with the router generates the dial tone and detects dtmf digits. It 
> looks like asterisk is not doing that. I'm going through the asterisk 
> code to see if I can find out how to enable dial tone generation and 
> dtmf digit detection.

channels/chan_zap.c (the horrors). zaptel/chan_zap generally does not
rely on the chip to generate/detect DTMF.

Alternatively, there is chan_unicall (not part of the standard Asterisk
codebase).

-- 
Tzafrir Cohen  sip:[EMAIL PROTECTED]
icq#16849755   iax:[EMAIL PROTECTED]
+972-50-7952406   
[EMAIL PROTECTED]  http://www.xorcom.com
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Re: [Asterisk-Users] Echo and crackle

[Joshua West]

Crackling is usually a sign of IRQ issues, as Mojo wrote.  Digium's full
documentation on solving IRQ issues are here: 
http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting

Regarding echo on POTS lines, I wish you the best of luck.  Fixing your
IRQ problems may reduce the delay enough that echo will not be noticed,
but we were unable to do just that with our previous system's
configuration (13 POTS lines).  After extensive testing, we finally
moved over to a T1 (ESF .. not an ISDN PRI).  Unfortunately, we still
experienced echo when calling or receiving calls from people/systems
using analog telephone lines.  Implementing a Ditech Communications
SX-24 T1 echo cancellation module inside the Gemini-T shelf solved our
problems.  Adios echo, no more users upset, etc.  This unit is however
no longer available from Ditech Communications
(ditechcommunications.com) as the product has gone end of life.

I recommend doing your best to remove the crackling first as the cause
of crackling could be also affecting echo.  There have been some people
reporting results of getting rid of echo on POTS lines by just tweaking
Asterisk/Zaptel.

If you get stuck on solving echo - or find its really hard to determine
if echo still exists - let me know.  I have a lot of good tricks up my
sleeve regarding exact measurement of echo using free software and a
little bit of analysis (with pretty pictures etc).  I should probably
put this info on the voip-info.org wiki... hmm...

-- 
Joshua West
Linux Infrastructure Engineer
Boston Engineering Corporation
http://www.boston-engineering.com



Mojo with Horan & Company, LLC wrote:
> I will agree that switching to the TDM card significantly helped my
> echo and sound quality, I would take a second to point out that
> interrupt sharing on your * server might cause crackling-like noises. 
> Try
>
> lspci -vb
>  and
> cat /proc/interrupts
>
> to see if you discern any hardware using the same irq the x101p is.
>
> Also run zttest in the zaptel source directory and see what the
> average number is.  s/b 100%, but I believe as low as 98% or
> thereabouts should have pretty acceptable call quality.  If it dips
> even lower than that regularly, or concurrently with the crackling
> sounds, then your server is probably working too hard on irrelevant
> processes or has irq sharing issues.
>
> Moj
>
> M.Hockings wrote:
>> We are running asterisk with a single POTS line for local calls and a
>> voip line for long distance.  Whenever we receive a call on the POTS
>> line it is more than likely, but not always, going to have
>> significant distracting echo.  In addition to that there is
>> occasional heavy crackle or static.  I have tried to follow the
>> guidelines at :
>> http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation
>> but it does not seem to have made much difference.
>>
>> Currently we are using a X101P for the POTS line and a TDM401B
>> (TDM400P with one FXS port) for the conventional telephones.  From
>> what I have read, it may help to use an FXO port on the TDM400P
>> rather than the X101P for the POTS line.
>>
>> What is the "conventional wisdom" about the echo in this
>> configuration and what causes the crackle and how should I tackle
>> that?  I have tried rebooting/power-cycling the machine to reset the
>> hardware but it really hasn't noticeably helped.
>>
>> Is there anything else that I can try?
>>
>> Thanks,  Mike
>>
>> ___
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>>
>> Asterisk-Users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>


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RE: [Asterisk-Users] Showing Current Calls

[Douglas Garstang]

Using this as an example:

hestia*CLI> show channels
Channel  Location State   Application(Data)
SIP/2944093-f9e2 (None)   Up  BridgedCall(SIP/2944079-e7f2)
SIP/2944079-e7f2 [EMAIL PROTECTED]:2  Up  Dial(SIP/2944093|36|tr)

Why does the first line show bridged call, while the second does not?
Why is the Location for the first line (None)?


> -Original Message-
> From: C F [mailto:[EMAIL PROTECTED]
> Sent: Thursday, June 22, 2006 1:23 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Showing Current Calls
> 
> 
> Whats wrong with show channels?
> 
> On 6/22/06, Douglas Garstang <[EMAIL PROTECTED]> wrote:
> >
> >
> > Can someone recommend the best way to view current calls in 
> progress on the
> > Asterisk console?
> > Neither the 'show channels' or 'sip show channels' commands 
> are easy to
> > read.
> >
> > hestia*CLI> show channels
> > Channel  Location State   Application(Data)
> >
> > SIP/2944093-f9e2 (None)   Up  Bridged
> > Call(SIP/2944079-e7f2)
> > SIP/2944079-e7f2 [EMAIL PROTECTED]:2  Up  
> Dial(SIP/2944093|36|tr)
> >
> > 2 active channels
> > 1 active call
> >
> > hestia*CLI>
> > hestia*CLI> sip show channels
> > Peer User/ANRCall ID  Seq (Tx/Rx)  Form 
>  Hold Last
> > Message
> > xxx.yyy.128.115  (None)  e77bba33-cc  00101/02261  unkn 
>  No   Rx:
> > REGISTER
> > xxx.yyy.128.110  (None)  739f4603-e8  00101/00778  unkn 
>  No   Rx:
> > REGISTER
> > xxx.yyy.128.86   (None)  56caad3a-eb  00101/01046  unkn 
>  No   Rx:
> > REGISTER
> > xxx.yyy.128.115  (None)  91ea0410-60  00101/02262  unkn 
>  No   Rx:
> > REGISTER
> > xxx.yyy.128.86   (None)  488801e-105  00101/01046  unkn 
>  No   Rx:
> > REGISTER
> > xxx.yyy.128.86   (None)  c3b27274-ef  00101/01194  unkn 
>  No   Rx:
> > REGISTER
> > xxx.yyy.128.77   2944093 2405f1ef74d  00102/0  ulaw 
>  No   Tx:
> > ACK
> > xxx.yyy.128.83   2944079 cf1722ef-cc  00101/2  ulaw 
>  No   Rx:
> > ACK
> >
> > Doug.
> >
> > ___
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> >
> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
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RE: [Asterisk-Users] Re: Can I enter an extension to dial whilevoicemail is playing?

[Tim Sharp]

The options are not seperated by commas.
 exten => s,1,Dial(SIP/50,23,r,d)
should be
 exten => s,1,Dial(SIP/50,23,rd)

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John Klimek
Sent: Thursday, June 22, 2006 2:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: Can I enter an extension to dial
whilevoicemail is playing?


Any idea why it wouldn't work in my dial plan?

On 6/22/06, Peter Antonacci <[EMAIL PROTECTED]> wrote:
> d: This flag trumps the 'H' flag and intercepts any dtmf while waiting for
> the call to be answered and returns that value on the spot. This allows you
> to dial a 1-digit exit extension while waiting for the call to be answered -
> see also
>
>
> On 6/22/06, John Klimek <[EMAIL PROTECTED]> wrote:
> > Anybody have any more information on this Dial() "d" option for incoming
> calls?
> >
> > On 6/19/06, John Klimek < [EMAIL PROTECTED]> wrote:
> > > Thanks for the information...
> > >
> > > After doing some reading it looks like I can use the "d" option with
> > > the Dial() command to be able to enter a 1-digit extension while the
> > > other extension is ringing, but this doesn't seem to be working for me
> > > either...
> > >
> > > Here is my new config:
> > >
> > > exten => s,1,Dial(SIP/50,23,r,d)
> > > exten => s,2,VoiceMail( [EMAIL PROTECTED])
> > > exten => s,3,Playback(vm-goodbye)
> > > exten => s,4,Hangup
> > >
> > > exten => 1,1,SayDigits(1)
> > > exten => 2,1,SayDigits(2)
> > > exten => 10,1,SayDigits(10)
> > >
> > > However, when my phone is ringing (eg. extension 50), I try entering
> > > "1" or "2" (to be forwarded via the Dial "d" option), but it doesn't
> > > do anything.
> > >
> > > What am I doing wrong?
> > >
> > > I like your solution above, but if I use that I'll need to wait 23
> > > seconds for Dial() to timeout before I can do anything.  I'd like to
> > > be immediately able to enter an extension (if possible, which maybe
> > > it's not...)
> > >
> > > On 6/19/06, Leah Newmark <[EMAIL PROTECTED]> wrote:
> > > > Using the Background command, you will be able to play the voicemail
> > > > while still being allowed to enter digits.
> > > >
> > > > exten => s,1,Wait(2)
> > > > exten =>
> 108,2,Background(voicemail/default/108/unavail)
> > > >
> > > >
> > > > exten => s,1,Dial(SIP/50,23,r)
> > > > exten =>
> s,2,Background(/voicemail/default/50/unavail) ;or whatever
> the
> > > > soundfile is called
> > > > exten => s,3,Voicemail(s50) ;s will skip the greeting and just go to
> the
> > > > beep
> > > > exten => s,4,Playback(vm-goodbye)
> > > > exten => s,5,Hangup
> > > >
> > > > You can then put
> > > > exten => 1, Dial(sip/me)
> > > > exten => 2, Dial(sip/her)
> > > > or whatever your dial statements look like.
> > > >
> > > > Leah Newmark
> > > > Capalon VoIP
> > > >
> > > >
> > > > [EMAIL PROTECTED] wrote:
> > > >
> > > > Message: 9
> > > > Date: Mon, 19 Jun 2006 14:18:22 -0400
> > > > From: "John Klimek" <[EMAIL PROTECTED]>
> > > > Subject: [Asterisk-Users] Can I enter an extension to dial while
> > > > voicemail   is playing?
> > > > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> > > > 
> > > > Message-ID:
> > > >
> <[EMAIL PROTECTED]>
> > > > Content-Type: text/plain; charset=ISO-8859-1; format=flowed
> > > >
> > > > I have a very, very simple Asterisk setup in my house.  I have a
> > > > Sipura 3000 with a PSTN line connected and one analog phone connected.
> > > >
> > > > The [incoming] context looks like this:
> > > >
> > > > exten => s,1,Dial(SIP/50,23,r)
> > > > exten => s,2,VoiceMail([EMAIL PROTECTED])
> > > > exten => s,3,Playback(vm-goodbye)
> > > > exten => s,4,Hangup
> > > >
> > > > As you can see, when somebody calls in if I don't answer in 23 seconds
> > > > then they are forwarded to my voicemail.
> > > >
> > > > How can I make it so I can call an enter extensions either while the
> > > > phone is ringing or while the voicemail message is playing?  I want
> > > > the system to be as seemless as possible so the wife is happy =)
> > > >
> > > > Right now it works great because my Sipura 3000 forwards to call to
> > > > Asterisk and Asterisk rings my analog phone, but the incoming caller
> > > > hears a steady dial-tone the whole time.  I wouldn't want that to
> > > > change.  (so the caller isn't wondering what is going on)
> > > >
> > > > Any help is appriciated  :)
> > > >
> > > > ___
> > > > --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
> > > >
> > >
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> >
> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >
> http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
> 

Re: [Asterisk-Users] record until silence, playback, repeat

[Mojo with Horan & Company, LLC]

replace the beep sound file with a silent one :)  I think it's beep.gsm

James Harper wrote:

I want to have something for the kids to play with which just records
until silence is detected, plays back what was recorded, then repeats.
They are having fun with Echo() at the moment :)

I have mocked something up with:
exten => *93,1,Answer
exten => *93,n,Record(/tmp/echo:alaw|1)
exten => *93,n,Playback(/tmp/echo)
exten => *93,n,Goto(2)

But it has the shortcomings that a beep is played before recording, and
that I can't detect any less silence than 1 second. The former is the
really annoying bit.

Any suggestions?

Thanks

James
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--
Mojo <[EMAIL PROTECTED]>
Office Manger, Horan & Company, LLC
(907) 747- x112
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[Asterisk-Users] Realtime monitor of a channel

[Ronan de Kermadec]

Hi,I would like to monitor in realtime the status of a given sip channel with the manager API and a web page. What is the better way to do that without using Asterisk 
Flash Operator Panel ?Thanks in advance.Ronan
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Re: [Asterisk-Users] Quality monitoring

[Anthony Rodgers]

Great - thanks, Curt!

A.

On Jun 22, 2006, at 11:30 AM, Curt Shaffer wrote:

It is really just a play on the check_icmp plugin. You could 
accomplish the

same thing by doing the following:


$USER1$/check_icmp -H $HOSTADDRESS$ -w 80.0,80% -c 100.0,100% -n 1

Where in this example it is an RTA of 80ms or 80% packet loss for a 
warning
and 100ms or 100% packet loss for critical. The perfdata is then 
passed to

perfparse for graphing.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony
Rodgers
Sent: Thursday, June 22, 2006 2:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Quality monitoring

Care to share your Nagios plugin?

Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp


On Jun 22, 2006, at 9:53 AM, Curt Shaffer wrote:

> Does anyone out there have a recommendation for tools that will
> monitor the quality of VoIP systems? I am looking for jitter and MOS
> monitoring. I have a custom Nagios plugin that is alerting me if the
> jitter jumps out of a 20ms but I am looking for a little more detail.
> I would not be against writing something in Perl for Nagios to do but
> I don’t really know where to start on measuring jitter other than 
with

> ICMP pulls and really don’t know where to start with doing MOS.
>  
> Any ideas?
>  
> Thanks
>  
> Curt


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Re: [Asterisk-Users] Showing Current Calls

[C F]

Whats wrong with show channels?

On 6/22/06, Douglas Garstang <[EMAIL PROTECTED]> wrote:



Can someone recommend the best way to view current calls in progress on the
Asterisk console?
Neither the 'show channels' or 'sip show channels' commands are easy to
read.

hestia*CLI> show channels
Channel  Location State   Application(Data)

SIP/2944093-f9e2 (None)   Up  Bridged
Call(SIP/2944079-e7f2)
SIP/2944079-e7f2 [EMAIL PROTECTED]:2  Up  Dial(SIP/2944093|36|tr)

2 active channels
1 active call

hestia*CLI>
hestia*CLI> sip show channels
Peer User/ANRCall ID  Seq (Tx/Rx)  Form  Hold Last
Message
xxx.yyy.128.115  (None)  e77bba33-cc  00101/02261  unkn  No   Rx:
REGISTER
xxx.yyy.128.110  (None)  739f4603-e8  00101/00778  unkn  No   Rx:
REGISTER
xxx.yyy.128.86   (None)  56caad3a-eb  00101/01046  unkn  No   Rx:
REGISTER
xxx.yyy.128.115  (None)  91ea0410-60  00101/02262  unkn  No   Rx:
REGISTER
xxx.yyy.128.86   (None)  488801e-105  00101/01046  unkn  No   Rx:
REGISTER
xxx.yyy.128.86   (None)  c3b27274-ef  00101/01194  unkn  No   Rx:
REGISTER
xxx.yyy.128.77   2944093 2405f1ef74d  00102/0  ulaw  No   Tx:
ACK
xxx.yyy.128.83   2944079 cf1722ef-cc  00101/2  ulaw  No   Rx:
ACK

Doug.

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[Asterisk-Users] freepbx centos 4 install script?

[Warren]

Has anyone created a script that will download and install all of the
freepbx prerequisites in the INSTALL file automatically on a Centos 4 box?

TIA,
W
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Re: [Asterisk-Users] Don't use CDRTool From AG-projescts

[Brian Capouch]

[EMAIL PROTECTED] wrote:

hello to all,

I advice you to not use 


Harry!!

Only one post is needed for each of your silly complaints.

Please don't give people even more reason to relegate you to their 
killfiles.


B.

--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.

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Re: [Asterisk-Users] Echo and crackle

[Mojo with Horan & Company, LLC]

I will agree that switching to the TDM card significantly helped my echo 
and sound quality, I would take a second to point out that interrupt 
sharing on your * server might cause crackling-like noises.  Try


lspci -vb
 and
cat /proc/interrupts

to see if you discern any hardware using the same irq the x101p is.

Also run zttest in the zaptel source directory and see what the average 
number is.  s/b 100%, but I believe as low as 98% or thereabouts should 
have pretty acceptable call quality.  If it dips even lower than that 
regularly, or concurrently with the crackling sounds, then your server 
is probably working too hard on irrelevant processes or has irq sharing 
issues.


Moj

M.Hockings wrote:
We are running asterisk with a single POTS line for local calls and a 
voip line for long distance.  Whenever we receive a call on the POTS 
line it is more than likely, but not always, going to have significant 
distracting echo.  In addition to that there is occasional heavy crackle 
or static.  I have tried to follow the guidelines at :

http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation
but it does not seem to have made much difference.

Currently we are using a X101P for the POTS line and a TDM401B (TDM400P 
with one FXS port) for the conventional telephones.  From what I have 
read, it may help to use an FXO port on the TDM400P rather than the 
X101P for the POTS line.


What is the "conventional wisdom" about the echo in this configuration 
and what causes the crackle and how should I tackle that?  I have tried 
rebooting/power-cycling the machine to reset the hardware but it really 
hasn't noticeably helped.


Is there anything else that I can try?

Thanks,  Mike

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--
Mojo <[EMAIL PROTECTED]>
Office Manger, Horan & Company, LLC
(907) 747- x112
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[Asterisk-Users] Soekris net4801 and IAXy dhcp issue

[Juan Luis Moyano]

Hi all, I have a Soekris net4801-50 board with OpenBSD 3.9 where I've 
configured a dhcp server and tested it with a regular PC connected 
directly via a crossover cable with success. The problem comes when I 
try to connect my IAXy device instead of the PC. I can see with 'tcpdump 
-nettti sis1' that the IAXy isn't sending any packets to the dhcp 
server. I thought my IAXy was bad but then I configured a second dhcp 
server with the exact same config file and the IAXy worked right out. So 
I don't have a clue of what could be happening. Please shed me some 
light on this issue. Thanks in advance.


Juan Luis Moyano

#cat /etc/dhcpd.conf
shared-network LOCAL-NET {

   option  domain-name "b-fon.com.ar";
   option  domain-name-servers 10.32.2.254, 200.69.193.1, 200.69.193.2;

   subnet 10.32.2.0 netmask 255.255.255.0 {
   option routers 10.32.2.254;
   range 10.32.2.32 10.32.2.64;
   }
}


# tcpdump -nettti sis1
tcpdump: listening on sis1, link-type EN10MB
^C
0 packets received by filter
0 packets dropped by kernel


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Re: [Asterisk-Users] Re: Can I enter an extension to dial while voicemail is playing?

[John Klimek]

Any idea why it wouldn't work in my dial plan?

On 6/22/06, Peter Antonacci <[EMAIL PROTECTED]> wrote:

d: This flag trumps the 'H' flag and intercepts any dtmf while waiting for
the call to be answered and returns that value on the spot. This allows you
to dial a 1-digit exit extension while waiting for the call to be answered -
see also


On 6/22/06, John Klimek <[EMAIL PROTECTED]> wrote:
> Anybody have any more information on this Dial() "d" option for incoming
calls?
>
> On 6/19/06, John Klimek < [EMAIL PROTECTED]> wrote:
> > Thanks for the information...
> >
> > After doing some reading it looks like I can use the "d" option with
> > the Dial() command to be able to enter a 1-digit extension while the
> > other extension is ringing, but this doesn't seem to be working for me
> > either...
> >
> > Here is my new config:
> >
> > exten => s,1,Dial(SIP/50,23,r,d)
> > exten => s,2,VoiceMail( [EMAIL PROTECTED])
> > exten => s,3,Playback(vm-goodbye)
> > exten => s,4,Hangup
> >
> > exten => 1,1,SayDigits(1)
> > exten => 2,1,SayDigits(2)
> > exten => 10,1,SayDigits(10)
> >
> > However, when my phone is ringing (eg. extension 50), I try entering
> > "1" or "2" (to be forwarded via the Dial "d" option), but it doesn't
> > do anything.
> >
> > What am I doing wrong?
> >
> > I like your solution above, but if I use that I'll need to wait 23
> > seconds for Dial() to timeout before I can do anything.  I'd like to
> > be immediately able to enter an extension (if possible, which maybe
> > it's not...)
> >
> > On 6/19/06, Leah Newmark <[EMAIL PROTECTED]> wrote:
> > > Using the Background command, you will be able to play the voicemail
> > > while still being allowed to enter digits.
> > >
> > > exten => s,1,Wait(2)
> > > exten =>
108,2,Background(voicemail/default/108/unavail)
> > >
> > >
> > > exten => s,1,Dial(SIP/50,23,r)
> > > exten =>
s,2,Background(/voicemail/default/50/unavail) ;or whatever
the
> > > soundfile is called
> > > exten => s,3,Voicemail(s50) ;s will skip the greeting and just go to
the
> > > beep
> > > exten => s,4,Playback(vm-goodbye)
> > > exten => s,5,Hangup
> > >
> > > You can then put
> > > exten => 1, Dial(sip/me)
> > > exten => 2, Dial(sip/her)
> > > or whatever your dial statements look like.
> > >
> > > Leah Newmark
> > > Capalon VoIP
> > >
> > >
> > > [EMAIL PROTECTED] wrote:
> > >
> > > Message: 9
> > > Date: Mon, 19 Jun 2006 14:18:22 -0400
> > > From: "John Klimek" <[EMAIL PROTECTED]>
> > > Subject: [Asterisk-Users] Can I enter an extension to dial while
> > > voicemail   is playing?
> > > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> > > 
> > > Message-ID:
> > >
<[EMAIL PROTECTED]>
> > > Content-Type: text/plain; charset=ISO-8859-1; format=flowed
> > >
> > > I have a very, very simple Asterisk setup in my house.  I have a
> > > Sipura 3000 with a PSTN line connected and one analog phone connected.
> > >
> > > The [incoming] context looks like this:
> > >
> > > exten => s,1,Dial(SIP/50,23,r)
> > > exten => s,2,VoiceMail([EMAIL PROTECTED])
> > > exten => s,3,Playback(vm-goodbye)
> > > exten => s,4,Hangup
> > >
> > > As you can see, when somebody calls in if I don't answer in 23 seconds
> > > then they are forwarded to my voicemail.
> > >
> > > How can I make it so I can call an enter extensions either while the
> > > phone is ringing or while the voicemail message is playing?  I want
> > > the system to be as seemless as possible so the wife is happy =)
> > >
> > > Right now it works great because my Sipura 3000 forwards to call to
> > > Asterisk and Asterisk rings my analog phone, but the incoming caller
> > > hears a steady dial-tone the whole time.  I wouldn't want that to
> > > change.  (so the caller isn't wondering what is going on)
> > >
> > > Any help is appriciated  :)
> > >
> > > ___
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> > >
> > > Asterisk-Users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > >
http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
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Re: SV: [Asterisk-Users] periodic-announce not working

[BJ Weschke]

On 6/22/06, Michiel van Baak <[EMAIL PROTECTED]> wrote:


On Jun 22, 2006, at 7:18 PM, BJ Weschke wrote:

> On 6/22/06, Michiel van Baak <[EMAIL PROTECTED]> wrote:
>> On Jun 22, 2006, at 5:22 PM, BJ Weschke wrote:
>> >
>> > Submit a feature request/patch to bugs.digium.com. There isn't
>> > presently a way to do hold time announcements without queue
>> position
>> > along with it.
>>
>> I dont want hold time neither.
>> I simply want 1 file played to the ppl waiting in queue every 25
>> seconds.
>>
>> I think I can fix it by creating a soundfile with 1 ms silence and
>> put that as position soundfiles in every queue, but that's ugly.
>>
>> If that is what it takes, I will go implement it in my dialplan and
>> look into app_queue.c later
>>
>
> Then I'd say that's a bug and it needs to be fixed. Please post a bug
> to bugs.digium.com. Thanks.

I found out what the problem is.
exten => 12,1,Queue(12|tr) ; does not play announcement
exten => 13,1,Queue(12|t)  ; does play announcement

I reverted my queues.conf to read:
announce-frequency = 0
periodic-announce-frequency = 15

Providing the r flag to the queue call will kill the announcements,
even the position/holdtime announcement.
Is this a bug or on purpose ?
___


That depends. Has the channel going into Queue been "answered" yet
before you send it in with r ?

--
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http://www.btwtech.com/
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Re: [Asterisk-Users] caller id

[Joshua West]

Do you have the Caller ID feature with your telephone service package?

sdgesa gaeharth wrote:
> How can I get the external caller id to show on the polycom 501
> phones. Currently, when someone calls our office, we only see the word
> "asterisk" in the caller id.
>
> This is our set up:
>
> VOIP(polycom)<--->Asterisk 1.2.4<--->PSTN
>
> Thanks
>
> 
> Yahoo! Groups gets better. Check out the new email design.
> 
> Plus there’s much more to come.
> 
>
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>   


-- 
Joshua West
Linux Infrastructure Engineer
Boston Engineering Corporation
http://www.boston-engineering.com


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Re: [Asterisk-Users] How to configure asterisk to emulate FXO signaling ?

[Carlos Munoz]

Tzafrir Cohen wrote:


On Wed, Jun 21, 2006 at 05:21:17PM -0700, Carlos Munoz wrote:
 


Tzafrir Cohen wrote:

   


On Wed, Jun 21, 2006 at 03:46:15PM -0700, Carlos Munoz wrote:


 

I'm unable to configure asterisk to provide dial tone, busy tone, detect 
dtmf digits, etc to an analog phone connected to a FXS port. 
Unfortunately, this particular  hardware does not provide FXO signaling. 
Incoming calls work perfect, the phone rings and voice flows on both 
directions. However, I can't place outgoing calls, there is no dial tone 
and dialing out makes no difference.


Does anyone know how to configure asterisk to emulate FXO signaling or 
point me in the right direction ?
 

   


What hardware is it, exactly?



 

It is a small SOHO router from LinkSys that has 2 FXS ports, an ethernet 
switch with 4 ports and a wireless interface. It came with Linux 2.6.14 
installed and a VOIP application which uses 40% CPU when a call is up. 
I'm trying to replace this VOIP application with asterisk configured as 
TA (basically asterisk is a VOIP phone). I think the phone hardware is 
based on legarity slics.
   



Did you actually get Asterisk running on it?

Are there free drivers for the wireless innterface (so you won't depend
on the specific kernel version)? 


You can't simply use the existing zaptel drivers, as the adapters don't
even sit on a PCI bus. If the slics are indeed the same ProSlics used by
the TDM400P, it could help. But some driver rewriting would probably be
needed.

Are there any existing free drivers that use those adapters?

 

Yes, I got asterisk to come up. I can receive calls with no problems. It 
is outgoing calls that doesn't work. I think the VOIP application that 
came with the router generates the dial tone and detects dtmf digits. It 
looks like asterisk is not doing that. I'm going through the asterisk 
code to see if I can find out how to enable dial tone generation and 
dtmf digit detection.



Carlos Munoz
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[Asterisk-Users] Don't use CDRTool From AG-projescts

[hgaillac-sip]

hello to all,

I advice you to not use CDRtool from ag-projects :
Fisrt ag-projects talk about is product like a gpl
software however they don't provide at least some
documentation for non commercial users .

try to call them !!
i'll offer you some money .

You can not Call them for some advices ...

It's really a bad product don't waste your time to
setup it. 
this enterprise must be  fogotten it's ag-projects .
it's not a reliable society ... more and more 

projects around open(ser) asterisk and more are
offered good unliked projects cdrtool please do not
use ag-projects products !


Harry is not Harry Potter !

Regards

  











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RE: [Asterisk-Users] Quality monitoring

[Curt Shaffer]

It is really just a play on the check_icmp plugin. You could accomplish the
same thing by doing the following:


$USER1$/check_icmp -H $HOSTADDRESS$ -w 80.0,80% -c 100.0,100% -n 1

Where in this example it is an RTA of 80ms or 80% packet loss for a warning
and 100ms or 100% packet loss for critical. The perfdata is then passed to
perfparse for graphing. 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony
Rodgers
Sent: Thursday, June 22, 2006 2:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Quality monitoring

Care to share your Nagios plugin?

Regards,
-- 
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp


On Jun 22, 2006, at 9:53 AM, Curt Shaffer wrote:

> Does anyone out there have a recommendation for tools that will 
> monitor the quality of VoIP systems? I am looking for jitter and MOS 
> monitoring. I have a custom Nagios plugin that is alerting me if the 
> jitter jumps out of a 20ms but I am looking for a little more detail. 
> I would not be against writing something in Perl for Nagios to do but 
> I don’t really know where to start on measuring jitter other than with 
> ICMP pulls and really don’t know where to start with doing MOS.
>  
> Any ideas?
>  
> Thanks
>  
> Curt
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Re: [Asterisk-Users] Re: Can I enter an extension to dial while voicemail is playing?

[Peter Antonacci]

d: This flag trumps the 'H' flag and intercepts any dtmf while waiting for the call to be answered and returns that value on the spot. This allows you to dial a 1-digit exit extension while waiting for the call to be answered - see also

On 6/22/06, John Klimek <[EMAIL PROTECTED]> wrote:
Anybody have any more information on this Dial() "d" option for incoming calls?On 6/19/06, John Klimek <
[EMAIL PROTECTED]> wrote:> Thanks for the information...>> After doing some reading it looks like I can use the "d" option with> the Dial() command to be able to enter a 1-digit extension while the
> other extension is ringing, but this doesn't seem to be working for me> either...>> Here is my new config:>> exten => s,1,Dial(SIP/50,23,r,d)> exten => s,2,VoiceMail(
[EMAIL PROTECTED])> exten => s,3,Playback(vm-goodbye)> exten => s,4,Hangup>> exten => 1,1,SayDigits(1)> exten => 2,1,SayDigits(2)> exten => 10,1,SayDigits(10)>
> However, when my phone is ringing (eg. extension 50), I try entering> "1" or "2" (to be forwarded via the Dial "d" option), but it doesn't> do anything.>> What am I doing wrong?
>> I like your solution above, but if I use that I'll need to wait 23> seconds for Dial() to timeout before I can do anything.  I'd like to> be immediately able to enter an extension (if possible, which maybe
> it's not...)>> On 6/19/06, Leah Newmark <[EMAIL PROTECTED]> wrote:> > Using the Background command, you will be able to play the voicemail
> > while still being allowed to enter digits.> >> > exten => s,1,Wait(2)> > exten => 108,2,Background(voicemail/default/108/unavail)> >> >> > exten => s,1,Dial(SIP/50,23,r)
> > exten => s,2,Background(/voicemail/default/50/unavail) ;or whatever the> > soundfile is called> > exten => s,3,Voicemail(s50) ;s will skip the greeting and just go to the> > beep
> > exten => s,4,Playback(vm-goodbye)> > exten => s,5,Hangup> >> > You can then put> > exten => 1, Dial(sip/me)> > exten => 2, Dial(sip/her)> > or whatever your dial statements look like.
> >> > Leah Newmark> > Capalon VoIP> >> >> > [EMAIL PROTECTED] wrote:> >
> > Message: 9> > Date: Mon, 19 Jun 2006 14:18:22 -0400> > From: "John Klimek" <[EMAIL PROTECTED]>> > Subject: [Asterisk-Users] Can I enter an extension to dial while
> > voicemail   is playing?> > To: "Asterisk Users Mailing List - Non-Commercial Discussion"> > > > Message-ID:> > <[EMAIL PROTECTED]>> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed
> >> > I have a very, very simple Asterisk setup in my house.  I have a> > Sipura 3000 with a PSTN line connected and one analog phone connected.> >> > The [incoming] context looks like this:
> >> > exten => s,1,Dial(SIP/50,23,r)> > exten => s,2,VoiceMail([EMAIL PROTECTED])> > exten => s,3,Playback(vm-goodbye)> > exten => s,4,Hangup> >> > As you can see, when somebody calls in if I don't answer in 23 seconds
> > then they are forwarded to my voicemail.> >> > How can I make it so I can call an enter extensions either while the> > phone is ringing or while the voicemail message is playing?  I want
> > the system to be as seemless as possible so the wife is happy =)> >> > Right now it works great because my Sipura 3000 forwards to call to> > Asterisk and Asterisk rings my analog phone, but the incoming caller
> > hears a steady dial-tone the whole time.  I wouldn't want that to> > change.  (so the caller isn't wondering what is going on)> >> > Any help is appriciated  :)> >
> > ___> > --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 
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Re: SV: [Asterisk-Users] periodic-announce not working

[Michiel van Baak]

On Jun 22, 2006, at 7:18 PM, BJ Weschke wrote:


On 6/22/06, Michiel van Baak <[EMAIL PROTECTED]> wrote:

On Jun 22, 2006, at 5:22 PM, BJ Weschke wrote:
>
> Submit a feature request/patch to bugs.digium.com. There isn't
> presently a way to do hold time announcements without queue  
position

> along with it.

I dont want hold time neither.
I simply want 1 file played to the ppl waiting in queue every 25
seconds.

I think I can fix it by creating a soundfile with 1 ms silence and
put that as position soundfiles in every queue, but that's ugly.

If that is what it takes, I will go implement it in my dialplan and
look into app_queue.c later



Then I'd say that's a bug and it needs to be fixed. Please post a bug
to bugs.digium.com. Thanks.


I found out what the problem is.
exten => 12,1,Queue(12|tr) ; does not play announcement
exten => 13,1,Queue(12|t)  ; does play announcement

I reverted my queues.conf to read:
announce-frequency = 0
periodic-announce-frequency = 15

Providing the r flag to the queue call will kill the announcements,  
even the position/holdtime announcement.

Is this a bug or on purpose ?
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Re: [Asterisk-Users] PRI Issue - Calls being rejected with unacceptable channel

[Steve Totaro]

Andy Brezinsky wrote:
Hey all.  We have a DS3 circuit with GBLX split off into 7 systems 
with a 4 port sangoma card (A104D) in the first 2 systems, and digium 
T410P cards in the other 5.  GBLX numbers their spans from 0 to 3 
instead of 1-4 and we have a NFAS configuration with the d-channel on 
chan 96.  All of our systems are running 1.0.7 for stability reasons 
(and no good time for maintaince, the entire platform is used most of 
the day) but if an upgrade will help us, we'll schedule it soon.


We've recently been experiencing people not being able to get in.  We 
have a hunt group tied in over our 7 trunks which will roll them if a 
trunk is busy or out of order.  It seems that call comes into this 
termination system (see trace below), we fire back a "Cause: Channel 
unacceptable (6)" event to GBLX and they try the next system, even if 
this system isn't busy.  Because of this, calls can eventually try all 
7 systems, get rejected, and then users get busy messages even though 
we're not at total capacity yet.  Below I've attached the entire pri 
debug of one of these events happening.  Can anyone shed some light on 
where we should be looking to fix this stuff?


milwia1-terma-2*CLI> pri debug span 4
Enabled debugging on span 4
< Protocol Discriminator: Q.931 (8)  len=47
< Call Ref: len= 2 (reference 15996/0x3E7C) (Originator)
< Message type: SETUP (5)
< [04 03 80 90 a2]
< Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer 
capability: Speech (0)
<  Ext: 1  Trans mode/rate: 64kbps, 
circuit-mode (16)
<  Ext: 1  User information layer 1: u-Law 
(34)

< [18 04 e9 80 83 8e]
< Channel ID (len= 6) [ Ext: 1  IntID: Explicit, PRI Spare: 0, 
Exclusive Dchan: 0

 Protocol Discriminator: Q.931 (8)  len=10
> Call Ref: len= 2 (reference 48764/0xBE7C) (Terminator)
> Message type: CALL PROCEEDING (2)
> [18 03 a9 83 8e]
> Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, 
Exclusive Dchan: 0

>ChanSel: Reserved
>   Ext: 1  Coding: 0   Number Specified   Channel 
Type: 3

>   Ext: 1  Channel: 14 ]
   -- Accepting voice call from '323801' to '800978' on 
channel 0/14, span 4

   -- Executing SetVar("Zap/14-1", "SERVER_ID=2") in new stack
   -- Executing Answer("Zap/14-1", "") in new stack
> Protocol Discriminator: Q.931 (8)  len=14
> Call Ref: len= 2 (reference 48764/0xBE7C) (Terminator)
> Message type: CONNECT (7)
> [18 03 a9 83 8e]
> Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, 
Exclusive Dchan: 0

>ChanSel: Reserved
>   Ext: 1  Coding: 0   Number Specified   Channel 
Type: 3

>   Ext: 1  Channel: 14 ]
> [1e 02 81 82]
> Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard 
(0) 0: 0   Location: Private network serving the local user (1)
>   Ext: 1  Progress Description: Called 
equipment is non-ISDN. (2) ]

   -- Executing AGI("Zap/14-1", "incoming_call.pl") in new stack
   -- Launched AGI Script /var/lib/asterisk/agi-bin/incoming_call.pl
< Protocol Discriminator: Q.931 (8)  len=9
< Call Ref: len= 2 (reference 15996/0x3E7C) (Originator)
< Message type: RELEASE (77)
< [08 02 83 86]
< Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0   
Location: Transit network (3)
<  Ext: 1  Cause: Channel unacceptable (6), class = 
Normal Event (0) ]

-- Processing IE 8 (cs0, Cause)
   -- Channel 0/14, span 4 got hangup
< Protocol Discriminator: Q.931 (8)  len=13
< Call Ref: len= 2 (reference 15996/0x3E7C) (Originator)
< Message type: STATUS (125)
< [08 03 83 e5 07]
< Cause (len= 5) [ Ext: 1  Coding: CCITT (ITU) standard (

[Asterisk-Users] Showing Current Calls

[Douglas Garstang]

Can someone 
recommend the best way to view current calls in progress on the Asterisk 
console?
Neither the 'show 
channels' or 'sip show channels' commands are easy to read.
 
hestia*CLI> show 
channelsChannel  
Location 
State   
Application(Data) 
SIP/2944093-f9e2 
(None)   
Up  Bridged 
Call(SIP/2944079-e7f2)SIP/2944079-e7f2 [EMAIL PROTECTED]:2  
Up  
Dial(SIP/2944093|36|tr)   2 active 
channels1 active call
hestia*CLI> 
hestia*CLI> sip show 
channelsPeer 
User/ANR    Call ID  Seq 
(Tx/Rx)  Form  Hold Last 
Message   xxx.yyy.128.115  
(None)  e77bba33-cc  00101/02261  
unkn  No   Rx: 
REGISTER   xxx.yyy.128.110  
(None)  739f4603-e8  00101/00778  
unkn  No   Rx: 
REGISTER   xxx.yyy.128.86   
(None)  56caad3a-eb  00101/01046  
unkn  No   Rx: 
REGISTER   xxx.yyy.128.115  
(None)  91ea0410-60  00101/02262  
unkn  No   Rx: 
REGISTER   xxx.yyy.128.86   
(None)  488801e-105  00101/01046  
unkn  No   Rx: 
REGISTER   xxx.yyy.128.86   
(None)  c3b27274-ef  00101/01194  
unkn  No   Rx: 
REGISTER   xxx.yyy.128.77   
2944093 2405f1ef74d  00102/0  ulaw  
No   Tx: 
ACKxxx.yyy.128.83   
2944079 cf1722ef-cc  00101/2  ulaw  
No   Rx: 
ACK    
 
Doug.
 
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[Asterisk-Users] caller id

[sdgesa gaeharth]

How can I get the external caller id to show on the polycom 501 phones.  Currently, when someone calls our office,  we only see the word  "asterisk" in the caller id.This is our set up:VOIP(polycom)<--->Asterisk 1.2.4<--->PSTNThanks 
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[Asterisk-Users] Playing sounds from the CLI

[J.J. Feminella]

Once 
I'm inside the the asterisk CLI and I'm on a call with another extension, how do 
I play sounds from the CLI? It doesn't appear that I can run AGI commands 
directly -- is there another way that I'm missing?
 
thanks,
JJ
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Re: [Asterisk-Users] Out of Office Auto Reply:

[Anthony Rodgers]

We use MS Exchange too and, as far as I am aware, it is cognizant of 
mailing list headers and doesn't send OOO notices to mailing list 
postings. The only mailing list from which I receive my own OOO notices 
is one that doesn't have the proper mailing list headers set.


When you receive a lot of email from outside your organization from 
people who expect a response, it is helpful to us (and them) if they 
receive OOO notifications.


Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp


On Jun 22, 2006, at 10:12 AM, Colin Anderson wrote:

He's probably using Exchange which has a global setting to either send 
OOO
replies to SMTP addresses or not. It's a dumbass Exchange 
administrator who

enables this option (it is actually on by default)



-Original Message-
From: Anthony Rodgers [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 22, 2006 10:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Out of Office Auto Reply:


Actually, if his MTA is configured properly, it shouldn't happen at
all.

A.

On Jun 22, 2006, at 9:32 AM, Doug Geary wrote:

> Should only happen once if his email system is config'd in a standard
> method. Otherwise just *plonk* his address.
>
> > -Original Message-
> > From: [EMAIL PROTECTED] 
[mailto:asterisk-users-

> > [EMAIL PROTECTED] On Behalf Of Dean Collins
> > Sent: Thursday, June 22, 2006 12:03 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Cc: [EMAIL PROTECTED]
> > Subject: RE: [Asterisk-Users] Out of Office Auto Reply:
> >
> > You got to be freaking kidding, a month of this?
> > Cant we get an easy process for the list owner to take care of 
these?


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Re: [Asterisk-Users] Re: Can I enter an extension to dial while voicemail is playing?

[John Klimek]

Anybody have any more information on this Dial() "d" option for incoming calls?

On 6/19/06, John Klimek <[EMAIL PROTECTED]> wrote:

Thanks for the information...

After doing some reading it looks like I can use the "d" option with
the Dial() command to be able to enter a 1-digit extension while the
other extension is ringing, but this doesn't seem to be working for me
either...

Here is my new config:

exten => s,1,Dial(SIP/50,23,r,d)
exten => s,2,VoiceMail([EMAIL PROTECTED])
exten => s,3,Playback(vm-goodbye)
exten => s,4,Hangup

exten => 1,1,SayDigits(1)
exten => 2,1,SayDigits(2)
exten => 10,1,SayDigits(10)

However, when my phone is ringing (eg. extension 50), I try entering
"1" or "2" (to be forwarded via the Dial "d" option), but it doesn't
do anything.

What am I doing wrong?

I like your solution above, but if I use that I'll need to wait 23
seconds for Dial() to timeout before I can do anything.  I'd like to
be immediately able to enter an extension (if possible, which maybe
it's not...)

On 6/19/06, Leah Newmark <[EMAIL PROTECTED]> wrote:
> Using the Background command, you will be able to play the voicemail
> while still being allowed to enter digits.
>
> exten => s,1,Wait(2)
> exten => 108,2,Background(voicemail/default/108/unavail)
>
>
> exten => s,1,Dial(SIP/50,23,r)
> exten => s,2,Background(/voicemail/default/50/unavail) ;or whatever the
> soundfile is called
> exten => s,3,Voicemail(s50) ;s will skip the greeting and just go to the
> beep
> exten => s,4,Playback(vm-goodbye)
> exten => s,5,Hangup
>
> You can then put
> exten => 1, Dial(sip/me)
> exten => 2, Dial(sip/her)
> or whatever your dial statements look like.
>
> Leah Newmark
> Capalon VoIP
>
>
> [EMAIL PROTECTED] wrote:
>
> Message: 9
> Date: Mon, 19 Jun 2006 14:18:22 -0400
> From: "John Klimek" <[EMAIL PROTECTED]>
> Subject: [Asterisk-Users] Can I enter an extension to dial while
> voicemail   is playing?
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Message-ID:
> <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> I have a very, very simple Asterisk setup in my house.  I have a
> Sipura 3000 with a PSTN line connected and one analog phone connected.
>
> The [incoming] context looks like this:
>
> exten => s,1,Dial(SIP/50,23,r)
> exten => s,2,VoiceMail([EMAIL PROTECTED])
> exten => s,3,Playback(vm-goodbye)
> exten => s,4,Hangup
>
> As you can see, when somebody calls in if I don't answer in 23 seconds
> then they are forwarded to my voicemail.
>
> How can I make it so I can call an enter extensions either while the
> phone is ringing or while the voicemail message is playing?  I want
> the system to be as seemless as possible so the wife is happy =)
>
> Right now it works great because my Sipura 3000 forwards to call to
> Asterisk and Asterisk rings my analog phone, but the incoming caller
> hears a steady dial-tone the whole time.  I wouldn't want that to
> change.  (so the caller isn't wondering what is going on)
>
> Any help is appriciated  :)
>
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Re: [Asterisk-Users] Out of Office Auto Reply:

[Martin Joseph]

On Jun 22, 2006, at 10:12 AM, Colin Anderson wrote:

He's probably using Exchange which has a global setting to either send 
OOO
replies to SMTP addresses or not. It's a dumbass Exchange 
administrator who

enables this option (it is actually on by default)

Same thing happened to the mac-asterisk list last week, except the OOO
message would reply to every post, and to every user. In a period of 2 
or 3

hours, I got a couple of thousand OOO replies from the offender. The
solution was to unsubscribe since the list owner was out of town 
apparently,
and so the list members did just that - I think the only guy that was 
still
subscribed once the dust settled was the list owner and the tool that 
set up

the OOO message in the first place.

Actually there where 440 messages, and I didn't unsubscribe...

So your story is whacked...  As is the mac asterisk list.


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Re: [Asterisk-Users] Quality monitoring

[Anthony Rodgers]

Care to share your Nagios plugin?

Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp


On Jun 22, 2006, at 9:53 AM, Curt Shaffer wrote:

Does anyone out there have a recommendation for tools that will 
monitor the quality of VoIP systems? I am looking for jitter and MOS 
monitoring. I have a custom Nagios plugin that is alerting me if the 
jitter jumps out of a 20ms but I am looking for a little more detail. 
I would not be against writing something in Perl for Nagios to do but 
I don’t really know where to start on measuring jitter other than with 
ICMP pulls and really don’t know where to start with doing MOS.

 
Any ideas?
 
Thanks
 
Curt
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[Asterisk-Users] php-snmp

[Matthew Warren]

Has anyone been able to get PHP-SNMP working on an asterisk box.  I have
downloaded the net-snmp, utils,libs, perl and php-snmp but unable to get the
php to work wit it.  It works via command line // without php.  This is set
up on an AAH box.

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[Asterisk-Users] Passing DID to external number?

[Brian McCarey]

Hi,
 
We run a small 
switchboard using Asterisk and Free PBX.
 
We have two main 
extensions and two ring groups. The first ring group rings the two internal 
extensions. If the internal extensions do not pick up the call after 15 seconds 
then the second ring group kicks in which should ring the two internal 
extensions plus two external numbers.
 
Firstly, how do I 
pass the DID number of an incoming call to the external number so that the 
external number sees the incoming number and not the voip dial out 
number?
 
Secondly, when the 
second ring group kicks in only one of the external numbers dials when both 
internal extensions and both external numbers should ring according to the ring 
group setting. Any ideals what's going wrong?
 
Kind 
regards
 
Brian. 
UK
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Re: [Asterisk-Users] 1.2.9.1 crashed today

[Tim C. Lewis]

On Thu, 22 Jun 2006, BJ Weschke wrote:

On 6/22/06, Matt <[EMAIL PROTECTED]> wrote:

 We're now back on 1.2.6 and running stable.  Been running for over 17
 hours.  Something is wrong with 1.2.9.1



Sorry. I may have asked this already, but are you running the tarball
releases or checkouts from SVN? I've seen some similar behavior on
some of our client systems and I'm trying to get a better read of
which date/commit things went wrong.


i was using the released tar, NOT svn.

-tcl.

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[Asterisk-Users] PRI Issue - Calls being rejected with unacceptable channel

[Andy Brezinsky]

Hey all.  We have a DS3 circuit with GBLX split off into 7 systems with 
a 4 port sangoma card (A104D) in the first 2 systems, and digium T410P 
cards in the other 5.  GBLX numbers their spans from 0 to 3 instead of 
1-4 and we have a NFAS configuration with the d-channel on chan 96.  All 
of our systems are running 1.0.7 for stability reasons (and no good time 
for maintaince, the entire platform is used most of the day) but if an 
upgrade will help us, we'll schedule it soon.


We've recently been experiencing people not being able to get in.  We 
have a hunt group tied in over our 7 trunks which will roll them if a 
trunk is busy or out of order.  It seems that call comes into this 
termination system (see trace below), we fire back a "Cause: Channel 
unacceptable (6)" event to GBLX and they try the next system, even if 
this system isn't busy.  Because of this, calls can eventually try all 7 
systems, get rejected, and then users get busy messages even though 
we're not at total capacity yet.  Below I've attached the entire pri 
debug of one of these events happening.  Can anyone shed some light on 
where we should be looking to fix this stuff?


milwia1-terma-2*CLI> pri debug span 4
Enabled debugging on span 4
< Protocol Discriminator: Q.931 (8)  len=47
< Call Ref: len= 2 (reference 15996/0x3E7C) (Originator)
< Message type: SETUP (5)
< [04 03 80 90 a2]
< Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer 
capability: Speech (0)
<  Ext: 1  Trans mode/rate: 64kbps, 
circuit-mode (16)

<  Ext: 1  User information layer 1: u-Law (34)
< [18 04 e9 80 83 8e]
< Channel ID (len= 6) [ Ext: 1  IntID: Explicit, PRI Spare: 0, Exclusive 
Dchan: 0

 Protocol Discriminator: Q.931 (8)  len=10
> Call Ref: len= 2 (reference 48764/0xBE7C) (Terminator)
> Message type: CALL PROCEEDING (2)
> [18 03 a9 83 8e]
> Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, 
Exclusive Dchan: 0

>ChanSel: Reserved
>   Ext: 1  Coding: 0   Number Specified   Channel 
Type: 3

>   Ext: 1  Channel: 14 ]
   -- Accepting voice call from '323801' to '800978' on channel 
0/14, span 4

   -- Executing SetVar("Zap/14-1", "SERVER_ID=2") in new stack
   -- Executing Answer("Zap/14-1", "") in new stack
> Protocol Discriminator: Q.931 (8)  len=14
> Call Ref: len= 2 (reference 48764/0xBE7C) (Terminator)
> Message type: CONNECT (7)
> [18 03 a9 83 8e]
> Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, 
Exclusive Dchan: 0

>ChanSel: Reserved
>   Ext: 1  Coding: 0   Number Specified   Channel 
Type: 3

>   Ext: 1  Channel: 14 ]
> [1e 02 81 82]
> Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard 
(0) 0: 0   Location: Private network serving the local user (1)
>   Ext: 1  Progress Description: Called 
equipment is non-ISDN. (2) ]

   -- Executing AGI("Zap/14-1", "incoming_call.pl") in new stack
   -- Launched AGI Script /var/lib/asterisk/agi-bin/incoming_call.pl
< Protocol Discriminator: Q.931 (8)  len=9
< Call Ref: len= 2 (reference 15996/0x3E7C) (Originator)
< Message type: RELEASE (77)
< [08 02 83 86]
< Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0   
Location: Transit network (3)
<  Ext: 1  Cause: Channel unacceptable (6), class = 
Normal Event (0) ]

-- Processing IE 8 (cs0, Cause)
   -- Channel 0/14, span 4 got hangup
< Protocol Discriminator: Q.931 (8)  len=13
< Call Ref: len= 2 (reference 15996/0x3E7C) (Originator)
< Message type: STATUS (125)
< [08 03 83 e5 07]
< Cause (len= 5) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0   
Location: Tra

Re: SV: [Asterisk-Users] periodic-announce not working

[BJ Weschke]

On 6/22/06, Michiel van Baak <[EMAIL PROTECTED]> wrote:

On Jun 22, 2006, at 5:22 PM, BJ Weschke wrote:
>
> Submit a feature request/patch to bugs.digium.com. There isn't
> presently a way to do hold time announcements without queue position
> along with it.

I dont want hold time neither.
I simply want 1 file played to the ppl waiting in queue every 25
seconds.

I think I can fix it by creating a soundfile with 1 ms silence and
put that as position soundfiles in every queue, but that's ugly.

If that is what it takes, I will go implement it in my dialplan and
look into app_queue.c later



Then I'd say that's a bug and it needs to be fixed. Please post a bug
to bugs.digium.com. Thanks.

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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RE: [Asterisk-Users] Out of Office Auto Reply:

[Colin Anderson]

He's probably using Exchange which has a global setting to either send OOO
replies to SMTP addresses or not. It's a dumbass Exchange administrator who
enables this option (it is actually on by default)

Same thing happened to the mac-asterisk list last week, except the OOO
message would reply to every post, and to every user. In a period of 2 or 3
hours, I got a couple of thousand OOO replies from the offender. The
solution was to unsubscribe since the list owner was out of town apparently,
and so the list members did just that - I think the only guy that was still
subscribed once the dust settled was the list owner and the tool that set up
the OOO message in the first place. 

-Original Message-
From: Anthony Rodgers [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 22, 2006 10:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Out of Office Auto Reply: 


Actually, if his MTA is configured properly, it shouldn't happen at 
all.

A.

On Jun 22, 2006, at 9:32 AM, Doug Geary wrote:

> Should only happen once if his email system is config'd in a standard
> method. Otherwise just *plonk* his address.
>
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Dean Collins
> > Sent: Thursday, June 22, 2006 12:03 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Cc: [EMAIL PROTECTED]
> > Subject: RE: [Asterisk-Users] Out of Office Auto Reply:
> >
> > You got to be freaking kidding, a month of this?
> > Cant we get an easy process for the list owner to take care of these?

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Re: [Asterisk-Users] voip to voip bridge

[Benoît Mérouze]

Hi,

I've got some problems with bridged calls, the quality is extremely poor 
(more or less blanks or one way voice issues). But if I do a normal call 
with the same provider, there is no problem.


Reinvite is enabled, but what are the parameters in the dial command 
that force asterisk to stay in the loop ?
Are the H (to allow caller to hang up by dialing *) or L (to limit the 
call) parameters ones of them ?


As an example, here is a Dial command I execute to bridge a call to a 
new one :

SIP/kddi/0033172699611|30|HL(162:6:3)

Thanks,
Benoit



[EMAIL PROTECTED] wrote:


Hi,

Check if reinvites are enabled, and that you don’t use any parameter 
in the dial command that forces asterisk to stay in the loop.


Ohad



*From:* [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] *On Behalf Of *Erick Baum

*Sent:* Wednesday, June 14, 2006 5:00 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [Asterisk-Users] voip to voip bridge

Has anyone had any good experiences with a voip to voip bridge... 
where you have an incoming call on a voip line which is redirected out 
another voip line to a regular phone line? Whenever we do this, the 
connected call is kinda lagged and the quality isn't always that 
great. It seems to me this is just a problem with the inherent delay 
in the voip connections. But I was wondering if there's any special 
configurations that could make the situation better?


Erick




--
Benoît Mérouze
_._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._.
Groupe IPercom - The VoIP Enabling Company -  http://www.ipercom.com
Ingénieur R&D - courriel : [EMAIL PROTECTED]
Network Software Developer - mailto: [EMAIL PROTECTED]
Tél. / Phone : +33 1 7269 9611
._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._
Siège Social 43, rue Fessart 92100 Boulogne Billancourt
RCS NANTERRE B 440 345 528 - Capital social: 100 000 €
CE COURRIEL COMME LES DOCUMENTS EVENTUELLEMENT ASSOCIES
SONT CONFIDENTIELS & COUVERTS PAR LE SECRET PROFESSIONNEL

THIS E MAIL AND ANY DOCUMENT POSSIBLY ATTACHED ARE
CONFIDENTIAL AND COVERED BY THE PROFESSIONAL SECRECY
._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._
"Only two things are infinite, the universe and human stupidity, and I'm
 not sure about the former."
Albert Einstein

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[Asterisk-Users] New VICIDIAL astGUIclient Release: 1.1.12

[Matt Florell]

Hello,

We've released another update to our Asterisk GUI Client suite: 1.1.12

http://astguiclient.sourceforge.net/

The client suite runs on most modern web browsers on almost any
GUI-capable operating system, and it includes the astGUIclient
client-side web app which extends your phone's functionality and the
VICIDIAL client-side web app inbound/outbound call center software
suite.
This package is free as in GPL.
(the suite is not an asterisk configuration tool)
This package is geared towards Asterisk installations with SIP,IAX or
Zap phones and Zaptel, IAX or SIP trunks.

For this revision, we have fixed some serious security issues, several
bugs and have added many new features like customizable call time
settings, an internal DNC list, automated lead recycling and FTC
safe-harbor compliance just to name a few. We have also tested the
suite on Asterisk versions through 1.2.9.1

All client web-apps and administration pages are available in English,
Spanish, Greek and German, with rough translations of French,
Brazillian Portuguese, Italian and Portuguese for the client web-apps
only.

Check out the project blog for more information:
http://astguiclient.blogspot.com

Let me know what you think.

Thanks,


MATT---
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Re: [Asterisk-Users] South Africa DIDs

[Steve Kennedy]

On Thu, Jun 22, 2006 at 05:47:47PM +0100, Steve Kennedy wrote:

> Is it possible to get Joburg DIDs (probably need 4 at the moment), to be
> delivered via SIP preferrably to UK.
> If it's legal, please send pricing.

And that should have gone to the biz list, sorry.


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
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[Asterisk-Users] Quality monitoring

[Curt Shaffer]

Does anyone out there have a recommendation for tools that
will monitor the quality of VoIP systems? I am looking for jitter and MOS
monitoring. I have a custom Nagios plugin that is alerting me if the jitter
jumps out of a 20ms but I am looking for a little more detail. I would not be
against writing something in Perl for Nagios to do but I don’t really
know where to start on measuring jitter other than with ICMP pulls and really
don’t know where to start with doing MOS. 

 

Any ideas?

 

Thanks

 

Curt






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Re: [Asterisk-Users] Out of Office Auto Reply:

[Anthony Rodgers]

Actually, if his MTA is configured properly, it shouldn't happen at 
all.


A.

On Jun 22, 2006, at 9:32 AM, Doug Geary wrote:


Should only happen once if his email system is config'd in a standard
method. Otherwise just *plonk* his address.

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Dean Collins
> Sent: Thursday, June 22, 2006 12:03 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Cc: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] Out of Office Auto Reply:
>
> You got to be freaking kidding, a month of this?
> Cant we get an easy process for the list owner to take care of these?


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[Asterisk-Users] South Africa DIDs

[Steve Kennedy]

Is it possible to get Joburg DIDs (probably need 4 at the moment), to be
delivered via SIP preferrably to UK.

If it's legal, please send pricing.


Thanks


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
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RE: [Asterisk-Users] Out of Office Auto Reply:

[Doug Geary]

Should only happen once if his email system is config'd in a standard
method. Otherwise just *plonk* his address. 

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Dean Collins
> Sent: Thursday, June 22, 2006 12:03 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Cc: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] Out of Office Auto Reply:
> 
> You got to be freaking kidding, a month of this?
> Cant we get an easy process for the list owner to take care of these?
> 
> 
> 
> 
> 
> > -Original Message-
> > From: [EMAIL PROTECTED]
[mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of
> [EMAIL PROTECTED]
> > Sent: Thursday, 22 June 2006 11:45 AM
> > To: asterisk-users@lists.digium.com
> > Subject: [Asterisk-Users] Out of Office Auto Reply:
> >
> >
> > I will be on vacation from <22/06/06> to <30/06/06>.
> >
> > I will not be reachable on my mobile. I will have limited access to
> mails, and please
> > expect a delayed response.
> >
> > In my absence, please contact the following:
> > Ray Richard or Safeer Mohammed
> >
> > Thanks
> > H.Gireesh
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Re: [Asterisk-Users] when I press "transfer" -> blind -> 700 . The user is not able to hear what extension the call was parked on

[sdgesa gaeharth]

Title: Re:  [Asterisk-Users] when I press "transfer" -> blind -> 700 . The  user is not able to hear what extension the call was parked on
I have blindxfer  => #1 set  in features.Doesn't this means #1 is the same as  transfer -> blind, correct? Both are blind transfers..Is so, why when I transfer using #1 do I hear what extension the call was parked at but not transfer -> blind?James Texter <[EMAIL PROTECTED]> wrote:  This  is the way blind transfers work.  The transferring party doesn’t  get to hear anything.  For call parking,
 you have no choice but to  use supervised transfer if you want the user to hear the parking space.   If it worked before, it must have been dumb luck with the timing.  On 6/22/06 10:24 AM, "sdgesa gaeharth" <[EMAIL PROTECTED]> wrote:I am using Polycom 501s with asterisk 1.2.4.        When transfering to call parking wih "#1" -> 700 the  user is able to  hear asterisk tell him what extension the call  was parked on.   However, when I press "transfer" -> blind  -> 700 . The user is  not able to hear what extension the call  was parked on. It seems like  the polycom is hanging up before  asterisk is able to finish telling the  user the extension. I can  not tell if this is a problem with the phone,  asterisk, transfers  or call parking. This problem just started
  happening a few weeks  ago.  Before then , blind transfer worked  fine. It must be a  config issue somewhere        using "#1" -> 700:    -- Started music on hold, class 'default', on channel 'Zap/1-1'  == Parked Zap/1-1 on 701. Will timeout back to extension [incoming] s, 1 in 45 seconds    -- Added extension '701' priority 1 to parkedcalls    -- Playing 'digits/7' (language 'en')     -- Executing ParkedCall("SIP/1000-300e", "701") in new stack    -- Stopped music on hold on Zap/1-1    -- Channel SIP/1000-300e connected to parked call 701    -- Hungup 'Zap/1-1'        using "transfer" -> blind -> 700 
   -- Started music on hold, class 'default', on channel 'Zap/1-1'  == Parked Zap/1-1 on 701. Will timeout back to extension [incoming] s, 1 in 45 seconds    -- Added extension '701' priority 1 to parkedcalls    -- Playing 'digits/7' (language 'en')    -- Playing 'digits/0' (language 'en')    -- Playing 'digits/1' (language 'en')  == Spawn extension (local-access, 97037551131, 1) exited non-zero on 'SIP/1000-d779'    -- Executing ParkedCall("SIP/1000-5f5a", "701") in new stack    -- Stopped music on hold on Zap/1-1    -- Channel SIP/1000-5f5a connected to parked call 701    -- Started
 music on hold, class 'default', on channel 'Zap/1-1'    -- Stopped music on hold on Zap/1-1            Even though the user can not hear the extension the call  was parked on,  the call can be retrieved by guessing. Which I am  assumming means the  call was successfully parked.        Digit map:    [2-9]11|0T|011xxx.T|[0-1][2-9]x|[2-9]x|[2-9]xxxT|1xxxT        extensions.conf:    [general]    static=yes    writeprotect=no    autofallthrough=yes    clearglobalvars=no    priorityjumping=no        [globals]    ATTENDANT=SIP/1006&SIP/1002&SIP/1011&SIP/1009    OUTBOUNDTRUNK=ZAP/g1        [meetme-ext]    exten =>
 600,1,MeetMe(1234|Mp|98765)        [extentions]    include => parkedcalls    include => meetme-ext    include => direct-to-voicemail    exten => _10XX,1,Dial(SIP/${EXTEN},20,t)    exten => _10XX,n,Answer    exten => _10XX,n,VoiceMail([EMAIL PROTECTED])    exten => _10XX,n,Hangup()        [voicemail]    exten => _910XX,1,Wait(1)    exten => _910XX,n,VoiceMailMain(${EXTEN:[EMAIL PROTECTED])        [direct-to-voicemail]    exten => _810XX,1,VoiceMail(u${EXTEN:[EMAIL PROTECTED])    exten => _810XX,n,Hangup()        [local]    include => extentions    include => voicemail        [incoming]    exten =>
 s,1,Answer    exten => s,n,Wait(2)    exten => s,n,Set(TIMEOUT(response)=15)    exten => s,n,Background(intro)    exten => s,n,WaitExten()    exten => s,n,Playback(vm-goodbye)    exten => s,n,Hangup()    exten => 0,1,Dial(${ATTENDANT},20,t)    exten => 0,n,Playback(vm-nobodyavail)    exten => 0,n,Hangup()    exten => 1,1,Directory(voicemail,extentions,f)    exten => 2,1,Directory(voicemail,extentions)    include => meetme-ext    include => extentions    exten => i,1,Playback(pbx-invalid)    exten => i,2,Goto(incoming,s,1)    exten => t,1,Playback(vm-goodbye)    exten => t,2,Hangup()        [outbound]    ignorepat => 9    include
 => parkedcalls    exten => _9XX,1,Dial(${OUTBOUNDTRUNK}/ww${EXTEN:1},30,T)    exten => _9XX,2,Congestion()    exten => _9XX,102,Congestion()    exten => _91900NXX,1,Congestion()    exten => _91976NXX,1,Congestion()    exten => _91[123456789]XXNXX,1,Dial(${OUTBOUNDTRUNK}/ww${EXTEN:1},30,T)    exten => _91[123456789]XXNXX,2,Congestion()    exten => _91[123456789]XXNXX,102,Congestion()    exten => 9911,1,Dial(${OUTBOUNDTRUNK}/ww911)    exten => 9411,1,Dial(${OUTBO

Re: SV: [Asterisk-Users] periodic-announce not working

[Michiel van Baak]

On Jun 22, 2006, at 5:22 PM, BJ Weschke wrote:


Submit a feature request/patch to bugs.digium.com. There isn't
presently a way to do hold time announcements without queue position
along with it.


I dont want hold time neither.
I simply want 1 file played to the ppl waiting in queue every 25  
seconds.


I think I can fix it by creating a soundfile with 1 ms silence and  
put that as position soundfiles in every queue, but that's ugly.


If that is what it takes, I will go implement it in my dialplan and  
look into app_queue.c later


Michiel
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RE: [Asterisk-Users] Out of Office Auto Reply:

[Dean Collins]

You got to be freaking kidding, a month of this?
Cant we get an easy process for the list owner to take care of these?





> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
> Sent: Thursday, 22 June 2006 11:45 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Out of Office Auto Reply:
> 
> 
> I will be on vacation from <22/06/06> to <30/06/06>.
> 
> I will not be reachable on my mobile. I will have limited access to
mails, and please
> expect a delayed response.
> 
> In my absence, please contact the following:
> Ray Richard or Safeer Mohammed
> 
> Thanks
> H.Gireesh
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Re: [Asterisk-Users] when I press "transfer" -> blind -> 700 . The user

[Doug Lytle]

sdgesa gaeharth wrote:

I am using Polycom 501s with asterisk 1.2.4.

When transfering to call parking wih "#1" -> 700 the user is able to 
hear asterisk tell him what extension the call was parked on.  
However, when I press "transfer" -> blind -> 700 . The user is not 
able to hear what extension the call was parked on. It seems like the 
polycom is hanging up before asterisk is able to finish telling the 
user the extension. I can not tell if this is a problem with 


Don't press the blind button.  Do a Transfer, 700, send.  You'll here 
the parked extension and then press Transfer.


Doug

--

Ben Franklin quote:

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


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Re: [Asterisk-Users] when I press "transfer" -> blind -> 700 . The user is not able to hear what extension the call was parked on

[Andrew Kohlsmith]

On Thursday 22 June 2006 11:24, sdgesa gaeharth wrote:
> transfers or call parking. This problem just started  happening a few weeks
> ago.  Before then , blind transfer worked  fine. It must be a config issue
> somewhere

What did you change?  Can you roll back and get it to work properly again?  If 
so, get packet traces of the communication between the phone and Asterisk and 
compare them.

-A.
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[Asterisk-Users] CDRTool / asterisk billing based on realtime

[hgaillac-sip]

Hello,

I read ag-projects it does'not provide support for its
cdrtool pseudo gpl licensed .

It's one of the bad open source project even seen .
No documentation ...

A real bad project !!

Don't waste time with that

Anybody could advise me a billing system based on cdr
records with realtime support cdr_obdc.so?


Regards

Harry































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RE: [Asterisk-Users] when I press "transfer" -> blind -> 700 . The useris not able to hear what extension the call was parked on

[Brian Vincent \(C\)]

> However, when I press "transfer" -> blind -> 700 . 
> The user is not able to hear what extension the call 
> was parked on.

It's blind - so it's working as expected.  On a blind transfer the phone
set disconnects as soon as you press "blind".  Just make sure you do a
supervised transfer instead. 

---
Brian Vincent
Copper Mountain Telecom
[EMAIL PROTECTED] 
 
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[Asterisk-Users] Out of Office Auto Reply:

[Gireesh . Hariharasubramani]

I will be on vacation from <22/06/06> to <30/06/06>.

I will not be reachable on my mobile. I will have limited access to mails, and 
please expect a delayed response.

In my absence, please contact the following:
Ray Richard or Safeer Mohammed

Thanks
H.Gireesh
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[Asterisk-Users] Sharing experiences

[Christophe Ngo Van Duc]

Hi guys,

  Not a so technical question for the moment :) but I would like to share
some experience with people that have done some deployments on:
  - big call center (>300 call agents), 4 queues, high availability
  - SER and Asterisk integration
  - vPBX

  Feel free to contact me directly.

Best regards,
Christophe.


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Re: [Asterisk-Users] when I press "transfer" -> blind -> 700 . The user is not able to hear what extension the call was parked on

[James Texter]

Title: Re: [Asterisk-Users] when I press "transfer" -> blind -> 700 . The user is not able to hear what extension the call was parked on



This is the way blind transfers work.  The transferring party doesn’t get to hear anything.  For call parking, you have no choice but to use supervised transfer if you want the user to hear the parking space.  If it worked before, it must have been dumb luck with the timing.


On 6/22/06 10:24 AM, "sdgesa gaeharth" <[EMAIL PROTECTED]> wrote:

I am using Polycom 501s with asterisk 1.2.4.
  
  When transfering to call parking wih "#1" -> 700 the user is able to  hear asterisk tell him what extension the call was parked on.   However, when I press "transfer" -> blind -> 700 . The user is  not able to hear what extension the call was parked on. It seems like  the polycom is hanging up before asterisk is able to finish telling the  user the extension. I can not tell if this is a problem with the phone,  asterisk, transfers or call parking. This problem just started  happening a few weeks ago.  Before then , blind transfer worked  fine. It must be a config issue somewhere
  
  using "#1" -> 700:
  -- Started music on hold, class 'default', on channel 'Zap/1-1'
== Parked Zap/1-1 on 701. Will timeout back to extension [incoming] s, 1 in 45 seconds
  -- Added extension '701' priority 1 to parkedcalls
  -- Playing 'digits/7' (language 'en')
   -- Executing ParkedCall("SIP/1000-300e", "701") in new stack
  -- Stopped music on hold on Zap/1-1
  -- Channel SIP/1000-300e connected to parked call 701
  -- Hungup 'Zap/1-1'
  
  using "transfer" -> blind -> 700
  -- Started music on hold, class 'default', on channel 'Zap/1-1'
== Parked Zap/1-1 on 701. Will timeout back to extension [incoming] s, 1 in 45 seconds
  -- Added extension '701' priority 1 to parkedcalls
  -- Playing 'digits/7' (language 'en')
  -- Playing 'digits/0' (language 'en')
  -- Playing 'digits/1' (language 'en')
== Spawn extension (local-access, 97037551131, 1) exited non-zero on 'SIP/1000-d779'
  -- Executing ParkedCall("SIP/1000-5f5a", "701") in new stack
  -- Stopped music on hold on Zap/1-1
  -- Channel SIP/1000-5f5a connected to parked call 701
  -- Started music on hold, class 'default', on channel 'Zap/1-1'
  -- Stopped music on hold on Zap/1-1
  
  
  Even though the user can not hear the extension the call was parked on,  the call can be retrieved by guessing. Which I am assumming means the  call was successfully parked.
  
  Digit map:
  [2-9]11|0T|011xxx.T|[0-1][2-9]x|[2-9]x|[2-9]xxxT|1xxxT
  
  extensions.conf:
  [general]
  static=yes
  writeprotect=no
  autofallthrough=yes
  clearglobalvars=no
  priorityjumping=no
  
  [globals]
  ATTENDANT=SIP/1006&SIP/1002&SIP/1011&SIP/1009
  OUTBOUNDTRUNK=ZAP/g1
  
  [meetme-ext]
  exten => 600,1,MeetMe(1234|Mp|98765)
  
  [extentions]
  include => parkedcalls
  include => meetme-ext
  include => direct-to-voicemail
  exten => _10XX,1,Dial(SIP/${EXTEN},20,t)
  exten => _10XX,n,Answer
  exten => _10XX,n,VoiceMail([EMAIL PROTECTED])
  exten => _10XX,n,Hangup()
  
  [voicemail]
  exten => _910XX,1,Wait(1)
  exten => _910XX,n,VoiceMailMain(${EXTEN:[EMAIL PROTECTED])
  
  [direct-to-voicemail]
  exten => _810XX,1,VoiceMail(u${EXTEN:[EMAIL PROTECTED])
  exten => _810XX,n,Hangup()
  
  [local]
  include => extentions
  include => voicemail
  
  [incoming]
  exten => s,1,Answer
  exten => s,n,Wait(2)
  exten => s,n,Set(TIMEOUT(response)=15)
  exten => s,n,Background(intro)
  exten => s,n,WaitExten()
  exten => s,n,Playback(vm-goodbye)
  exten => s,n,Hangup()
  exten => 0,1,Dial(${ATTENDANT},20,t)
  exten => 0,n,Playback(vm-nobodyavail)
  exten => 0,n,Hangup()
  exten => 1,1,Directory(voicemail,extentions,f)
  exten => 2,1,Directory(voicemail,extentions)
  include => meetme-ext
  include => extentions
  exten => i,1,Playback(pbx-invalid)
  exten => i,2,Goto(incoming,s,1)
  exten => t,1,Playback(vm-goodbye)
  exten => t,2,Hangup()
  
  [outbound]
  ignorepat => 9
  include => parkedcalls
  exten => _9XX,1,Dial(${OUTBOUNDTRUNK}/ww${EXTEN:1},30,T)
  exten => _9XX,2,Congestion()
  exten => _9XX,102,Congestion()
  exten => _91900NXX,1,Congestion()
  exten => _91976NXX,1,Congestion()
  exten => _91[123456789]XXNXX,1,Dial(${OUTBOUNDTRUNK}/ww${EXTEN:1},30,T)
  exten => _91[123456789]XXNXX,2,Congestion()
  exten => _91[123456789]XXNXX,102,Congestion()
  exten => 9911,1,Dial(${OUTBOUNDTRUNK}/ww911)
  exten => 9411,1,Dial(${OUTBOUNDTRUNK}/ww411)
  exten => 0,1,Dial(${OUTBOUNDTRUNK}/ww0)
  
  [local-access]
  include => local
  include => outbound
  
  
  features.conf:
  [general]
  parkext =>  700   ; What ext. to dial to park
  parkpos =>  701-720   ; What extensions to park calls on
  context => parkedcalls  ; Which context parked calls are in
  parkingtime =>  45    ; Number of seconds a call can be pa

[Asterisk-Users] disconnect with mute

[Will Glass-Husain]

Hi,

I'm having problems with an occasional disconnect from phone calls while 
my phone is on mute.  This is a problem with long conference calls, for 
example.  I've a GrandStream GXP-2000 and Asterisk 1.2.1.  Anyone have 
experience with similar issues?


Best, WILL

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[Asterisk-Users] when I press "transfer" -> blind -> 700 . The user is not able to hear what extension the call was parked on

[sdgesa gaeharth]

I am using Polycom 501s with asterisk 1.2.4.When transfering to call parking wih "#1" -> 700 the user is able to  hear asterisk tell him what extension the call was parked on.   However, when I press "transfer" -> blind -> 700 . The user is  not able to hear what extension the call was parked on. It seems like  the polycom is hanging up before asterisk is able to finish telling the  user the extension. I can not tell if this is a problem with the phone,  asterisk, transfers or call parking. This problem just started  happening a few weeks ago.  Before then , blind transfer worked  fine. It must be a config issue somewhereusing "#1" -> 700:      -- Started music on hold, class 'default', on channel 'Zap/1-1'    == Parked Zap/1-1 on 701. Will timeout back to extension [incoming] s, 1 in 45 seconds      -- Added extension '701' priority 1 to parkedcalls      --
 Playing 'digits/7' (language 'en')   -- Executing ParkedCall("SIP/1000-300e", "701") in new stack      -- Stopped music on hold on Zap/1-1      -- Channel SIP/1000-300e connected to parked call 701      -- Hungup 'Zap/1-1'using "transfer" -> blind -> 700      -- Started music on hold, class 'default', on channel 'Zap/1-1'    == Parked Zap/1-1 on 701. Will timeout back to extension [incoming] s, 1 in 45 seconds      -- Added extension '701' priority 1 to parkedcalls      -- Playing 'digits/7' (language 'en')      -- Playing 'digits/0' (language 'en')      -- Playing 'digits/1' (language 'en')    == Spawn extension (local-access, 97037551131, 1) exited non-zero on 'SIP/1000-d779'      -- Executing ParkedCall("SIP/1000-5f5a", "701") in new stack 
     -- Stopped music on hold on Zap/1-1      -- Channel SIP/1000-5f5a connected to parked call 701      -- Started music on hold, class 'default', on channel 'Zap/1-1'      -- Stopped music on hold on Zap/1-1  Even though the user can not hear the extension the call was parked on,  the call can be retrieved by guessing. Which I am assumming means the  call was successfully parked.Digit map:  [2-9]11|0T|011xxx.T|[0-1][2-9]x|[2-9]x|[2-9]xxxT|1xxxTextensions.conf:  [general]  static=yes  writeprotect=no  autofallthrough=yes  clearglobalvars=no  priorityjumping=no[globals]  ATTENDANT=SIP/1006&SIP/1002&SIP/1011&SIP/1009  OUTBOUNDTRUNK=ZAP/g1[meetme-ext]  exten => 600,1,MeetMe(1234|Mp|98765)[extentions]  include => parkedcalls  include =>
 meetme-ext  include => direct-to-voicemail  exten => _10XX,1,Dial(SIP/${EXTEN},20,t)  exten => _10XX,n,Answer  exten => _10XX,n,VoiceMail([EMAIL PROTECTED])  exten => _10XX,n,Hangup()[voicemail]  exten => _910XX,1,Wait(1)  exten => _910XX,n,VoiceMailMain(${EXTEN:[EMAIL PROTECTED])[direct-to-voicemail]  exten => _810XX,1,VoiceMail(u${EXTEN:[EMAIL PROTECTED])  exten => _810XX,n,Hangup()[local]  include => extentions  include => voicemail[incoming]  exten => s,1,Answer  exten => s,n,Wait(2)  exten => s,n,Set(TIMEOUT(response)=15)  exten => s,n,Background(intro)  exten => s,n,WaitExten()  exten => s,n,Playback(vm-goodbye)  exten => s,n,Hangup()  exten => 0,1,Dial(${ATTENDANT},20,t)  exten => 0,n,Playback(vm-nobodyavail)  exten => 0,n,Hangup()  exten =>
 1,1,Directory(voicemail,extentions,f)  exten => 2,1,Directory(voicemail,extentions)  include => meetme-ext  include => extentions  exten => i,1,Playback(pbx-invalid)  exten => i,2,Goto(incoming,s,1)  exten => t,1,Playback(vm-goodbye)  exten => t,2,Hangup()[outbound]  ignorepat => 9  include => parkedcalls  exten => _9XX,1,Dial(${OUTBOUNDTRUNK}/ww${EXTEN:1},30,T)  exten => _9XX,2,Congestion()  exten => _9XX,102,Congestion()  exten => _91900NXX,1,Congestion()  exten => _91976NXX,1,Congestion()  exten => _91[123456789]XXNXX,1,Dial(${OUTBOUNDTRUNK}/ww${EXTEN:1},30,T)  exten => _91[123456789]XXNXX,2,Congestion()  exten => _91[123456789]XXNXX,102,Congestion()  exten => 9911,1,Dial(${OUTBOUNDTRUNK}/ww911)  exten => 9411,1,Dial(${OUTBOUNDTRUNK}/ww411)  exten =>
 0,1,Dial(${OUTBOUNDTRUNK}/ww0)[local-access]  include => local  include => outbound  features.conf:  [general]  parkext =>  700   ; What ext. to dial to park  parkpos =>  701-720   ; What extensions to park calls on  context => parkedcalls  ; Which context parked calls are in  parkingtime =>  45    ; Number of seconds a call can be parked for[featuremap]  blindxfer => #1Thanks   
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