date:20071024

Joseph Begumisa wrote:

 Has anyone had any compatibility issues with a TE110P card installed 
 on a Dell Poweredge 1950?  I noted the following error on the LCD 
 display of the Dell Poweredge 1950:

  

 E1711 PCI PErr Slot 1 E171F PCIE Fatal Error B0 D4 F0.

  

 The Dell hardware owners manual states that it means the system BIOS 
 has reported a PCI parity error on a component that resides in PCI 
 configuration space at bus 0, device 4, function 0 and advises that 
 the PCI expansion card be removed and reseated.

  

 Any suggestions on what exactly might be causing this are welcome.

  

 Thanks.

  

 Joseph

My guess would be that the Digium card is causing the issue although you 
would probably be led to believe that the Dell is not compatible with 
the card and not visa versa.

It would be interesting to see if a Sangoma board would have that same 
issue.  I have not had any of these compatibility issues since going 
Sangoma.

Is this an older card or one with the New and Improved Bus thing? 

Have you called Digium?

Thanks,
Steve

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[asterisk-users] Help with loop counting?

2007-10-24 Thread Phil Knighton

Hi
 
I have a situation where I want to be able to count how many times a
caller goes round a loop of Please hold..., please continue to hold.
I have found an example on voip-info but I can't get it to work.  Not
sure if I've got some syntax wrong somewhere?  All that happens at the
moment, is I hit is the playback of som-debug at . Any ideas would
be appreciated!
 
extensions.conf:
 
[so-mainmenu]
exten = s,1,Answer
exten = s,2,Set(trips=1)
exten = s,3,SetMusicOnHold(default)
exten = s,4,Set(TIMEOUT(digit)=5)
exten = s,5,Set(TIMEOUT(response)=10)
exten = s,6,Background(softopt/som-mainmenu)
exten = s,7,GotoIf($[${trips}=4]?,8)
exten = s,8,WaitExten(5)
exten = s,9,Wait(5)
exten = 1,1,Goto(so-sandm,s,1)
exten = 2,1,Goto(so-support,s,1)
exten = 3,1,Goto(so-accbill,s,1)
exten = 4,1,Goto(so-switchboard,s,1)
exten = 5,1,Goto(so-silentdial),s,1)
exten = s,10,Background(softopt/som-mainmenuretry)
exten = s,11,Wait(1)
exten = s,12,Background(softopt/som-mainmenuopts)
exten = s,13,Goto(s,7)
exten = ,1,Playback(softopt/som-debug)
exten = ,2,Hangup()
exten = i,1,Set(trips=$[${trips} + 1])
exten = i,2,Goto(s,7)
 
Cheers
 
Phil
 
Phil Knighton
Soft Option Technologies
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Re: [asterisk-users] Help with loop counting?

2007-10-24 Thread Il Neofita

Hi
I believe that
exten = s,7,GotoIf($[${trips}=4]?,8)

the , should be :

On 10/24/07, Phil Knighton [EMAIL PROTECTED] wrote:

  Hi

 I have a situation where I want to be able to count how many times a
 caller goes round a loop of Please hold..., please continue to hold.  I
 have found an example on voip-info but I can't get it to work.  Not sure if
 I've got some syntax wrong somewhere?  All that happens at the moment, is I
 hit is the playback of som-debug at . Any ideas would be appreciated!

 extensions.conf:

 [so-mainmenu]
 exten = s,1,Answer
 exten = s,2,Set(trips=1)
 exten = s,3,SetMusicOnHold(default)
 exten = s,4,Set(TIMEOUT(digit)=5)
 exten = s,5,Set(TIMEOUT(response)=10)
 exten = s,6,Background(softopt/som-mainmenu)
 exten = s,7,GotoIf($[${trips}=4]?,8)
 exten = s,8,WaitExten(5)
 exten = s,9,Wait(5)
 exten = 1,1,Goto(so-sandm,s,1)
 exten = 2,1,Goto(so-support,s,1)
 exten = 3,1,Goto(so-accbill,s,1)
 exten = 4,1,Goto(so-switchboard,s,1)
 exten = 5,1,Goto(so-silentdial),s,1)
 exten = s,10,Background(softopt/som-mainmenuretry)
 exten = s,11,Wait(1)
 exten = s,12,Background(softopt/som-mainmenuopts)
 exten = s,13,Goto(s,7)
 exten = ,1,Playback(softopt/som-debug)
 exten = ,2,Hangup()
 exten = i,1,Set(trips=$[${trips} + 1])
 exten = i,2,Goto(s,7)

 Cheers

 Phil

 Phil Knighton
 Soft Option Technologies

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Re: [asterisk-users] Grandstream GXP-2000's and Asterisk.

2007-10-24 Thread Thomas Kenyon

Thomas Kenyon wrote:
 I have 5 GXP 2000's with firmware 1.1.4.25 running with Asterisk 1.4.13.
 
 Is anyone else getting the following error in the asterisk console:
 
 [Oct 22 11:39:01] WARNING[7100]: rtp.c:1142 ast_rtp_read: RTP Read too short
 
 every couple of seconds when a handset is in a call?
 
 I didn't notice this happening when I was using an older GXP2000 with 
 the same firmware (doesn't mean that it didn't happen).
 
 The Call in question is using G.729.
 
 TIA for any help with this.
 
 I will hopefully get a bit more time to play with this today. (When I'm 
 in the office in question).
 
Changing codec doesn't appear to matter. I gather that the cause is that 
the GXP-2000 sends empty udp packets as keep-alives. (which is all well 
and good, but even with a handful of handsets with light call volume the 
logs fill up with notices, at the moment there is only 1 call going 
through the server and this is generating 2 notices/second.

Is there any way to make asterisk ignore the empty packets from certain 
peers?


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Re: [asterisk-users] register = to let Asterisk register to another softswitch via SIP

2007-10-24 Thread bilal ghayyad

Dear Alex;

Thanks for your great help and nice replies.

I would like to confirm that I understood your request
very well, so please advise me for the following:

1) If no need for registering asterisk with the
softswitch, then no need to use register = but we
will configure the section with type=peer and
host=softswitch_ipaddress, correct?

2) If no need to register asterisk with the
softswitch, then this kind of trunk is called trunk
tie and it is 'trusted', correct?

3) For receiving calls from the softswitch via the
trunk tie, then username and secret are not important
for the section configuration as the insecure=very,
correct?

From the other side, I do not know if it possible to
help me in another issue related also to registering
asterisk with another softswitch:

A) If nat=yes, then I have to set canreinvite=no to be
able to register, correct?

B) In case of using firefly softphone, how it possible
to set it to have nat=yes (at the firefly it self and
not at the sip user configuration section)? As most of
the sip endpoint give an option to set nat=yes and so
on, how it can be done with firefly softphone?

C) One time I succeed to register my asterisk on
another softswitch (sip registeration), but when I
routed calls via this IP Trunk, then the calls are not
deliver to the softswitch at all, and the error at
asterisk says that eveyone is bussy. I do not know
why? Registeration succeed but calls are not appear at
all on the softswitch screen. By the way: my Asterisk
still does not support G711 while the softswitch that
I am attempting to register with it support only G729
and G723, is that the reason that the call does not
appear on the softswitch (after registeration
complete)? Normally on that softswitch, when endpoints
are registering and they dot match the codec, then
calls are delviered and it appears on the softswitch
but it gives a message that codec miss match, but in
my case it does not even display and kind of receiving
the call from asterisk with fail or without, any
advise? Is it because my softswitch does not support
G711? I beleive it should process the call with fail
(codec miss match), but I do not see the call.

Looking to hear from you.
Regards
Bilal


Bilal,

On Tue, 23 Oct 2007, bilal ghayyad wrote:

 This is if I need to let Asterisk register with
another softswitch
 (so I 
 used register =), what if I need asterisk to send
call for the
 
 softswitch without register to it (directly)? If I
removed the
 register 
 = then how it will distiguish the IP address in the
host at
 the 
 [sip_trunk] is the IP address of the softswitch that
need to
 register
 
 with it and not the IP address of the original
caller sip
 endpoint?

   Unless I am missing something here, I suppose the
answer is that 
Asterisk can distinguish the IP endpoints because they
are ...
 distinct.

   Here is the essence of the situation:

   In Asterisk it is possible to peer with an endpoint
with and without
registrations.  Registrations are mostly intended for
dynamic endpoints
whose IP address can potentially change, such as
end-user phones off of
broadband connections, or other clients whose IP
address is not
 desirable
to track or cannot be trusted.

   The other type of peer is a 'trusted' trunk tied to
a particular IP 
endpoint on the far end.  The trust can be done only
by IP address,
or by IP address + SIP UDP port.  This type of peer is
typically used
when doing SIP handoff from origination and
termination carriers on any
kind of large-scale, or in other intra-industrial
and/or internal
 and/or
intra-platform SIP connections where it is not
desirable to position
 one
endpoint of the SIP trunk as a UAC (client)
registering against a UAS
(server) per se, as such, in the respect that one
challenges the other
for authentication credentials.

   So, what I would do is build a trusted trunk
(type=peer,
 insecure=very) 
to the softswitch that has a static IP (host=)
endpoint defined.  Then,
Asterisk can accept registrations from your users. 
Where to route the
call is determined entirely in the dial plan
(extensions.conf), where
you can send calls to particular SIP peers.  So, for
example, here is a
regular user defined in sip.conf:

[Alex_Evariste_2]
type=friend
host=dynamic
canreinvite=no
username=Alex_Evariste_2
secret=xx
nat=yes
allow=ulaw
qualify=yes
[EMAIL PROTECTED]
context=default-user-dial

   And here is a dedicated trunk to a provider:

[my_sip_provider]

host=xxx.yyy.zzz.www
insecure=very
type=peer
qualify=no
canreinvite=no
dtmfmode=rfc2833

   Then, your dial plan for a user can be set up like
this, for
 example,
in extensions.conf:

[default-user-dial]

; Any North American ten-digit number.

exten =
_NX,1,Dial(SIP/[EMAIL PROTECTED])

   In our case, we actually register with our SIP
origination provider,
 so 
we have this IP trunk:

[junction_networks]

fromdomain=jnctn.net
host=sip.jnctn.net
port=5060
insecure=very

[asterisk-users] Asterisk integration with IBM Sametime

2007-10-24 Thread reshmi.nair


Hi,
 
I wanted to know if anyone has experience in integration asterisk with
IBM Sametime server (by implementing TCSPI).
Any pointers for this would be very helpful. 
 
Have been reading/googling around a bit and I get to understand that the
communication between the Sametime server and Asterisk is SIP.
Wanted to know if my understanding is right. Since this is part of some
experiment I'm doing, I only have the trial version of Sametime Server
with me which doesn't have the Sametime Gateway component (and that is
what talks SIP). Just wanted to know if this means that I cannot
integrate asterisk with the trial version of Sametime server.
 
Would really help a lot, if someone clarifies my doubts.
 
Regards,
Reshmi



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Re: [asterisk-users] Help with loop counting?

2007-10-24 Thread Doug Lytle

Phil Knighton wrote:
 exten = i,1,Set(trips=$[${trips} + 1])
 exten = i,2,Goto(s,7)

i=invalid, t=timeout

exten = t,1,Set(trips=$[${trips} + 1])


You'll also want to initialize ${trips} with a Set(trips=0) at the beginning of 
your routine.

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] Snom 360 lights not working on subscription / fail to extend xx to xx error

2007-10-24 Thread Carlos Maimone

Dear guys,

many people have been using Snom with Subscription/notify lights I  
tried almost every tip in the google.
But there's one thing related to the snom phones and asterisk I  
didn't see in any forum
The Asterisk console show very often a message like:

fail to extend from xx to xxx

This message appears ver often and when snom phones do reboot or  
subscribe or while it receives notify messages.

Any idea?

BTW. in Snom's sip trace content length is 0 in every NOTIFY message  
received by phone and there's no XML
thanks and regards,


Carlos


On Oct 23, 2007, at 8:55 PM, Craig Guy wrote:

 The Linksys SPA962 with SPA932 sidecar support both speed dial and  
 BLF.
 IMHO very good for the money and very easy to provision once you  
 get a hold
 of the proper provisioning guide.  These things are designed for mass
 deployment and remote provisioning.  As other people have noted,  
 you need to
 provision via http rather than tftp for best effect.  I also have two
 provisioning files, a shared settings file with the bulk of the  
 config and
 then a per handset file based on the mac address containing the  
 account and
 any special customisations.  The only bad bit is that a resync usually
 causes a reboot of the handset which interrupts the connection of  
 anything
 attached to the PC port of the phone.

 Craig

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Omar  
 A. Sabek
 Sent: Tuesday, 23 October 2007 1:08 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Snom 360 lights not working on  
 subscription

 Hey Mike,

 We started deploying exclusively Polycom and Linksys. The Polycom's
 support presence, they call it 'Buddy List'. I am not sure about the
 Linksys phones, I don't think they do although I did see support for
 SLA (Shared Line Appearance).

 Omar

 On 10/23/07, Michael J. Liberatore  
 [EMAIL PROTECTED]
 wrote:
 I also have problems with these phones.  I have deployed many of them
 and have had nothing but problems.  Omar, what phones did you  
 switch to?
 I needed some of the features of the snom phones, like the multiple
 buttons with prescence lights.

 Mike



 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Omar A.
 Sabek
 Sent: Monday, October 22, 2007 9:27 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Snom 360 lights not working on
 subscription

 I used to deploy these phones, it was these types of issues that  
 forced
 me to drop it. It took way too long to troubleshoot the problems and
 there was a general lack of documentation. This was 2 years ago,  
 things
 might have changed. If I remember correctly, it was this issue you  
 are
 having that was the final straw.

 Good luck,

 Omar

 On 10/22/07, Carlos Maimone [EMAIL PROTECTED] wrote:
 Dear friends,

 I am working around with a Snom 360 and Asterisk 1.4 + FreePBX

 In order to get subscriptions working and the Snom 360 lights turns
 on, I have set everything just like all the pages in the net  
 explain.

 So, I get subsciption working. I can list subscription on the  
 asterisk

 and if I use the SIP trace function built in at the SNOM nad see
 NOTIFY messages and 200 OK responses. But I realized that content
 length = 0 in all messsages and there isn't any XML content in those
 Notify headers..


 any idea of what's going on?

 IN SNOM 360 I am currently using firmware 6.5.12

 I am pretty sick dealing with this issue.


 thanks and regards,


 Charlie

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Re: [asterisk-users] Asterisk integration with IBM Sametime

2007-10-24 Thread ram

On 10/24/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:

  Hi,

 I wanted to know if anyone has experience in integration asterisk with IBM
 Sametime server (by implementing TCSPI).
 Any pointers for this would be very helpful.

 Have been reading/googling around a bit and I get to understand that the
 communication between the Sametime server and Asterisk is SIP.
 Wanted to know if my understanding is right. Since this is part of some
 experiment I'm doing, I only have the trial version of Sametime Server with
 me which doesn't have the Sametime Gateway component (and that is what talks
 SIP). Just wanted to know if this means that I cannot integrate asterisk
 with the trial version of Sametime server.

 Would really help a lot, if someone clarifies my doubts.



Hi

what are you trying to achieve.

Integrating with Asterisk, Samtime server send the calls to asterisk ?

or asterisk expect to send calls to Samtime Server

ram
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Re: [asterisk-users] Asterisk integration with IBM Sametime

2007-10-24 Thread reshmi.nair


Hi,
 
I am trying to setup a conference between Sametime users using
conferencing infrastructure of asterisk.
 
Sametime server has a component called TCSPI, which we can implement to
talk to any PBX, including asterisk (as per documentation). I was trying
to implement the TCSPI for Asterisk.
 
Regards,
Reshmi



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of ram
Sent: Wednesday, October 24, 2007 6:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk integration with IBM Sametime




On 10/24/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: 

Hi,
 
I wanted to know if anyone has experience in integration
asterisk with IBM Sametime server (by implementing TCSPI).
Any pointers for this would be very helpful. 
 
Have been reading/googling around a bit and I get to understand
that the communication between the Sametime server and Asterisk is SIP. 
Wanted to know if my understanding is right. Since this is part
of some experiment I'm doing, I only have the trial version of Sametime
Server with me which doesn't have the Sametime Gateway component (and
that is what talks SIP). Just wanted to know if this means that I cannot
integrate asterisk with the trial version of Sametime server. 
 
Would really help a lot, if someone clarifies my doubts.
 

 
Hi
 
what are you trying to achieve.
 
Integrating with Asterisk, Samtime server send the calls to asterisk ?
 
or asterisk expect to send calls to Samtime Server
 
ram




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intended recipient, you should not disseminate, distribute or copy this e-mail. 
Please notify the sender immediately and destroy all copies of this message and 
any attachments. 

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check this email and any attachments for the presence of viruses. The company 
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Re: [asterisk-users] Compatibility Issues with dell poweredge 1950 and TE110P card

2007-10-24 Thread David Gomillion

On 10/24/07, Steve Totaro [EMAIL PROTECTED] wrote:

 Joseph Begumisa wrote:
 
  Has anyone had any compatibility issues with a TE110P card installed
  on a Dell Poweredge 1950?  I noted the following error on the LCD
  display of the Dell Poweredge 1950:
 
 
 
  E1711 PCI PErr Slot 1 E171F PCIE Fatal Error B0 D4 F0.


Yes, I have had this problem with a dell PE1650, 1850, SC1400, and PE650. I
have a TE410P that does it. It may not be wise, but I just ignore the orange
blinking LCD display (or light, depending on the model). I did try reseating
the card, and it works for a few weeks, and then goes back to the same old
thing.
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[asterisk-users] Internal LAN echo problem

2007-10-24 Thread Jonn R Taylor

Hi all,

I have an internal echo problem on my LAN only. I replaced the LAN 
switch with a new linksys 2024 with QOS and seemed to help but not fix 
the problem. Any ideas? Here in my setup - Dell PE6400 Dual 700, 
Asterisk 1.2.24/FreePBX, 2-NIC cards, one with a public ip and one with 
an internal ip, one PAP2, one SPA3102 and 2 BT101. I know that bt's are 
cheap that are known for echo problem in the handset. I have one remote 
user that never has a problem. I have a remote test server at home 
connect via IAX with no problems, also a PAP2 with no problem. External 
faxing from the rest of the world via our voip provider is working 
great. One strange thing that I noticed is that we can not fax to our 
iaxmodem, ATA --- iaxmodem, but works perfect ATA --- rx_fax. Not sure 
why either.

Jonn

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Re: [asterisk-users] asterisk and Skype - your experiences please

2007-10-24 Thread Tomislav Petrovic

randulo said on 24.10.2007 10:17:
 From time to time, various ways of connecting asterisk SIP channels to
 skype has been discussed here. This Friday, one of the subjects of our
 weekly Voip Users Conference will be about trying to connect our
 asterisk pbx with Skype.

Has anyone tried http://www.chanskype.com/
I know its not free, but would like to know if it works as adwertised?

-- 
Tomy


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Re: [asterisk-users] Polycom Phone and bitmaps

2007-10-24 Thread Doug Bailey


Shaun wrote:
I've been trying to get the polycom 550 phones to show a idle display bitmap 
but have not been successful.  Anybody have any experience with this?  The 
manual gives instructions 
(http://www.polycom.com/common/documents/support/setup_maintenance/products/voice/soundpoint_ip_soundstation_ip_administrators_guide_v2_2.pdf)
 
but they do not seam to work.  So far i've done the following in my sip.conf
 

Also beware that there is enough bitmap quota allocated to your machine.  (See 
quotas in the admin guide) I believe the 550 phone provides 10KB of bitmap 
space.  You should still see the bitmap served to the phone.  The phone throws 
the image away if it is too big. 

- Doug 


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[asterisk-users] OSLEC and zaptel-1.4.5.1

Hi all,

After reading great things about the OSLEC Echo Canceller 
(http://www.rowetel.com/ucasterisk/oslec) and seeing the reactions of 
people who have tried it on a recent Trixbox thread 
(http://www.trixbox.org/forums/trixbox-forums/open-discussion/need-people-echo-problems),
 
it sounds like it is the bees knees for sorting out echo problems with 
cards like the x100p.

Has anyone managed to get oslec to work with recent zaptel and kernel 
(I'm running 2.6.23)?

Lots of information below. Comments/suggestions welcome.

Having followed the instructions on the oslec site, and ensuring the 
patch for zaptel takes O.K (I manually installed the patch into the 
zaptel source tree just to make sure). I can build the oslec module, and 
build a patched zaptel-1.4.5.1-oslec without any compilation issues.

However when I reload the system during boot-up dmesg tells me:

Zapata Telephony Interface Registered on major 196
Zaptel Version: 1.4.5.1
Zaptel Echo Canceller: MG2
Zaptap registered 'sample' char driver on major 33 (This means the patch 
went in O.K.)
ACPI: PCI Interrupt :00:08.0[A] - GSI 17 (level, low) - IRQ 22
wcfxo: DAA mode is 'FCC'
Found a Wildcard FXO: Wildcard X100P

Notice the choice of echo canceller

If I look at what modules are installed:

# lsmod
Module  Size  Used by
zttranscode 6280  0
ztdummy 3432  0
wcfxo   9760  0
zaptel200120  7 zttranscode,ztdummy,wcfxo
crc_ccitt   1792  1 zaptel

No oslec :-(

In my kernel modules/misc directory I have:
-rw-r--r-- 1 root root 10727 2007-10-24 14:44 oslec.ko
-rw-r--r-- 1 root root 65372 2007-10-24 14:41 pciradio.ko
-rw-r--r-- 1 root root 91321 2007-10-24 14:41 tor2.ko
-rw-r--r-- 1 root root 18901 2007-10-24 14:41 torisa.ko
-rw-r--r-- 1 root root 12605 2007-10-24 14:41 wcfxo.ko
-rw-r--r-- 1 root root 15989 2007-10-24 14:41 wct1xxp.ko
drwxr-xr-x 2 root root  4096 2007-10-24 14:41 wct4xxp
drwxr-xr-x 2 root root  4096 2007-10-24 14:41 wctc4xxp
drwxr-xr-x 2 root root  4096 2007-10-24 14:41 wctdm24xxp
-rw-r--r-- 1 root root 41046 2007-10-24 14:41 wctdm.ko
-rw-r--r-- 1 root root 32882 2007-10-24 14:41 wcte11xp.ko
-rw-r--r-- 1 root root 45804 2007-10-24 14:41 wcte12xp.ko
-rw-r--r-- 1 root root 16527 2007-10-24 14:41 wcusb.ko
drwxr-xr-x 2 root root  4096 2007-10-24 14:41 xpp
-rw-r--r-- 1 root root 81616 2007-10-24 14:41 zaptel.ko
-rw-r--r-- 1 root root  8270 2007-10-24 14:41 ztd-eth.ko
-rw-r--r-- 1 root root  5530 2007-10-24 14:41 ztd-loc.ko
-rw-r--r-- 1 root root  5297 2007-10-24 14:41 ztdummy.ko
-rw-r--r-- 1 root root 11687 2007-10-24 14:41 ztdynamic.ko
-rw-r--r-- 1 root root  8639 2007-10-24 14:41 zttranscode.ko

My /etc/zaptel.conf is:
loadzone=uk
defaultzone=uk

fxsks=1

My /etc/asterisk/zapata.conf is

; Zapata telephony interface ;
; Configuration file
[channels]
;Hardware defaults for the x100p card
;usecallerid=yes
;hidecallerid=no
;callwaiting=no
;threewaycalling=yes
;usedistinctiveringdetection=yes
;transfer=yes
;usecallingpres=yes
;callwaitingcallerid=yes
;cancallforward=yes
;callreturn=yes
echocancel=yes
echotrainingwhenbridged=no
;echotraining=400
rxwink=300 ; Atlas seems to use long (250ms) winks

;cidsignalling=v23 ; Added for UK CLI detection
;cidstart=usehist ; After patching the driver from here :
; http://www.lusyn.com/resources/asterisk/usehist.htm
;callerid=asreceived ; propagate the CID received from BT
;rxgain=1.0
;txgain=1.0

;define channel
context=main_menu
language=en
signalling=fxs_ks
channel = 1 ;Our x100p

--

Alan

-- 
The way out is open!
http://www.theopensourcerer.com


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[asterisk-users] Unusual DTMF behavior

2007-10-24 Thread Jason Landrey

We are having an issue where DTMF is not being sent out right away and the
tone duration is inconsistent.  For a test we send a '5', then a second
later we send a '9', and then five seconds later we send a '5'.  If you look
at the logs below you can see the first '5' is played right away, then the
'9' comes in and gets queued, but it doesn't start playing until five
seconds later and it plays for six seconds.  Then the last '5' is played.

The DTMF is coming in as only 'end' packets and we can't change that.  For
this reason we have turned on rfc2833compensate.  Using Asterisk 1.4.11.

Any ideas?


asteriskpri04*CLI
 Protocol Discriminator: Q.931 (8)  len=5
 Call Ref: len= 2 (reference 499/0x1F3) (Terminator)
 Message type: CONNECT (7)
q931.c:3371 q931_receive: call 33267 on channel 3 enters state 10 (Active)
 Protocol Discriminator: Q.931 (8)  len=5
 Call Ref: len= 2 (reference 499/0x1F3) (Originator)
 Message type: CONNECT ACKNOWLEDGE (15)
[Oct 23 10:39:56] DEBUG[6136]: chan_zap.c:1413 zt_enable_ec: Echo
cancellation already on
-- Zap/3-1 answered SIP/test.com-08dc1ef8

[Oct 23 10:39:58] DTMF[13914]: channel.c:2346 __ast_read: DTMF end '5'
received on SIP/test.com-08dc1ef8, duration 0 ms
[Oct 23 10:39:58] DTMF[13914]: channel.c :2382 __ast_read: DTMF begin
emulation of '5' with duration 100 queued on SIP/test.com-08dc1ef8
[Oct 23 10:39:58] DEBUG[13914]: chan_zap.c:1048 zt_digit_begin: Started
VLDTMF digit '5'

[Oct 23 10:39:59] DTMF[13914]: channel.c:2346 __ast_read: DTMF end '9'
received on SIP/test.com-08dc1ef8, duration 0 ms
[Oct 23 10:39:59] DTMF[13914]: channel.c:2352 __ast_read: DTMF end '9' put
into dtmf queue on SIP/test.com-08dc1ef8

[Oct 23 10:39:59] DTMF[13914]: channel.c:2434 __ast_read: DTMF end emulation
of '5' queued on SIP/test.com-08dc1ef8
[Oct 23 10:39:59] DEBUG[13914]: chan_zap.c:1083 zt_digit_end: Ending VLDTMF
digit '5'

[Oct 23 10:40:04] DTMF[13914]: channel.c:2215 __ast_read: DTMF begin
emulation of '9' with duration 100 queued on SIP/test.com-08dc1ef8
[Oct 23 10:40:04] DEBUG[13914]: chan_zap.c:1048 zt_digit_begin: Started
VLDTMF digit '9'
[Oct 23 10:40:04] DTMF[13914]: channel.c:2346 __ast_read: DTMF end '5'
received on SIP/test.com-08dc1ef8, duration 0 ms
[Oct 23 10:40:04] DTMF[13914]: channel.c:2352 __ast_read: DTMF end '5' put
into dtmf queue on SIP/test.com-08dc1ef8

[Oct 23 10:40:10] DTMF[13914]: channel.c:2465 __ast_read: DTMF end emulation
of '9' queued on SIP/test.com-08dc1ef8
[Oct 23 10:40:10] DEBUG[13914]: chan_zap.c:1083 zt_digit_end: Ending VLDTMF
digit '9'
[Oct 23 10:40:10] DTMF[13914]: channel.c:2215 __ast_read: DTMF begin
emulation of '5' with duration 100 queued on SIP/test.com-08dc1ef8
[Oct 23 10:40:10] DEBUG[13914]: chan_zap.c:1048 zt_digit_begin: Started
VLDTMF digit '5'

[Oct 23 10:40:10] DTMF[13914]: channel.c:2465 __ast_read: DTMF end emulation
of '5' queued on SIP/test.com-08dc1ef8
[Oct 23 10:40:10] DEBUG[13914]: chan_zap.c:1083 zt_digit_end: Ending VLDTMF
digit '5'


Thanks,
Jason
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[asterisk-users] whisper chanspy in asterisk 1.2

2007-10-24 Thread Carles Pina i Estany


Hello,

I would like to have whisper channel spy (not private whisper) in
Asterisk 1.2. I see here:
http://www.the-asterisk-book.com/unstable/applikationen-chanspy.html
That is only available for Asterisk 1.4.

I wonder if there is any way to emulate it in Asterisk 1.2. For example,
to join two calls: one to use a private whisper and other one to use
a normal chanspy.

Thank you,

-- 
Carles Pina i EstanyGPG id: 0x8CBDAE64
http://pinux.info   Manresa - Barcelona

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[asterisk-users] How to tune Asterisk AMD - Answering Machine Detection hacks

2007-10-24 Thread Costa Dinoteli

Hello Everyone,

Can someone point me to reliable links on how to tweak Asterisk AMD
I am calling a number and have to two files to play depending if it is a
real person or an
answering machine.
Most everytime Asterisk calls  it thinks it is an Answering Machine and it
starts playing
the AMD message, instead of the delivering the 1st real message

Any hints?
Thanks in advance,
-C
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Re: [asterisk-users] OSLEC and zaptel-1.4.5.1

2007-10-24 Thread Dave Fullerton

Replies/Comments inline...

Alan Lord wrote:
 Hi all,
 
 After reading great things about the OSLEC Echo Canceller 
 (http://www.rowetel.com/ucasterisk/oslec) and seeing the reactions of 
 people who have tried it on a recent Trixbox thread 
 (http://www.trixbox.org/forums/trixbox-forums/open-discussion/need-people-echo-problems),
  
 it sounds like it is the bees knees for sorting out echo problems with 
 cards like the x100p.

I am using OSLEC on my home pbx. I used to have echo on some calls prior 
to OSLEC but have been echo free since I installed it.

 Has anyone managed to get oslec to work with recent zaptel and kernel 
 (I'm running 2.6.23)?

I'm only using 2.6.17 and zaptel-1.4.4 at the moment. But if the patches 
apply it should work.

 Having followed the instructions on the oslec site, and ensuring the 
 patch for zaptel takes O.K (I manually installed the patch into the 
 zaptel source tree just to make sure). I can build the oslec module, and 
 build a patched zaptel-1.4.5.1-oslec without any compilation issues.
 
 However when I reload the system during boot-up dmesg tells me:
 
 Zapata Telephony Interface Registered on major 196
 Zaptel Version: 1.4.5.1
 Zaptel Echo Canceller: MG2
 Zaptap registered 'sample' char driver on major 33 (This means the patch 
 went in O.K.)
 ACPI: PCI Interrupt :00:08.0[A] - GSI 17 (level, low) - IRQ 22
 wcfxo: DAA mode is 'FCC'
 Found a Wildcard FXO: Wildcard X100P
 
 Notice the choice of echo canceller

Check the zconfig.h file in the zaptel source and make sure that the line:
#define ECHO_CAN_OSLEC
is not commented out but all the lines for the other echo cancelers are.

Did you start with a clean source (or at least did a make clean) before 
you compiled? Are you using the zaptel-1.4.4.patch from the oslec SVN or 
some other patch?

 If I look at what modules are installed:
 
 # lsmod
 Module  Size  Used by
 zttranscode 6280  0
 ztdummy 3432  0
 wcfxo   9760  0
 zaptel200120  7 zttranscode,ztdummy,wcfxo
 crc_ccitt   1792  1 zaptel

Just for kicks, try inserting the oslec module by hand (insmod oslec) 
and see if that makes a difference.

   In my kernel modules/misc directory I have:

snip

Hope that helps.

-Dave

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Re: [asterisk-users] whisper chanspy in asterisk 1.2

Carles Pina i Estany wrote:
 Hello,

 I would like to have whisper channel spy (not private whisper) in
 Asterisk 1.2. I see here:
 http://www.the-asterisk-book.com/unstable/applikationen-chanspy.html
 That is only available for Asterisk 1.4.

 I wonder if there is any way to emulate it in Asterisk 1.2. For example,
 to join two calls: one to use a private whisper and other one to use
 a normal chanspy.

 Thank you,

   
If this were possible, I would never consider going to 1.4 even when ABE 
is available in 1.4.  Chan_mobile would be nice in 1.2 also.

We need a 1.2 spoon.  Let's start a list people!

Thanks,
Steve

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Re: [asterisk-users] Voicemail playback on iPhone

Sorry, it's clear my question was far too vague.

To clarify, is there a recipe to make * record voicemail in a format  
that allows playback on iPhone's media/music player playback for  
voicemails that are received say, in an email message.

It seems the * voicemail defaults don't work.  This link seems to  
describe codecs that do work, however I haven't seen any indications  
as to whether * voicemail can be tweaked to record in any of the  
supported formats:  http://www.kehlet.cx/

Any success out there?

On 22-Oct-07, at 7:38 PM, Ron Stephan wrote:


 Trick question I assume?

 It was mind numbingly simple on my iPhone...(though none of the  
 voice mail worked when London a few weeks ago).

 - tap voice mail -
 - tap speaker (upper right) until it turns blue (is activate)
 - tap the message you want to playback
 - use assorted  controls to delete - replay etc.


 Now...if the question is ... how do you get asterisk voice mail to  
 show up on an iPhone...I am all ears.  Groovy concept - if
 anybody has a hack - I'd love to see it.



 Elvis







 -Original Message-
 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 
 ] On Behalf Of Jason Lixfeld
 Sent: Monday, October 22, 2007 4:16 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Voicemail playback on iPhone

 Anyone managed to get this to work?  What's the recipe?

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Re: [asterisk-users] Voicemail playback on iPhone

2007-10-24 Thread Baji Panchumarti

 Jason,

 I think there is a bit of terminology confusion here,
 you can easily convert from format to another using
 sox, so if your * server is going to record and email
 you a voicemail file, it can surely  sox  the file to whatever
 format the iphone needs it in and then send the email.

 It does not appear that the iPhone is using a proprietary
 format so just try the default recording format and see
 what happens.

 -baji.

 ps : I don't have an iPhone, nor have I used * voicemail yet
caveat emptor :-)

--

 On 10/24/07, Jason Lixfeld  wrote:

 Sorry, it's clear my question was far too vague.

 To clarify, is there a recipe to make * record voicemail in a format
 that allows playback on iPhone's media/music player playback for
 voicemails that are received say, in an email message.

 It seems the * voicemail defaults don't work.  This link seems to
 describe codecs that do work, however I haven't seen any indications
 as to whether * voicemail can be tweaked to record in any of the
 supported formats:  http://www.kehlet.cx/

 Any success out there?

 On 22-Oct-07, at 7:38 PM, Ron Stephan wrote:

 
  Trick question I assume?
 
  It was mind numbingly simple on my iPhone...(though none of the
  voice mail worked when London a few weeks ago).
 
  - tap voice mail -
  - tap speaker (upper right) until it turns blue (is activate)
  - tap the message you want to playback
  - use assorted  controls to delete - replay etc.
 
 
  Now...if the question is ... how do you get asterisk voice mail to
  show up on an iPhone...I am all ears.  Groovy concept - if
  anybody has a hack - I'd love to see it.
 
 
 
  Elvis
 
 
  -Original Message-
  From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
  ] On Behalf Of Jason Lixfeld
  Sent: Monday, October 22, 2007 4:16 PM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: [asterisk-users] Voicemail playback on iPhone
 
  Anyone managed to get this to work?  What's the recipe?
 
  ___

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Re: [asterisk-users] Voicemail playback on iPhone

2007-10-24 Thread C. Savinovich


  It is doable.  The iPhone uses a subset of the Apple OS.  Sometime ago I
reviewed the file structure of the iPhone.  It is just a matter of placing
the voicemail files from * into the voicemail folder of the iPhone.
Somebody with more time than me though :)

CS


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason Lixfeld
Sent: Wednesday, October 24, 2007 7:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voicemail playback on iPhone

Sorry, it's clear my question was far too vague.

To clarify, is there a recipe to make * record voicemail in a format  
that allows playback on iPhone's media/music player playback for  
voicemails that are received say, in an email message.

It seems the * voicemail defaults don't work.  This link seems to  
describe codecs that do work, however I haven't seen any indications  
as to whether * voicemail can be tweaked to record in any of the  
supported formats:  http://www.kehlet.cx/

Any success out there?

On 22-Oct-07, at 7:38 PM, Ron Stephan wrote:


 Trick question I assume?

 It was mind numbingly simple on my iPhone...(though none of the  
 voice mail worked when London a few weeks ago).

 - tap voice mail -
 - tap speaker (upper right) until it turns blue (is activate)
 - tap the message you want to playback
 - use assorted  controls to delete - replay etc.


 Now...if the question is ... how do you get asterisk voice mail to  
 show up on an iPhone...I am all ears.  Groovy concept - if
 anybody has a hack - I'd love to see it.



 Elvis







 -Original Message-
 From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] 
 ] On Behalf Of Jason Lixfeld
 Sent: Monday, October 22, 2007 4:16 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Voicemail playback on iPhone

 Anyone managed to get this to work?  What's the recipe?

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 http://www.eset.com



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Re: [asterisk-users] Grandstream GXP-2000's and Asterisk.

2007-10-24 Thread Drew Gibson


Thomas Kenyon wrote:

Thomas Kenyon wrote:
  

I have 5 GXP 2000's with firmware 1.1.4.25 running with Asterisk 1.4.13.

Is anyone else getting the following error in the asterisk console:

[Oct 22 11:39:01] WARNING[7100]: rtp.c:1142 ast_rtp_read: RTP Read too short

every couple of seconds when a handset is in a call?

I didn't notice this happening when I was using an older GXP2000 with 
the same firmware (doesn't mean that it didn't happen).


The Call in question is using G.729.

TIA for any help with this.

I will hopefully get a bit more time to play with this today. (When I'm 
in the office in question).



Changing codec doesn't appear to matter. I gather that the cause is that 
the GXP-2000 sends empty udp packets as keep-alives. (which is all well 
and good, but even with a handful of handsets with light call volume the 
logs fill up with notices, at the moment there is only 1 call going 
through the server and this is generating 2 notices/second.


Is there any way to make asterisk ignore the empty packets from certain 
peers?


  

Hi Thomas,

I have tried to work through these (and other) issues with Grandstream 
but they seem to have a short attention span. We now buy Aastra phones.


regards,

Drew

--
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com

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Re: [asterisk-users] How to tune Asterisk AMD - Answering Machine Detection hacks

2007-10-24 Thread James FitzGibbon

On 10/24/07, Costa Dinoteli [EMAIL PROTECTED] wrote:

 Most everytime Asterisk calls  it thinks it is an Answering Machine and it
 starts playing
 the AMD message, instead of the delivering the 1st real message


Why is it thinking that it's a machine?  If you're on the console at verbose
3 or higher, you'll see what thresholds were tripped.  You can also get the
reason in the ${AMDCAUSE} variable:

[Oct 23 09:58:34] VERBOSE[25147] logger.c: -- AMD: Zap/81-1 416XXX
(null) (Fmt: 4)
[Oct 23 09:58:34] VERBOSE[25147] logger.c: -- AMD: initialSilence [2500]
greeting [1500] afterGreetingSilence [800] totalAnalysisTime [5000]
minimumWordLength [100] betweenWordsSilence [50] maximumNumberOfWords [4]
silenceThreshold [256]
[Oct 23 09:58:37] VERBOSE[25147] logger.c: -- AMD: ANSWERING MACHINE:
silenceDuration:2500 initialSilence:2500

or

[Oct 23 09:43:37] VERBOSE[24313] logger.c: -- AMD: Zap/4-1 4166XXX
(null) (Fmt: 4)
[Oct 23 09:43:37] VERBOSE[24313] logger.c: -- AMD: initialSilence [2500]
greeting [1500] afterGreetingSilence [800] totalAnalysisTime [5000]
minimumWordLength [100] betweenWordsSilence [50] maximumNumberOfWords [4]
silenceThreshold [256]
[Oct 23 09:43:38] VERBOSE[24313] logger.c: -- AMD: Word detected.
iWordsCount:1
[Oct 23 09:43:38] VERBOSE[24313] logger.c: -- AMD: Changed state to
STATE_IN_SILENCE
[Oct 23 09:43:39] VERBOSE[24313] logger.c: -- AMD: HUMAN:
silenceDuration:800 afterGreetingSilence:800

Figure out why AMD thinks it's a machine and you can change the thresholds,
either in amd.conf or in the call to AMD().

-- 
j.
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[asterisk-users] HD Voice

2007-10-24 Thread Dean Collins

Video on HD Voice. Worth a watch but nothing you wouldn't already know
about.

http://www.eweek.com/article2/0,1895,2193922,00.asp

 

My question however is this - when are ITSP's going to start offering
digital voice services with HD codecs?

It's crazy that calls to my clients via skype are better quality than
the calls via my itsp's that I pay money to are.

 

 

 

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).

 

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[asterisk-users] Remote provisioning for ATA's

2007-10-24 Thread Rizwan Hisham

Hi all,
I need a fully developed web based remote provisioning system. I cant find
anything reliable on the internet. Have already checked ataconfig.com  and
voxilla-ays.com. have tried to contact them but got no response. So if
anybody knows a good provisioning system then plz tell me about it.

-- 
Best Regards
Rizwan Hisham
Software Engineer
Axvoice Inc.
www.axvoice.com
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Re: [asterisk-users] Unusual DTMF behavior

2007-10-24 Thread John Meksavan


What is your setup, hardware wise?  

If you have the digium cards- FXO or FXS, you must make sure you tune them.  I 
had issues with DTMF's, when I went live with my Asterisk system.  Once I tune 
them, everything worked great.  

Date: Wed, 24 Oct 2007 09:05:35 -0500
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Unusual DTMF behavior

We are having an issue where DTMF is not being sent out right away and the tone 
duration is inconsistent.  For a test we send a '5', then a second later we 
send a '9', and then five seconds later we send a '5'.  If you look at the logs 
below you can see the first '5' is played right away, then the '9' comes in and 
gets queued, but it doesn't start playing until five seconds later and it plays 
for six seconds.  Then the last '5' is played.


The DTMF is coming in as only 'end' packets and we can't change that.  For this 
reason we have turned on rfc2833compensate.  Using Asterisk 1.4.11.

Any ideas?


asteriskpri04*CLI


 Protocol Discriminator: Q.931 (8)  len=5
 Call Ref: len= 2 (reference 499/0x1F3) (Terminator)
 Message type: CONNECT (7)
q931.c:3371 q931_receive: call 33267 on channel 3 enters state 10 (Active)


 Protocol Discriminator: Q.931 (8)  len=5
 Call Ref: len= 2 (reference 499/0x1F3) (Originator)
 Message type: CONNECT ACKNOWLEDGE (15)
[Oct 23 10:39:56] DEBUG[6136]: chan_zap.c:1413 zt_enable_ec: Echo cancellation 
already on

-- Zap/3-1 answered SIP/test.com-08dc1ef8

[Oct 23 10:39:58] DTMF[13914]: channel.c:2346 __ast_read: DTMF end '5' received 
on SIP/test.com-08dc1ef8, duration 0 ms
[Oct 23 10:39:58] DTMF[13914]: channel.c

:2382 __ast_read: DTMF begin emulation of '5' with duration 100 queued on 
SIP/test.com-08dc1ef8
[Oct 23 10:39:58] DEBUG[13914]: chan_zap.c:1048 zt_digit_begin: Started VLDTMF 
digit '5'

[Oct 23 10:39:59] DTMF[13914]: 
channel.c:2346 __ast_read: DTMF end '9' received on SIP/test.com-08dc1ef8, 
duration 0 ms
[Oct 23 10:39:59] DTMF[13914]: channel.c:2352 __ast_read: DTMF end '9' put into 
dtmf queue on SIP/test.com-08dc1ef8


[Oct 23 10:39:59] DTMF[13914]: channel.c:2434 __ast_read: DTMF end emulation of 
'5' queued on SIP/test.com-08dc1ef8
[Oct 23 10:39:59] DEBUG[13914]: chan_zap.c:1083 zt_digit_end: Ending VLDTMF 
digit '5'


[Oct 23 10:40:04] DTMF[13914]: channel.c:2215 __ast_read: DTMF begin emulation 
of '9' with duration 100 queued on SIP/test.com-08dc1ef8
[Oct 23 10:40:04] DEBUG[13914]: chan_zap.c:1048 zt_digit_begin: Started VLDTMF 
digit '9'

[Oct 23 10:40:04] DTMF[13914]: channel.c:2346 __ast_read: DTMF end '5' received 
on SIP/test.com-08dc1ef8, duration 0 ms
[Oct 23 10:40:04] DTMF[13914]: channel.c:2352 __ast_read: DTMF end '5' put into 
dtmf queue on SIP/test.com-08dc1ef8


[Oct 23 10:40:10] DTMF[13914]: channel.c:2465 __ast_read: DTMF end emulation of 
'9' queued on SIP/test.com-08dc1ef8
[Oct 23 10:40:10] DEBUG[13914]: chan_zap.c:1083 zt_digit_end: Ending VLDTMF 
digit '9'

[Oct 23 10:40:10] DTMF[13914]: channel.c:2215 __ast_read: DTMF begin emulation 
of '5' with duration 100 queued on SIP/test.com-08dc1ef8
[Oct 23 10:40:10] DEBUG[13914]: chan_zap.c:1048 zt_digit_begin: Started VLDTMF 
digit '5'


[Oct 23 10:40:10] DTMF[13914]: channel.c:2465 __ast_read: DTMF end emulation of 
'5' queued on SIP/test.com-08dc1ef8
[Oct 23 10:40:10] DEBUG[13914]: chan_zap.c:1083 zt_digit_end: Ending VLDTMF 
digit '5'



Thanks,
Jason



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[asterisk-users] Question about outgoing callerid

2007-10-24 Thread Jean-Yves Avenard

Hi

I have an ISDN connection with 100 DIDs assigned to it...

What I'm trying to achieve is set the proper outgoing callerID while
showing the local caller's extension in the CDR.

There is a behaviour that I just can't explain.

the callerid field in sip.conf is set as :
callerid=Jean-Yves/E 300

the callerid in iax.conf is set a:
callerid=Jean-Yves/E 300
(just the same)

Prior to making the call using the zap interface, I do:
[macro-zaptel]
;ARG1=Number to call
; set default outgoing caller ID if FROMNUMBER is empty
exten = s,1,GotoIf($[${FROMNUMBER} = ]?2:4)
exten = s,2,Set(CALLERID(number)=03)
exten = s,3,Goto(s,5)
exten = s,4,Set(CALLERID(number)=${FROMNUMBER})
exten = s,5,SetMusicOnHold(random)
exten = s,6,Dial,Zap/g1/${ARG1}

Now, after making a call using SIP, in the CDR I have:
channel = SIP/ipp...
source  = 03
clid = Jean-Yves/E 03
last data = Dial Zap/g1/0123456789

after making a call using IAX I get:
channel = IAX2/ia...
source = 300
clid = Jean-Yves/E 300
last data = Dial Zap/g1/0123456789

So my questions are:
why are the source and clid different between when a call was made
through IAX or SIP?

Ultimately, I want the clid to show up like it does for IAX:
that is:
outgoing caller ID is set to the public DID (03)
but in the CDR, I see clid = 300 (which is the local extension/account)

Is this possible?

I am using asterisk 1.2.24
Thank you
Jean-Yves

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[asterisk-users] Asterisk Shutting Down

2007-10-24 Thread Rob Schall

We've experienced the same problem twice now in the past month. The
asterisk pid stops responding. We aren't able to connect to the CLI and
all calls are dropped. The lots are pretty bare as well.

This is the message log:
Oct 24 09:12:35 WARNING[20711] channel.c: Avoided initial deadlock for
'0x8444a70', 10 retries!
Oct 24 09:12:35 WARNING[20711] channel.c: Avoided initial deadlock for
'0x84402c8', 10 retries!
Oct 24 09:12:35 WARNING[20711] channel.c: Avoided initial deadlock for
'0x83acb40', 10 retries!
Oct 24 09:12:35 WARNING[20711] channel.c: Avoided initial deadlock for
'0x8192f50', 10 retries!
Oct 24 09:12:35 WARNING[20711] channel.c: Avoided initial deadlock for
'0x8188eb0', 10 retries!
Oct 24 09:13:15 WARNING[25806] chan_sip.c: No such host: 5040
Oct 24 09:13:15 WARNING[25806] channel.c: No channel type registered for ''
Oct 24 09:13:15 NOTICE[25806] app_dial.c: Unable to create channel of
type '' (cause 66 - Channel not implemented)
Oct 24 09:18:31 WARNING[25905] chan_zap.c: getdtmf on channel 39:
Operation now in progress
Oct 24 09:19:44 WARNING[20740] chan_sip.c: Unknown SDP media type in
offer: image 5006 udptl t38
Oct 24 09:22:17 WARNING[20711] channel.c: Avoided initial deadlock for
'0x83b3788', 10 retries!
Oct 24 09:25:04 WARNING[26095] chan_sip.c: No such host: 5040
Oct 24 09:25:04 WARNING[26095] channel.c: No channel type registered for ''
Oct 24 09:25:04 NOTICE[26095] app_dial.c: Unable to create channel of
type '' (cause 66 - Channel not implemented)
Oct 24 09:27:23 WARNING[20711] channel.c: Avoided initial deadlock for
'0x83b45f0', 10 retries!
Oct 24 09:27:23 WARNING[20711] channel.c: Avoided initial deadlock for
'0x83a3f00', 10 retries!
Oct 24 09:27:23 WARNING[20711] channel.c: Avoided initial deadlock for
'0x81a16a8', 10 retries!
Oct 24 09:27:23 WARNING[20711] channel.c: Avoided initial deadlock for
'0x83a2540', 10 retries!
Oct 24 09:27:23 WARNING[20711] channel.c: Avoided initial deadlock for
'0x83a3560', 10 retries!
Oct 24 09:27:23 WARNING[20711] channel.c: Avoided initial deadlock for
'0x818a0d0', 10 retries!
Oct 24 09:36:36 WARNING[26383] app_voicemail.c: Unable to read password
Oct 24 09:37:57 WARNING[26449] chan_iax2.c: No such host: 6677
Oct 24 09:37:59 WARNING[20711] channel.c: Avoided initial deadlock for
'0x83a11d8', 10 retries!
Oct 24 09:37:59 WARNING[20711] channel.c: Avoided initial deadlock for
'0x81a16a8', 10 retries!
Oct 24 09:37:59 WARNING[20711] channel.c: Avoided initial deadlock for
'0x83acb40', 10 retries!
Oct 24 09:38:00 WARNING[20711] channel.c: Avoided initial deadlock for
'0x820c6f0', 10 retries!
Oct 24 09:38:29 WARNING[26491] app_voicemail.c: Unable to read password
Oct 24 09:39:21 WARNING[26497] file.c: Failed to write frame
Oct 24 09:43:53 WARNING[20711] channel.c: Avoided initial deadlock for
'0x83a2c48', 10 retries!
Oct 24 09:43:53 WARNING[20711] channel.c: Avoided initial deadlock for
'0x83e7f30', 10 retries!
Oct 24 09:43:53 WARNING[20711] channel.c: Avoided initial deadlock for
'0x83e8f50', 10 retries!
Oct 24 09:43:53 WARNING[20711] channel.c: Avoided initial deadlock for
'0x83ace58', 10 retries!
Oct 24 10:08:00 WARNING[27070] app_voicemail.c: Couldn't read username
Oct 24 10:10:11 WARNING[20711] channel.c: Avoided initial deadlock for
'0x843f4a0', 10 retries!


Our setup is as follows:
Dell Dimension 3000
Sangoma A101  Sangoma A102  Digium Analog FXO/FXO card
Asterisk 1.2.13-r1
Realtime for extensions/voicemail/sipiax buddies

Any insight would be much appreciated as asterisk is our current lifeline.

Thanks,
Rob

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[asterisk-users] Two DTMF tones on keypress with Handsfree cell

2007-10-24 Thread Andrew J. Barr

Hello, I am using Asterisk SVN, a cellular phone, and chan_mobile to
run a small home PBX with two analog telephones connected to a Linksys
ATA using SIP. It works great (except for some Bluetooth adapter bugs
that I am still trying to beat...seems the misaligned audio detection
still needs work), but I have encountered an interesting issue.

If I am using an automated system that accepts input using the
telephone keypad DTMF tones, pressing those keys on my analog
telephone emits the tone once, and then about a half-second to a
second later the cell phone does the same thing, because the ATA
adapter transmitted the DTMF press digitally as well. The only system
I've tried to interact with so far also accepted voice input, so I
didn't have a chance to test further and see if it was my fault
(pressing a wrong key) or the dual-tone problem that confused the
system on the other end. Is there some way to suppress DTMF tones at
some point in the call routing? Or disable the phone sending DTMF
tones during the actual call?

Thanks in advance for any ideas or input.

-- 
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[asterisk-users] Backport Func_ODBC question

2007-10-24 Thread JR Richardson

Hi All,

Ingnorant question, how do you apply the backport func_odbc to 1.2 branch?

Thanks.

JR
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Re: [asterisk-users] Voicemail playback on iPhone

I guess what I'm asking is if there is a recipe anyone has used to  
allow a voicemail message (once recorded by asterisk) to playback on  
iPhone when said recorded voicemail is received as an email  
attachment.  I understand you can convert using sox, so I guess that's  
the ingredient and some sort of * configs would be the glue - I  
suppose it's the glue I can't figure out.  I'm not trying to figure  
out how to get voicemails to show up in iPhone VVM or anything like  
that.

If the voicemail configs can't be tweaked enough to record in a format  
iPhone can play, how can I get something like sox convert the message  
to another format before * emails the voicemail off to the callee?  If  
I understand correctly, the voicemail app takes care of the entire  
process from the time voicemail is recorded from the caller to the  
time it is sent to the callee (ie: email).  If that's true, then I  
guess I need to understand how to tell asterisk to fork from voicemail  
to some script to convert the recording to something iPhone friendly  
before we fork back to voicemail where we left off and actually email  
the message to the callee.

Am I making any sense?

On 24-Oct-07, at 11:12 AM, Baji Panchumarti wrote:

 Jason,

 I think there is a bit of terminology confusion here,
 you can easily convert from format to another using
 sox, so if your * server is going to record and email
 you a voicemail file, it can surely  sox  the file to whatever
 format the iphone needs it in and then send the email.

 It does not appear that the iPhone is using a proprietary
 format so just try the default recording format and see
 what happens.

 -baji.

 ps : I don't have an iPhone, nor have I used * voicemail yet
caveat emptor :-)

 --

 On 10/24/07, Jason Lixfeld  wrote:

 Sorry, it's clear my question was far too vague.

 To clarify, is there a recipe to make * record voicemail in a format
 that allows playback on iPhone's media/music player playback for
 voicemails that are received say, in an email message.

 It seems the * voicemail defaults don't work.  This link seems to
 describe codecs that do work, however I haven't seen any indications
 as to whether * voicemail can be tweaked to record in any of the
 supported formats:  http://www.kehlet.cx/

 Any success out there?

 On 22-Oct-07, at 7:38 PM, Ron Stephan wrote:


 Trick question I assume?

 It was mind numbingly simple on my iPhone...(though none of the
 voice mail worked when London a few weeks ago).

 - tap voice mail -
 - tap speaker (upper right) until it turns blue (is activate)
 - tap the message you want to playback
 - use assorted  controls to delete - replay etc.


 Now...if the question is ... how do you get asterisk voice mail to
 show up on an iPhone...I am all ears.  Groovy concept - if
 anybody has a hack - I'd love to see it.



 Elvis


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
 ] On Behalf Of Jason Lixfeld
 Sent: Monday, October 22, 2007 4:16 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Voicemail playback on iPhone

 Anyone managed to get this to work?  What's the recipe?

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Re: [asterisk-users] OSLEC and zaptel-1.4.5.1

Dave Fullerton wrote:
Replies/Comments inline...

Ditto :-)

Alan Lord wrote:
Hi all,

After reading great things about the OSLEC Echo Canceller
(http://www.rowetel.com/ucasterisk/oslec) and seeing the reactions of
people who have tried it on a recent Trixbox thread
(http://www.trixbox.org/forums/trixbox-forums/open-discussion/need-people-echo-problems),

it sounds like it is the bees knees for sorting out echo problems with
cards like the x100p.

I am using OSLEC on my home pbx. I used to have echo on some calls prior
to OSLEC but have been echo free since I installed it.

This seems to be most peoples experience with it.

Has anyone managed to get oslec to work with recent zaptel and kernel
(I'm running 2.6.23)?

I'm only using 2.6.17 and zaptel-1.4.4 at the moment. But if the patches
apply it should work.

I might try and downgrade to zaptel-1.4.4 and see if that helps.

Having followed the instructions on the oslec site, and ensuring the
patch for zaptel takes O.K (I manually installed the patch into the
zaptel source tree just to make sure). I can build the oslec module, and
build a patched zaptel-1.4.5.1-oslec without any compilation issues.

However when I reload the system during boot-up dmesg tells me:

Zapata Telephony Interface Registered on major 196
Zaptel Version: 1.4.5.1
Zaptel Echo Canceller: MG2
Zaptap registered 'sample' char driver on major 33 (This means the patch
went in O.K.)
ACPI: PCI Interrupt :00:08.0[A] - GSI 17 (level, low) - IRQ 22
wcfxo: DAA mode is 'FCC'
Found a Wildcard FXO: Wildcard X100P

Notice the choice of echo canceller

Check the zconfig.h file in the zaptel source and make sure that the line:
#define ECHO_CAN_OSLEC
is not commented out but all the lines for the other echo cancelers are.

Yep - I manually went through the 1.4.4 patch and ensured it was applied
cleanly to the zaptel source tree. Like this:

.../* #define ECHO_CAN_MARK2 */
/* #define ECHO_CAN_MARK3 */
/* #define ECHO_CAN_KB1 */
/* This is the new latest and greatest */
/* #define ECHO_CAN_MG2 */
#define ECHO_CAN_OSLEC

Did you start with a clean source (or at least did a make clean) before
you compiled? Are you using the zaptel-1.4.4.patch from the oslec SVN or
some other patch?

Yes, clean source. I used the 1.4.4. and it applied cleanly (just with a
bit of fuzz). I went through and did it manually too - just to make sure.

If I look at what modules are installed:

# lsmod
Module Size Used by
zttranscode 6280 0
ztdummy 3432 0
wcfxo 9760 0
zaptel200120 7 zttranscode,ztdummy,wcfxo
crc_ccitt 1792 1 zaptel

Just for kicks, try inserting the oslec module by hand (insmod oslec)
and see if that makes a difference.

Tried that too!

In my kernel modules/misc directory I have:

snip

Hope that helps.

-Dave

Thanks for the comments. It's good to know it does work but perhaps
there is something in the 1.4.5.1 sources... Think I'll do a quick grep
for #define ECHO_CAN_MG2 and see if it being set elsewhere.

Alan

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Re: [asterisk-users] Asterisk integration with IBM Sametime

2007-10-24 Thread ram

On 10/24/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:

  Hi,

 I am trying to setup a conference between Sametime users using
 conferencing infrastructure of asterisk.

 Sametime server has a component called TCSPI, which we can implement to
 talk to any PBX, including asterisk (as per documentation). I was trying to
 implement the TCSPI for Asterisk.



Hi

you can  configure asterisk to trust any call from Samtime Server

and you can configure conference bridge in Asterisk

I never tried , but its possible.

since iam using 3rd party SIP server, and iam using Asterisk as bridge

ram
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Re: [asterisk-users] OSLEC and zaptel-1.4.5.1

2007-10-24 Thread marcotasto

Hi Alan.
I've installed OSLEC with zaptel-1.4.5.1 applying the patches made for the 1.4 
version and I have had the same problem.
Looking at the compiler options I've found that the symbol ECHO_CAN_FROMENV is 
defined by default and this prevents the echo selection from zconfig.h.
I've solved changing the first part of Makefile.kernel26 (in the zaptel 
directory) this way:

ifndef ECHO_CAN_NAME
  ECHO_CAN_NAME := OSLEC
endif

This forces the compiler to include OSLEC as echo cancellation engine (probably 
there is a better way but I don't know it).

I've then rebuilt zaptel and installed through normal make procedures.

To be able to modprobe it I've then copied the oslec.ko file build by the OSLEC 
distribution in the kernel driver directory (my own is 
/lib/modules/2.6.18.8-0.5-default/misc and it's where zaptel drivers are 
installed). I've then run the depmod command to regenerate the modules 
dependencies.

I'm now able to modprobe zaptel and to have oslec automatically installed as 
you can see below:

 lsmod | grep zaptel
zaptel12  6 zttranscode,wctdm
oslec  23332  1 zaptel
crc_ccitt   6272  1 zaptel

I hope this could help you.

Best regards,
Marco Signorini.


 
 Hi all,
 
 After reading great things about the OSLEC Echo Canceller
 (http://www.rowetel.com/ucasterisk/oslec) and seeing the reactions of
 people who have tried it on a recent Trixbox thread
 (http://www.trixbox.org/forums/trixbox-forums/open-discussion/need-people-echo-problems),
 it sounds like it is the bees knees for sorting out echo problems with
 cards like the x100p.
 
 Has anyone managed to get oslec to work with recent zaptel and kernel
 (I'm running 2.6.23)?
 
 Lots of information below. Comments/suggestions welcome.
 
 Having followed the instructions on the oslec site, and ensuring the
 patch for zaptel takes O.K (I manually installed the patch into the
 zaptel source tree just to make sure). I can build the oslec module, and
 build a patched zaptel-1.4.5.1-oslec without any compilation issues.
 
 However when I reload the system during boot-up dmesg tells me:
 
 Zapata Telephony Interface Registered on major 196
 Zaptel Version: 1.4.5.1
 Zaptel Echo Canceller: MG2
 Zaptap registered 'sample' char driver on major 33 (This means the patch
 went in O.K.)
 ACPI: PCI Interrupt :00:08.0[A] - GSI 17 (level, low) - IRQ 22
 wcfxo: DAA mode is 'FCC'
 Found a Wildcard FXO: Wildcard X100P
 
 Notice the choice of echo canceller
 
 If I look at what modules are installed:
 
 # lsmod
 Module  Size  Used by
 zttranscode 6280  0
 ztdummy 3432  0
 wcfxo   9760  0
 zaptel200120  7 zttranscode,ztdummy,wcfxo
 crc_ccitt   1792  1 zaptel
 
 No oslec :-(
 
 In my kernel modules/misc directory I have:
 -rw-r--r-- 1 root root 10727 2007-10-24 14:44 oslec.ko
 -rw-r--r-- 1 root root 65372 2007-10-24 14:41 pciradio.ko
 -rw-r--r-- 1 root root 91321 2007-10-24 14:41 tor2.ko
 -rw-r--r-- 1 root root 18901 2007-10-24 14:41 torisa.ko
 -rw-r--r-- 1 root root 12605 2007-10-24 14:41 wcfxo.ko
 -rw-r--r-- 1 root root 15989 2007-10-24 14:41 wct1xxp.ko
 drwxr-xr-x 2 root root  4096 2007-10-24 14:41 wct4xxp
 drwxr-xr-x 2 root root  4096 2007-10-24 14:41 wctc4xxp
 drwxr-xr-x 2 root root  4096 2007-10-24 14:41 wctdm24xxp
 -rw-r--r-- 1 root root 41046 2007-10-24 14:41 wctdm.ko
 -rw-r--r-- 1 root root 32882 2007-10-24 14:41 wcte11xp.ko
 -rw-r--r-- 1 root root 45804 2007-10-24 14:41 wcte12xp.ko
 -rw-r--r-- 1 root root 16527 2007-10-24 14:41 wcusb.ko
 drwxr-xr-x 2 root root  4096 2007-10-24 14:41 xpp
 -rw-r--r-- 1 root root 81616 2007-10-24 14:41 zaptel.ko
 -rw-r--r-- 1 root root  8270 2007-10-24 14:41 ztd-eth.ko
 -rw-r--r-- 1 root root  5530 2007-10-24 14:41 ztd-loc.ko
 -rw-r--r-- 1 root root  5297 2007-10-24 14:41 ztdummy.ko
 -rw-r--r-- 1 root root 11687 2007-10-24 14:41 ztdynamic.ko
 -rw-r--r-- 1 root root  8639 2007-10-24 14:41 zttranscode.ko
 
 My /etc/zaptel.conf is:
 loadzone=uk
 defaultzone=uk
 
 fxsks=1
 
 My /etc/asterisk/zapata.conf is
 
 ; Zapata telephony interface ;
 ; Configuration file
 [channels]
 ;Hardware defaults for the x100p card
 ;usecallerid=yes
 ;hidecallerid=no
 ;callwaiting=no
 ;threewaycalling=yes
 ;usedistinctiveringdetection=yes
 ;transfer=yes
 ;usecallingpres=yes
 ;callwaitingcallerid=yes
 ;cancallforward=yes
 ;callreturn=yes
 echocancel=yes
 echotrainingwhenbridged=no
 ;echotraining=400
 rxwink=300 ; Atlas seems to use long (250ms) winks
 
 ;cidsignalling=v23 ; Added for UK CLI detection
 ;cidstart=usehist ; After patching the driver from here :
 ; http://www.lusyn.com/resources/asterisk/usehist.htm
 ;callerid=asreceived ; propagate the CID received from BT
 ;rxgain=1.0
 ;txgain=1.0
 
 ;define channel
 context=main_menu
 language=en
 signalling=fxs_ks
 channel = 1 ;Our x100p
 
 --
 
 Alan
 
 -- 
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Re: [asterisk-users] Voicemail playback on iPhone

2007-10-24 Thread Peder @ NetworkOblivion

This is semi-related, but I have a Tmobile MDA and I couldn't play the 
files either.  The issue was not a codec issue, it was an email encoding 
issue.  If I sent the message to an email account and it was then 
downloaded to my desktop via outlook and then forwarded on to my phone, 
I can listen to them.  If I just send it direct to the phone, I see the 
attachment and it opens in media player, but it won't play.  I don't 
know if you are having codec issues or email encoding issues, but it is 
a place to look.

Incidentally, if someone knows how to get around the download email and 
then forward issue that I am having, I would like to hear it.

Peder


Jason Lixfeld wrote:
 I guess what I'm asking is if there is a recipe anyone has used to  
 allow a voicemail message (once recorded by asterisk) to playback on  
 iPhone when said recorded voicemail is received as an email  
 attachment.  I understand you can convert using sox, so I guess that's  
 the ingredient and some sort of * configs would be the glue - I  
 suppose it's the glue I can't figure out.  I'm not trying to figure  
 out how to get voicemails to show up in iPhone VVM or anything like  
 that.
 
 If the voicemail configs can't be tweaked enough to record in a format  
 iPhone can play, how can I get something like sox convert the message  
 to another format before * emails the voicemail off to the callee?  If  
 I understand correctly, the voicemail app takes care of the entire  
 process from the time voicemail is recorded from the caller to the  
 time it is sent to the callee (ie: email).  If that's true, then I  
 guess I need to understand how to tell asterisk to fork from voicemail  
 to some script to convert the recording to something iPhone friendly  
 before we fork back to voicemail where we left off and actually email  
 the message to the callee.
 
 Am I making any sense?
 
 On 24-Oct-07, at 11:12 AM, Baji Panchumarti wrote:
 
 Jason,

 I think there is a bit of terminology confusion here,
 you can easily convert from format to another using
 sox, so if your * server is going to record and email
 you a voicemail file, it can surely  sox  the file to whatever
 format the iphone needs it in and then send the email.

 It does not appear that the iPhone is using a proprietary
 format so just try the default recording format and see
 what happens.

 -baji.

 ps : I don't have an iPhone, nor have I used * voicemail yet
caveat emptor :-)

 --

 On 10/24/07, Jason Lixfeld  wrote:

 Sorry, it's clear my question was far too vague.

 To clarify, is there a recipe to make * record voicemail in a format
 that allows playback on iPhone's media/music player playback for
 voicemails that are received say, in an email message.

 It seems the * voicemail defaults don't work.  This link seems to
 describe codecs that do work, however I haven't seen any indications
 as to whether * voicemail can be tweaked to record in any of the
 supported formats:  http://www.kehlet.cx/

 Any success out there?

 On 22-Oct-07, at 7:38 PM, Ron Stephan wrote:

 Trick question I assume?

 It was mind numbingly simple on my iPhone...(though none of the
 voice mail worked when London a few weeks ago).

 - tap voice mail -
 - tap speaker (upper right) until it turns blue (is activate)
 - tap the message you want to playback
 - use assorted  controls to delete - replay etc.


 Now...if the question is ... how do you get asterisk voice mail to
 show up on an iPhone...I am all ears.  Groovy concept - if
 anybody has a hack - I'd love to see it.



 Elvis


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
 ] On Behalf Of Jason Lixfeld
 Sent: Monday, October 22, 2007 4:16 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Voicemail playback on iPhone

 Anyone managed to get this to work?  What's the recipe?

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Re: [asterisk-users] Voicemail playback on iPhone

2007-10-24 Thread Darrick Hartman (lists)

Jason Lixfeld wrote:
 I guess what I'm asking is if there is a recipe anyone has used to  
 allow a voicemail message (once recorded by asterisk) to playback on  
 iPhone when said recorded voicemail is received as an email  
 attachment.  I understand you can convert using sox, so I guess that's  
 the ingredient and some sort of * configs would be the glue - I  
 suppose it's the glue I can't figure out.  I'm not trying to figure  
 out how to get voicemails to show up in iPhone VVM or anything like  
 that.
 

The iPhone can't play back wav or wav49 files?  Check your 
voicemail.conf file.  What format are you currently using?

Darrick
-- 
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com

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[asterisk-users] AstManProxy Host Prefix?

2007-10-24 Thread Douglas Garstang

Can the Asterisk Manager Proxy, AstManProxy, prefix the host name that output 
applies to, to the start of each line? If you are proxying multiple systems, 
how can it uniquely identify the output from each system?

Thanks,
Doug.




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Re: [asterisk-users] Voicemail playback on iPhone

It plays wav, but as far as I understand, * encodes the wav using  
something like ulaw which iPhone doesn't support.  If I can switch the  
codec to pcm, that may work - is that possible?

On 24-Oct-07, at 12:25 PM, Darrick Hartman (lists) wrote:

 Jason Lixfeld wrote:
 I guess what I'm asking is if there is a recipe anyone has used to
 allow a voicemail message (once recorded by asterisk) to playback on
 iPhone when said recorded voicemail is received as an email
 attachment.  I understand you can convert using sox, so I guess  
 that's
 the ingredient and some sort of * configs would be the glue - I
 suppose it's the glue I can't figure out.  I'm not trying to figure
 out how to get voicemails to show up in iPhone VVM or anything like
 that.


 The iPhone can't play back wav or wav49 files?  Check your
 voicemail.conf file.  What format are you currently using?

 Darrick
 -- 
 Darrick Hartman
 DJH Solutions, LLC
 http://www.djhsolutions.com

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Re: [asterisk-users] Backport Func_ODBC question

2007-10-24 Thread Tilghman Lesher

On Wednesday 24 October 2007 10:50:47 JR Richardson wrote:
 Ingnorant question, how do you apply the backport func_odbc to 1.2 branch?

ASTSRC=/path/to/downloaded/asterisk/source make install

-- 
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[asterisk-users] SOLVED Re: Voicemail playback on iPhone

Seems the answer was simple enough - set format=wav and it works  
fine.  Mine was set at wav49.

On 24-Oct-07, at 1:02 PM, Jason Lixfeld wrote:

 It plays wav, but as far as I understand, * encodes the wav using
 something like ulaw which iPhone doesn't support.  If I can switch the
 codec to pcm, that may work - is that possible?

 On 24-Oct-07, at 12:25 PM, Darrick Hartman (lists) wrote:

 Jason Lixfeld wrote:
 I guess what I'm asking is if there is a recipe anyone has used to
 allow a voicemail message (once recorded by asterisk) to playback on
 iPhone when said recorded voicemail is received as an email
 attachment.  I understand you can convert using sox, so I guess
 that's
 the ingredient and some sort of * configs would be the glue - I
 suppose it's the glue I can't figure out.  I'm not trying to figure
 out how to get voicemails to show up in iPhone VVM or anything like
 that.


 The iPhone can't play back wav or wav49 files?  Check your
 voicemail.conf file.  What format are you currently using?

 Darrick
 -- 
 Darrick Hartman
 DJH Solutions, LLC
 http://www.djhsolutions.com

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Re: [asterisk-users] OSLEC and zaptel-1.4.5.1

2007-10-24 Thread Dave Fullerton

marcotasto wrote:
 Hi Alan.
 I've installed OSLEC with zaptel-1.4.5.1 applying the patches made for the 
 1.4 version and I have had the same problem.
 Looking at the compiler options I've found that the symbol ECHO_CAN_FROMENV 
 is defined by default and this prevents the echo selection from zconfig.h.
 I've solved changing the first part of Makefile.kernel26 (in the zaptel 
 directory) this way:
 
 ifndef ECHO_CAN_NAME
   ECHO_CAN_NAME := OSLEC
 endif
 
 This forces the compiler to include OSLEC as echo cancellation engine 
 (probably there is a better way but I don't know it).
 
 I've then rebuilt zaptel and installed through normal make procedures.
 
 To be able to modprobe it I've then copied the oslec.ko file build by the 
 OSLEC distribution in the kernel driver directory (my own is 
 /lib/modules/2.6.18.8-0.5-default/misc and it's where zaptel drivers are 
 installed). I've then run the depmod command to regenerate the modules 
 dependencies.
 
 I'm now able to modprobe zaptel and to have oslec automatically installed as 
 you can see below:
 

snip

This looks like it is isolated to 1.4.5.x. It looks like digium added a 
method of selecting the echo canceler by using environment variables but 
didn't get it quite right. It appears to be fixed in 1.4.6. In 1.4.6 it 
will use the selection made in zconfig.h UNLESS you set ECHO_CAN_NAME to 
your preferred echo canceler (OSLEC,MG2,etc) to override it.

-Dave

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Re: [asterisk-users] OSLEC and zaptel-1.4.5.1

marcotasto wrote:
 Hi Alan.
 I've installed OSLEC with zaptel-1.4.5.1 applying the patches made for the 
 1.4 version and I have had the same problem.
 Looking at the compiler options I've found that the symbol ECHO_CAN_FROMENV 
 is defined by default and this prevents the echo selection from zconfig.h.
 I've solved changing the first part of Makefile.kernel26 (in the zaptel 
 directory) this way:
 
 ifndef ECHO_CAN_NAME
   ECHO_CAN_NAME := OSLEC
 endif
 
 This forces the compiler to include OSLEC as echo cancellation engine 
 (probably there is a better way but I don't know it).
 
 I've then rebuilt zaptel and installed through normal make procedures.
 
 To be able to modprobe it I've then copied the oslec.ko file build by the 
 OSLEC distribution in the kernel driver directory (my own is 
 /lib/modules/2.6.18.8-0.5-default/misc and it's where zaptel drivers are 
 installed). I've then run the depmod command to regenerate the modules 
 dependencies.
 
 I'm now able to modprobe zaptel and to have oslec automatically installed as 
 you can see below:
 

Many thanks for the information. That sounds like it should do the trick!

I will try later on and report back if I have success.

Grazie Mille


Alan

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Re: [asterisk-users] Voicemail playback on iPhone

2007-10-24 Thread Norman Franke

On Oct 24, 2007, at 12:25 PM, [EMAIL PROTECTED]  
wrote:



This is semi-related, but I have a Tmobile MDA and I couldn't play the
files either.  The issue was not a codec issue, it was an email  
encoding

issue.  If I sent the message to an email account and it was then
downloaded to my desktop via outlook and then forwarded on to my  
phone,
I can listen to them.  If I just send it direct to the phone, I see  
the

attachment and it opens in media player, but it won't play.  I don't
know if you are having codec issues or email encoding issues, but  
it is

a place to look.


I have an iPhone and tried several things to get a message to play in  
an email and I gave up. I ended up mailing a link that then runs the  
file through a conversion CGI-like deal. Unfortunately, the iPhone  
also doesn't support many low bandwidth codecs. It does support AMR,  
but that's about it.


I eventually got this working, but not with Asterisk. It's for our  
legacy voice mail system.


-Norman Franke
ASD, Inc.

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Re: [asterisk-users] AstManProxy Host Prefix?

2007-10-24 Thread Richard Lyman

Douglas Garstang wrote:
 Can the Asterisk Manager Proxy, AstManProxy, prefix the host name that output 
 applies to, to the start of each line? If you are proxying multiple systems, 
 how can it uniquely identify the output from each system?

 Thanks,
 Doug.

   
each Event block should have a

Server: .

appended to it.


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Re: [asterisk-users] Remote provisioning for ATA's

2007-10-24 Thread Matt

Your best bet may be to write your own.  That's what we ended up doing and
it isn't that hard.

On 10/24/07, Rizwan Hisham [EMAIL PROTECTED] wrote:

 Hi all,
 I need a fully developed web based remote provisioning system. I cant find
 anything reliable on the internet. Have already checked ataconfig.com  and
 voxilla-ays.com. have tried to contact them but got no response. So if
 anybody knows a good provisioning system then plz tell me about it.

 --
 Best Regards
 Rizwan Hisham
 Software Engineer
 Axvoice Inc.
 www.axvoice.com
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Re: [asterisk-users] Unusual DTMF behavior

2007-10-24 Thread Jason Landrey

We have Digium PRI cards, TE110 and TE420 (with hardware echo cancellation).


On 10/24/07, John Meksavan [EMAIL PROTECTED] wrote:

  What is your setup, hardware wise?

 If you have the digium cards- FXO or FXS, you must make sure you tune
 them.  I had issues with DTMF's, when I went live with my Asterisk system.
 Once I tune them, everything worked great.

 --
 Date: Wed, 24 Oct 2007 09:05:35 -0500
 From: [EMAIL PROTECTED]
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Unusual DTMF behavior

 We are having an issue where DTMF is not being sent out right away and the
 tone duration is inconsistent.  For a test we send a '5', then a second
 later we send a '9', and then five seconds later we send a '5'.  If you look
 at the logs below you can see the first '5' is played right away, then the
 '9' comes in and gets queued, but it doesn't start playing until five
 seconds later and it plays for six seconds.  Then the last '5' is played.

 The DTMF is coming in as only 'end' packets and we can't change that.  For
 this reason we have turned on rfc2833compensate.  Using Asterisk 1.4.11.

 Any ideas?


 asteriskpri04*CLI
  Protocol Discriminator: Q.931 (8)  len=5
  Call Ref: len= 2 (reference 499/0x1F3) (Terminator)
  Message type: CONNECT (7)
 q931.c:3371 q931_receive: call 33267 on channel 3 enters state 10 (Active)
  Protocol Discriminator: Q.931 (8)  len=5
  Call Ref: len= 2 (reference 499/0x1F3) (Originator)
  Message type: CONNECT ACKNOWLEDGE (15)
 [Oct 23 10:39:56] DEBUG[6136]: chan_zap.c:1413 zt_enable_ec: Echo
 cancellation already on
 -- Zap/3-1 answered SIP/test.com-08dc1ef8

 [Oct 23 10:39:58] DTMF[13914]: channel.c:2346 __ast_read: DTMF end '5'
 received on SIP/test.com-08dc1ef8, duration 0 ms
 [Oct 23 10:39:58] DTMF[13914]: channel.c :2382 __ast_read: DTMF begin
 emulation of '5' with duration 100 queued on SIP/test.com-08dc1ef8
 [Oct 23 10:39:58] DEBUG[13914]: chan_zap.c:1048 zt_digit_begin: Started
 VLDTMF digit '5'

 [Oct 23 10:39:59] DTMF[13914]: channel.c:2346 __ast_read: DTMF end '9'
 received on SIP/test.com-08dc1ef8, duration 0 ms
 [Oct 23 10:39:59] DTMF[13914]: channel.c:2352 __ast_read: DTMF end '9' put
 into dtmf queue on SIP/test.com-08dc1ef8

 [Oct 23 10:39:59] DTMF[13914]: channel.c:2434 __ast_read: DTMF end
 emulation of '5' queued on SIP/test.com-08dc1ef8
 [Oct 23 10:39:59] DEBUG[13914]: chan_zap.c:1083 zt_digit_end: Ending
 VLDTMF digit '5'

 [Oct 23 10:40:04] DTMF[13914]: channel.c:2215 __ast_read: DTMF begin
 emulation of '9' with duration 100 queued on SIP/test.com-08dc1ef8
 [Oct 23 10:40:04] DEBUG[13914]: chan_zap.c:1048 zt_digit_begin: Started
 VLDTMF digit '9'
 [Oct 23 10:40:04] DTMF[13914]: channel.c:2346 __ast_read: DTMF end '5'
 received on SIP/test.com-08dc1ef8, duration 0 ms
 [Oct 23 10:40:04] DTMF[13914]: channel.c:2352 __ast_read: DTMF end '5' put
 into dtmf queue on SIP/test.com-08dc1ef8

 [Oct 23 10:40:10] DTMF[13914]: channel.c:2465 __ast_read: DTMF end
 emulation of '9' queued on SIP/test.com-08dc1ef8
 [Oct 23 10:40:10] DEBUG[13914]: chan_zap.c:1083 zt_digit_end: Ending
 VLDTMF digit '9'
 [Oct 23 10:40:10] DTMF[13914]: channel.c:2215 __ast_read: DTMF begin
 emulation of '5' with duration 100 queued on SIP/test.com-08dc1ef8
 [Oct 23 10:40:10] DEBUG[13914]: chan_zap.c:1048 zt_digit_begin: Started
 VLDTMF digit '5'

 [Oct 23 10:40:10] DTMF[13914]: channel.c:2465 __ast_read: DTMF end
 emulation of '5' queued on SIP/test.com-08dc1ef8
 [Oct 23 10:40:10] DEBUG[13914]: chan_zap.c:1083 zt_digit_end: Ending
 VLDTMF digit '5'


 Thanks,
 Jason


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Re: [asterisk-users] AstManProxy Host Prefix?

2007-10-24 Thread Douglas Garstang

Thanks, just realised that...

- Original Message 
From: Richard Lyman [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Wednesday, October 24, 2007 10:45:25 AM
Subject: Re: [asterisk-users] AstManProxy Host Prefix?


Douglas Garstang wrote:
 Can the Asterisk Manager Proxy, AstManProxy, prefix the host name
 that output applies to, to the start of each line? If you are proxying
 multiple systems, how can it uniquely identify the output from each
 system?

 Thanks,
 Doug.

   
each Event block should have a

Server: .

appended to it.


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[asterisk-users] AMI ActionID.... Doesn't work

2007-10-24 Thread Douglas Garstang

Is it well known that setting the ActionID when connecting to AMI has 
absolutely no effect?
Is this fixed in Asterisk 1.4?

If you add an ActionID to your Originate command for example, it looks like the 
only events that come back with an ActionID associated are the initial 
response, OriginateSuccess and OriginateFailure. That's it. No other events 
have an ActionID associated. This pretty much makes the AMI useless. 

What about all the other events? Newcallerid, Newstate, Link, Unlink and REALLY 
importantly the CDR events.

Really... someone please tell me it's fixed in 1.4?

Thanks,
Doug.



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[asterisk-users] [Fwd: Internal LAN echo problem]

2007-10-24 Thread Jonn Taylor

Any ideas ?

Jonn

 Original Message 
Subject:[asterisk-users] Internal LAN echo problem
Date:   Wed, 24 Oct 2007 08:34:32 -0500
From:   Jonn R Taylor [EMAIL PROTECTED]
Reply-To:   Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Hi all,

I have an internal echo problem on my LAN only. I replaced the LAN 
switch with a new linksys 2024 with QOS and seemed to help but not fix 
the problem. Any ideas? Here in my setup - Dell PE6400 Dual 700, 
Asterisk 1.2.24/FreePBX, 2-NIC cards, one with a public ip and one with 
an internal ip, one PAP2, one SPA3102 and 2 BT101. I know that bt's are 
cheap that are known for echo problem in the handset. I have one remote 
user that never has a problem. I have a remote test server at home 
connect via IAX with no problems, also a PAP2 with no problem. External 
faxing from the rest of the world via our voip provider is working 
great. One strange thing that I noticed is that we can not fax to our 
iaxmodem, ATA --- iaxmodem, but works perfect ATA --- rx_fax. Not sure 
why either.

Jonn

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Re: [asterisk-users] [Fwd: Internal LAN echo problem]

2007-10-24 Thread Eric ManxPower Wieling

Any echo you hear on pure IP calls is caused by the endpoint phone.  You 
cannot do ANYTHING about it on Asterisk.


Jonn Taylor wrote:
 Any ideas ?
 
 Jonn
 
  Original Message 
 Subject:  [asterisk-users] Internal LAN echo problem
 Date: Wed, 24 Oct 2007 08:34:32 -0500
 From: Jonn R Taylor [EMAIL PROTECTED]
 Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 To:   Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 
 
 
 Hi all,
 
 I have an internal echo problem on my LAN only. I replaced the LAN 
 switch with a new linksys 2024 with QOS and seemed to help but not fix 
 the problem. Any ideas? Here in my setup - Dell PE6400 Dual 700, 
 Asterisk 1.2.24/FreePBX, 2-NIC cards, one with a public ip and one with 
 an internal ip, one PAP2, one SPA3102 and 2 BT101. I know that bt's are 
 cheap that are known for echo problem in the handset. I have one remote 
 user that never has a problem. I have a remote test server at home 
 connect via IAX with no problems, also a PAP2 with no problem. External 
 faxing from the rest of the world via our voip provider is working 
 great. One strange thing that I noticed is that we can not fax to our 
 iaxmodem, ATA --- iaxmodem, but works perfect ATA --- rx_fax. Not sure 
 why either.

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Re: [asterisk-users] libdundi?

2007-10-24 Thread Michael Collins

 I would have thought an LGPL version wouldn't be out of the question.
 

I hope not!  LGPL is perfect for library-ish FOSS.  Releasing libraries
under standard GPL, while making Richard Stallman's heart go
pitter-patter, limits what they can do since they can only go into other
GPL projects.  

The LGPL is a great license that balances software freedom/protection
with the flexibility to be used in all sorts of software projects,
including (gasp!) commercial and (double gasp!) proprietary ones.

A libdundi that could be included in other OSS telephony projects would
definitely be a good thing.

-MC

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Re: [asterisk-users] [Fwd: Internal LAN echo problem]

2007-10-24 Thread Jonn Taylor

Eric ManxPower Wieling wrote:
 Any echo you hear on pure IP calls is caused by the endpoint phone.  You 
 cannot do ANYTHING about it on Asterisk.


 Jonn Taylor wrote:
   
 Any ideas ?

 Jonn

  Original Message 
 Subject: [asterisk-users] Internal LAN echo problem
 Date:Wed, 24 Oct 2007 08:34:32 -0500
 From:Jonn R Taylor [EMAIL PROTECTED]
 Reply-To:Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 To:  Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com



 Hi all,

 I have an internal echo problem on my LAN only. I replaced the LAN 
 switch with a new linksys 2024 with QOS and seemed to help but not fix 
 the problem. Any ideas? Here in my setup - Dell PE6400 Dual 700, 
 Asterisk 1.2.24/FreePBX, 2-NIC cards, one with a public ip and one with 
 an internal ip, one PAP2, one SPA3102 and 2 BT101. I know that bt's are 
 cheap that are known for echo problem in the handset. I have one remote 
 user that never has a problem. I have a remote test server at home 
 connect via IAX with no problems, also a PAP2 with no problem. External 
 faxing from the rest of the world via our voip provider is working 
 great. One strange thing that I noticed is that we can not fax to our 
 iaxmodem, ATA --- iaxmodem, but works perfect ATA --- rx_fax. Not sure 
 why either.
 

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That does not make sense. I can any one of these ata's or phones and 
connect them to the public ip side and they work fine.

Jonn

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Re: [asterisk-users] OSLEC and zaptel-1.4.5.1

On Wed, Oct 24, 2007 at 03:03:01PM +0100, Alan Lord wrote:
 Hi all,
 
 After reading great things about the OSLEC Echo Canceller 
 (http://www.rowetel.com/ucasterisk/oslec) and seeing the reactions of 
 people who have tried it on a recent Trixbox thread 
 (http://www.trixbox.org/forums/trixbox-forums/open-discussion/need-people-echo-problems),
  
 it sounds like it is the bees knees for sorting out echo problems with 
 cards like the x100p.
 
 Has anyone managed to get oslec to work with recent zaptel and kernel 
 (I'm running 2.6.23)?

I use it at home iwth zaptel 1.4.5.1 and kernel 2.6.18 of Debian Etch. I 
know it to build successfully with 2.6.22 .

You can find up-to-date OSLEC support (minimal patch an d an oslec
subdirectory) in recent zaptel packages of Debian. You need to set
ECHO_CAN_NAME=OSLEC to build OSLEC as the echo canceller.

-- 
   Tzafrir Cohen   
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] [Fwd: Internal LAN echo problem]

Buy a Polycom 301 off ebay and see if it echos on your LAN.

Thanks,
Steve Totaro

Jonn Taylor wrote:
 Any ideas ?

 Jonn

  Original Message 
 Subject:  [asterisk-users] Internal LAN echo problem
 Date: Wed, 24 Oct 2007 08:34:32 -0500
 From: Jonn R Taylor [EMAIL PROTECTED]
 Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 To:   Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com



 Hi all,

 I have an internal echo problem on my LAN only. I replaced the LAN 
 switch with a new linksys 2024 with QOS and seemed to help but not fix 
 the problem. Any ideas? Here in my setup - Dell PE6400 Dual 700, 
 Asterisk 1.2.24/FreePBX, 2-NIC cards, one with a public ip and one with 
 an internal ip, one PAP2, one SPA3102 and 2 BT101. I know that bt's are 
 cheap that are known for echo problem in the handset. I have one remote 
 user that never has a problem. I have a remote test server at home 
 connect via IAX with no problems, also a PAP2 with no problem. External 
 faxing from the rest of the world via our voip provider is working 
 great. One strange thing that I noticed is that we can not fax to our 
 iaxmodem, ATA --- iaxmodem, but works perfect ATA --- rx_fax. Not sure 
 why either.

 Jonn

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Re: [asterisk-users] OSLEC and zaptel-1.4.5.1

On Wed, Oct 24, 2007 at 01:20:31PM -0400, Dave Fullerton wrote:

 This looks like it is isolated to 1.4.5.x. 

Right. 

 It looks like digium added 

Just to set the record straight, it was me who added it, and thus caused
hte changed behaviour you noticed here. The behaviour was restored in later 
1.4.6 by qwell of Digium (thanks)

 a method of selecting the echo canceler by using environment variables but 
 didn't get it quite right. It appears to be fixed in 1.4.6. In 1.4.6 it 
 will use the selection made in zconfig.h UNLESS you set ECHO_CAN_NAME to 
 your preferred echo canceler (OSLEC,MG2,etc) to override it.

-- 
   Tzafrir Cohen   
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] AMI ActionID.... Doesn't work

2007-10-24 Thread Philipp Kempgen

Douglas Garstang wrote:

 Is it well known that setting the ActionID when connecting to AMI has 
 absolutely no effect?
 Is this fixed in Asterisk 1.4?
 
 If you add an ActionID to your Originate command for example, it looks like 
 the only events that come back with an ActionID associated are the initial 
 response, OriginateSuccess and OriginateFailure. That's it. No other events 
 have an ActionID associated.

Correct. That's how it's supposed to be.

 This pretty much makes the AMI useless.

No. That way it's possible to match responses and actions
because the order in which the responses arrive is not
guaranteed.

 What about all the other events? Newcallerid, Newstate, Link, Unlink and 
 REALLY importantly the CDR events.

I think you're looking for some kind of unique id for all
the AMI packets belonging to a specific call(?).

Regards,
  Philipp Kempgen

-- 
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
  Asterisk? - http://www.das-asterisk-buch.de

Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998

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Re: [asterisk-users] OSLEC and zaptel-1.4.5.1

Tzafrir Cohen wrote:
 On Wed, Oct 24, 2007 at 01:20:31PM -0400, Dave Fullerton wrote:
 
 This looks like it is isolated to 1.4.5.x. 
 
 Right. 
 
 It looks like digium added 
 
 Just to set the record straight, it was me who added it, and thus caused
 hte changed behaviour you noticed here. The behaviour was restored in later 
 1.4.6 by qwell of Digium (thanks)
 
 a method of selecting the echo canceler by using environment variables but 
 didn't get it quite right. It appears to be fixed in 1.4.6. In 1.4.6 it 
 will use the selection made in zconfig.h UNLESS you set ECHO_CAN_NAME to 
 your preferred echo canceler (OSLEC,MG2,etc) to override it.
 

Hi all.

The small tweak suggested by Marco Signorini did the trick.

I have oslec running on my cloned x100p card and it is fantastic. We 
have no more echo! *

* Well, I am using a Linux Ubuntu Desktop with the Twinkle Soft SIP 
phone and my audio device is the Polycom Communicator. Now the Polycom 
was built mainly for Skype and they have considerable echo cancellation 
technology built into their Windows *only* driver software. So it used 
to be the cause of much echo unless I connected a headset to the socket 
on the Communicator itself.

However, with the OSLEC running I can now use the Polycom handsfree and 
I hear almost zero echo (almost imperceptible).

I will drop the author a note and suggest that someone who understands 
this stuff, try and build a USB driver for devices like the Polycom 
using the OSLEC technology...

Thanks for the initial response Marco.

And anyone who has echo problems with x100p or other analogue cards 
should really give this a try. Most of the experiences I have read about 
have been very positive. Mine also :-)

Alan


-- 
The way out is open!
http://www.theopensourcerer.com


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[asterisk-users] bugs.digium.com

2007-10-24 Thread Doug Lytle


The bug tracker seems to be down.

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] AstManProxy Host Prefix?

2007-10-24 Thread asterisk

What would be nice if it you could specify the host per user in
astmanproy.users  
Anyone interested in making the change? $$$

Doug

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Richard
Lyman
Sent: Wednesday, October 24, 2007 1:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] AstManProxy Host Prefix?

Douglas Garstang wrote:
 Can the Asterisk Manager Proxy, AstManProxy, prefix the host name that
output applies to, to the start of each line? If you are proxying
multiple systems, how can it uniquely identify the output from each
system?

 Thanks,
 Doug.

   
each Event block should have a

Server: .

appended to it.


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[asterisk-users] reload manager.conf

2007-10-24 Thread Bob Pierce

I've made a change to my manager.conf file in asterisk 1.2.18

Is there a way to reload that config file from the CLI without
restarting asterisk?

Bob

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Re: [asterisk-users] bugs.digium.com

2007-10-24 Thread Michiel van Baak

On 15:38, Wed 24 Oct 07, Doug Lytle wrote:
 
 The bug tracker seems to be down.

And so is the public svn and downloads.digium.com and
ftp.digium.com and the websvn.
They are working on it.
-- 

Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD

Why is it drug addicts and computer afficionados are both called users?


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Re: [asterisk-users] [Fwd: Internal LAN echo problem]

2007-10-24 Thread Eric ManxPower Wieling

Jonn Taylor wrote:
 Eric ManxPower Wieling wrote:
 Any echo you hear on pure IP calls is caused by the endpoint phone.  You 
 cannot do ANYTHING about it on Asterisk.


 Jonn Taylor wrote:
   
 Any ideas ?

 Jonn

  Original Message 
 Subject:[asterisk-users] Internal LAN echo problem
 Date:   Wed, 24 Oct 2007 08:34:32 -0500
 From:   Jonn R Taylor [EMAIL PROTECTED]
 Reply-To:   Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com



 Hi all,

 I have an internal echo problem on my LAN only. I replaced the LAN 
 switch with a new linksys 2024 with QOS and seemed to help but not fix 
 the problem. Any ideas? Here in my setup - Dell PE6400 Dual 700, 
 Asterisk 1.2.24/FreePBX, 2-NIC cards, one with a public ip and one with 
 an internal ip, one PAP2, one SPA3102 and 2 BT101. I know that bt's are 
 cheap that are known for echo problem in the handset. I have one remote 
 user that never has a problem. I have a remote test server at home 
 connect via IAX with no problems, also a PAP2 with no problem. External 
 faxing from the rest of the world via our voip provider is working 
 great. One strange thing that I noticed is that we can not fax to our 
 iaxmodem, ATA --- iaxmodem, but works perfect ATA --- rx_fax. Not sure 
 why either.

 That does not make sense. I can any one of these ata's or phones and 
 connect them to the public ip side and they work fine.

It can make sense or not make sense, but you cannot have echo on a pure 
VoIP call unless the endpoints introduce it.


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Re: [asterisk-users] AstManProxy Host Prefix?

2007-10-24 Thread Sean Bright

I can do it for $10,000

On 10/24/07, asterisk [EMAIL PROTECTED] wrote:

 What would be nice if it you could specify the host per user in
 astmanproy.users
 Anyone interested in making the change? $$$

 Doug

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Richard
 Lyman
 Sent: Wednesday, October 24, 2007 1:45 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] AstManProxy Host Prefix?

 Douglas Garstang wrote:
  Can the Asterisk Manager Proxy, AstManProxy, prefix the host name that
 output applies to, to the start of each line? If you are proxying
 multiple systems, how can it uniquely identify the output from each
 system?
 
  Thanks,
  Doug.
 
 
 each Event block should have a

 Server: .

 appended to it.


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Re: [asterisk-users] bugs.digium.com

2007-10-24 Thread Michiel van Baak

On 22:06, Wed 24 Oct 07, Michiel van Baak wrote:
 On 15:38, Wed 24 Oct 07, Doug Lytle wrote:
  
  The bug tracker seems to be down.
 
 And so is the public svn and downloads.digium.com and
 ftp.digium.com and the websvn.
 They are working on it.

And it's working again for me
-- 

Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD

Why is it drug addicts and computer afficionados are both called users?


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Re: [asterisk-users] reload manager.conf

2007-10-24 Thread Richard Lyman

Bob Pierce wrote:
 I've made a change to my manager.conf file in asterisk 1.2.18

 Is there a way to reload that config file from the CLI without
 restarting asterisk?

 Bob

   
every time there is a new connection to the asterisk manager interface, 
the manager.conf file is reread.
(meaning, it reloads itself)


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[asterisk-users] TE210P issues

2007-10-24 Thread Jerry Geis

I have a box with a TE210P. Things work for a while then stop when 
making call files.
I get NOANSWER as the return code (right away).

I am running asterisk 1.2.12.1, libpri 1.2.3 and zap 1.2.9.1

When I try to update to newer zaptel the machine locks when loading the 
zaptel drivers.

I tried to manually load the wct1xxp module (I think that is the one for 
the dual T1 card???)
and the machine locks. I am in a remote location so I cannot see if 
anything is on the console.

I tried jumping to 1.4 and the same thing happens.
I have updated quite a few asterisk boxes remotely and never had this 
issue before.

Last thing I tried was chkconfig zaptel off, reboot, then try loading 
in new version and the same thing happened.
It locked up.

After rebooting I put back the old zaptel and it works again for  awhile.

What shall I try?


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Re: [asterisk-users] OSLEC and zaptel-1.4.5.1

2007-10-24 Thread marcotasto

Alan, 
I'm glad to see that you are able to run zaptel and OSLEC following my tweak!
Some days ago I've sent to David Rowe a little patch that preserves the echo 
cancel status between calls.
I'm using it since several weeks with my TDM400P home based PBX and I think 
that's a really effective solution. Unfortunately I can't test patches in all 
possible environments because I've only a single channel FXO.
I think David is still testing the patch before releasing it on the official 
OSLEC repository.

Thank you and best regards,

Marco Signorini.


 Hi all.
 
 The small tweak suggested by Marco Signorini did the trick.
 
 I have oslec running on my cloned x100p card and it is fantastic. We
 have no more echo! *
 
 * Well, I am using a Linux Ubuntu Desktop with the Twinkle Soft SIP
 phone and my audio device is the Polycom Communicator. Now the Polycom
 was built mainly for Skype and they have considerable echo cancellation
 technology built into their Windows *only* driver software. So it used
 to be the cause of much echo unless I connected a headset to the socket
 on the Communicator itself.
 
 However, with the OSLEC running I can now use the Polycom handsfree and
 I hear almost zero echo (almost imperceptible).
 
 I will drop the author a note and suggest that someone who understands
 this stuff, try and build a USB driver for devices like the Polycom
 using the OSLEC technology...
 
 Thanks for the initial response Marco.
 
 And anyone who has echo problems with x100p or other analogue cards
 should really give this a try. Most of the experiences I have read about
 have been very positive. Mine also :-)
 
 Alan
 
 
 -- 
 The way out is open!
 http://www.theopensourcerer.com
 

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Re: [asterisk-users] [Fwd: Internal LAN echo problem]

2007-10-24 Thread kevin bergner

On 10/24/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
 Jonn Taylor wrote:
  Eric ManxPower Wieling wrote:
  Any echo you hear on pure IP calls is caused by the endpoint phone.  You
  cannot do ANYTHING about it on Asterisk.
 
 
  Jonn Taylor wrote:
 
  Any ideas ?
 
  Jonn
 
   Original Message 
  Subject:[asterisk-users] Internal LAN echo problem
  Date:   Wed, 24 Oct 2007 08:34:32 -0500
  From:   Jonn R Taylor [EMAIL PROTECTED]
  Reply-To:   Asterisk Users Mailing List - Non-Commercial Discussion
  asterisk-users@lists.digium.com
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  asterisk-users@lists.digium.com
 
 
 
  Hi all,
 
  I have an internal echo problem on my LAN only. I replaced the LAN
  switch with a new linksys 2024 with QOS and seemed to help but not fix
  the problem. Any ideas? Here in my setup - Dell PE6400 Dual 700,
  Asterisk 1.2.24/FreePBX, 2-NIC cards, one with a public ip and one with
  an internal ip, one PAP2, one SPA3102 and 2 BT101. I know that bt's are
  cheap that are known for echo problem in the handset. I have one remote
  user that never has a problem. I have a remote test server at home
  connect via IAX with no problems, also a PAP2 with no problem. External
  faxing from the rest of the world via our voip provider is working
  great. One strange thing that I noticed is that we can not fax to our
  iaxmodem, ATA --- iaxmodem, but works perfect ATA --- rx_fax. Not sure
  why either.

  That does not make sense. I can any one of these ata's or phones and
  connect them to the public ip side and they work fine.

 It can make sense or not make sense, but you cannot have echo on a pure
 VoIP call unless the endpoints introduce it.


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i have seen this when the  headset volume is too high and simply
lowering the volume addressed the problem

as others have said an echo is simply not possible


-- 
kevin

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Re: [asterisk-users] TE210P issues

Calling Digium.  Post your /var/log/messages and /var/log/asterisk/full 
(just anything that looks relevant). 

Try a Sangoma card.

Thanks,
Steve

Jerry Geis wrote:
 I have a box with a TE210P. Things work for a while then stop when 
 making call files.
 I get NOANSWER as the return code (right away).

 I am running asterisk 1.2.12.1, libpri 1.2.3 and zap 1.2.9.1

 When I try to update to newer zaptel the machine locks when loading the 
 zaptel drivers.

 I tried to manually load the wct1xxp module (I think that is the one for 
 the dual T1 card???)
 and the machine locks. I am in a remote location so I cannot see if 
 anything is on the console.

 I tried jumping to 1.4 and the same thing happens.
 I have updated quite a few asterisk boxes remotely and never had this 
 issue before.

 Last thing I tried was chkconfig zaptel off, reboot, then try loading 
 in new version and the same thing happened.
 It locked up.

 After rebooting I put back the old zaptel and it works again for  awhile.

 What shall I try?


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Re: [asterisk-users] reload manager.conf

2007-10-24 Thread Bob Pierce

On Wed, 2007-10-24 at 13:31 -0700, Richard Lyman wrote:
 every time there is a new connection to the asterisk manager
 interface, the manager.conf file is reread.
 (meaning, it reloads itself)

Great. Thanks for your help!

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Re: [asterisk-users] [Fwd: Internal LAN echo problem]

Let me screw this thread up by top posting now.

Could echo be caused by late packets if jitterbuffer is on or something 
or would that just cause lag? 

Thanks,
Steve

kevin bergner wrote:
 On 10/24/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
   
 Jonn Taylor wrote:
 
 Eric ManxPower Wieling wrote:
   
 Any echo you hear on pure IP calls is caused by the endpoint phone.  You
 cannot do ANYTHING about it on Asterisk.


 Jonn Taylor wrote:

 
 Any ideas ?

 Jonn

  Original Message 
 Subject:[asterisk-users] Internal LAN echo problem
 Date:   Wed, 24 Oct 2007 08:34:32 -0500
 From:   Jonn R Taylor [EMAIL PROTECTED]
 Reply-To:   Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com



 Hi all,

 I have an internal echo problem on my LAN only. I replaced the LAN
 switch with a new linksys 2024 with QOS and seemed to help but not fix
 the problem. Any ideas? Here in my setup - Dell PE6400 Dual 700,
 Asterisk 1.2.24/FreePBX, 2-NIC cards, one with a public ip and one with
 an internal ip, one PAP2, one SPA3102 and 2 BT101. I know that bt's are
 cheap that are known for echo problem in the handset. I have one remote
 user that never has a problem. I have a remote test server at home
 connect via IAX with no problems, also a PAP2 with no problem. External
 faxing from the rest of the world via our voip provider is working
 great. One strange thing that I noticed is that we can not fax to our
 iaxmodem, ATA --- iaxmodem, but works perfect ATA --- rx_fax. Not sure
 why either.
   
 That does not make sense. I can any one of these ata's or phones and
 connect them to the public ip side and they work fine.
   
 It can make sense or not make sense, but you cannot have echo on a pure
 VoIP call unless the endpoints introduce it.


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 i have seen this when the  headset volume is too high and simply
 lowering the volume addressed the problem

 as others have said an echo is simply not possible


   


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Re: [asterisk-users] [Fwd: Internal LAN echo problem]

2007-10-24 Thread David Gomillion

On 10/24/07, Steve Totaro [EMAIL PROTECTED] wrote:

 Let me screw this thread up by top posting now.

 Could echo be caused by late packets if jitterbuffer is on or something
 or would that just cause lag?

 Thanks,
 Steve



So, does this qualify as an in-line reply, or a top post? Maybe it's a
medium post ;)

If both calls were in the LAN, chances are good that the phones will have
re-invited to go around the SIP server. If that's the case, then it
shouldn't be a problem.

Now, if dial options, recording, or SIP settings prevent reinvites, then
this might be part of the problem.

kevin bergner wrote:
  On 10/24/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
 
  Jonn Taylor wrote:
 
  Eric ManxPower Wieling wrote:
 
  Any echo you hear on pure IP calls is caused by the endpoint
 phone.  You
  cannot do ANYTHING about it on Asterisk.
 
 
  Jonn Taylor wrote:
 
 
  Any ideas ?
 
  Jonn
 
   Original Message 
  Subject:[asterisk-users] Internal LAN echo problem
  Date:   Wed, 24 Oct 2007 08:34:32 -0500
  From:   Jonn R Taylor [EMAIL PROTECTED]
  Reply-To:   Asterisk Users Mailing List - Non-Commercial Discussion
  asterisk-users@lists.digium.com
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  asterisk-users@lists.digium.com
 
 
 
  Hi all,
 
  I have an internal echo problem on my LAN only. I replaced the LAN
  switch with a new linksys 2024 with QOS and seemed to help but not
 fix
  the problem. Any ideas? Here in my setup - Dell PE6400 Dual 700,
  Asterisk 1.2.24/FreePBX, 2-NIC cards, one with a public ip and one
 with
  an internal ip, one PAP2, one SPA3102 and 2 BT101. I know that bt's
 are
  cheap that are known for echo problem in the handset. I have one
 remote
  user that never has a problem. I have a remote test server at home
  connect via IAX with no problems, also a PAP2 with no problem.
 External
  faxing from the rest of the world via our voip provider is working
  great. One strange thing that I noticed is that we can not fax to
 our
  iaxmodem, ATA --- iaxmodem, but works perfect ATA --- rx_fax. Not
 sure
  why either.
 
  That does not make sense. I can any one of these ata's or phones and
  connect them to the public ip side and they work fine.
 
  It can make sense or not make sense, but you cannot have echo on a pure
  VoIP call unless the endpoints introduce it.
 
 
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  To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
 
  i have seen this when the  headset volume is too high and simply
  lowering the volume addressed the problem
 
  as others have said an echo is simply not possible
 
 
 


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Re: [asterisk-users] [Fwd: Internal LAN echo problem]

2007-10-24 Thread David Gomillion

On 10/24/07, David Gomillion [EMAIL PROTECTED] wrote:

 On 10/24/07, Steve Totaro [EMAIL PROTECTED] wrote:
 
  Let me screw this thread up by top posting now.
 
  Could echo be caused by late packets if jitterbuffer is on or something
  or would that just cause lag?
 
  Thanks,
  Steve



 So, does this qualify as an in-line reply, or a top post? Maybe it's a
 medium post ;)

 If both calls were in the LAN, chances are good that the phones will have
 re-invited to go around the SIP server. If that's the case, then it
 shouldn't be a problem.

 Now, if dial options, recording, or SIP settings prevent reinvites, then
 this might be part of the problem.



Sorry, I need to clarify my own post. By part of the problem, I mean
magnifying the effect. The real problem is the handset leaking, probably too
much sidetone.

Anyway, the more the delay, the more noticeable this echo will usually be.

kevin bergner wrote:
   On 10/24/07, Eric ManxPower Wieling  [EMAIL PROTECTED] wrote:
  
   Jonn Taylor wrote:
  
   Eric ManxPower Wieling wrote:
  
   Any echo you hear on pure IP calls is caused by the endpoint
  phone.  You
   cannot do ANYTHING about it on Asterisk.
  
  
   Jonn Taylor wrote:
  
  
   Any ideas ?
  
   Jonn
  
    Original Message 
   Subject:[asterisk-users] Internal LAN echo problem
   Date:   Wed, 24 Oct 2007 08:34:32 -0500
   From:   Jonn R Taylor [EMAIL PROTECTED]
   Reply-To:   Asterisk Users Mailing List - Non-Commercial
  Discussion
   asterisk-users@lists.digium.com
   To: Asterisk Users Mailing List - Non-Commercial
  Discussion
asterisk-users@lists.digium.com
  
  
  
   Hi all,
  
   I have an internal echo problem on my LAN only. I replaced the LAN
 
   switch with a new linksys 2024 with QOS and seemed to help but not
  fix
   the problem. Any ideas? Here in my setup - Dell PE6400 Dual 700,
   Asterisk 1.2.24 /FreePBX, 2-NIC cards, one with a public ip and
  one with
   an internal ip, one PAP2, one SPA3102 and 2 BT101. I know that
  bt's are
   cheap that are known for echo problem in the handset. I have one
  remote
   user that never has a problem. I have a remote test server at home
   connect via IAX with no problems, also a PAP2 with no problem.
  External
   faxing from the rest of the world via our voip provider is working
 
   great. One strange thing that I noticed is that we can not fax to
  our
   iaxmodem, ATA --- iaxmodem, but works perfect ATA --- rx_fax.
  Not sure
   why either.
  
   That does not make sense. I can any one of these ata's or phones and
   connect them to the public ip side and they work fine.
  
   It can make sense or not make sense, but you cannot have echo on a
  pure
   VoIP call unless the endpoints introduce it.
  
  
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   asterisk-users mailing list
   To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
  
  
  
   i have seen this when the  headset volume is too high and simply
   lowering the volume addressed the problem
  
   as others have said an echo is simply not possible
  
  
  
 
 
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 http://lists.digium.com/mailman/listinfo/asterisk-users
 


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Re: [asterisk-users] [Fwd: Internal LAN echo problem]

2007-10-24 Thread Jason Parker

See response in-random-lined.

David Gomillion wrote:
 
 
 On 10/24/07, *David Gomillion* [EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED] wrote:
 
 On 10/24/07, *Steve Totaro* [EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED] wrote:
 
 Let me screw this thread up by top posting now.
 
 Could echo be caused by late packets if jitterbuffer is on or
 something
 or would that just cause lag?
 
 Thanks,
 Steve
 
 
 
 So, does this qualify as an in-line reply, or a top post? Maybe it's
 a medium post ;)
 
 If both calls were in the LAN, chances are good that the phones will
 have re-invited to go around the SIP server. If that's the case,
 then it shouldn't be a problem.
 
 Now, if dial options, recording, or SIP settings prevent reinvites,
 then this might be part of the problem. 
 
 
 
 Sorry, I need to clarify my own post. By part of the problem, I mean
 magnifying the effect. The real problem is the handset leaking, probably
 too much sidetone.
 
 Anyway, the more the delay, the more noticeable this echo will usually be.
 
 kevin bergner wrote:
  On 10/24/07, Eric ManxPower Wieling  [EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED] wrote:
 
  Jonn Taylor wrote:
 
  Eric ManxPower Wieling wrote:
 
  Any echo you hear on pure IP calls is caused by the endpoint
 phone.  You
  cannot do ANYTHING about it on Asterisk.
 
 
  Jonn Taylor wrote:
 
 
  Any ideas ?
 
  Jonn
 
   Original Message 
  Subject:[asterisk-users] Internal LAN echo problem
  Date:   Wed, 24 Oct 2007 08:34:32 -0500
  From:   Jonn R Taylor [EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED]


Will the madness never end?


  Reply-To:   Asterisk Users Mailing List - Non-Commercial
 Discussion
  asterisk-users@lists.digium.com
 mailto:asterisk-users@lists.digium.com
  To: Asterisk Users Mailing List - Non-Commercial
 Discussion
   asterisk-users@lists.digium.com
 mailto:asterisk-users@lists.digium.com
 
 
 
  Hi all,
 
  I have an internal echo problem on my LAN only. I replaced
 the LAN
  switch with a new linksys 2024 with QOS and seemed to help
 but not fix
  the problem. Any ideas? Here in my setup - Dell PE6400 Dual
 700,
  Asterisk 1.2.24 /FreePBX, 2-NIC cards, one with a public ip
 and one with
  an internal ip, one PAP2, one SPA3102 and 2 BT101. I know
 that bt's are
  cheap that are known for echo problem in the handset. I
 have one remote
  user that never has a problem. I have a remote test server
 at home
  connect via IAX with no problems, also a PAP2 with no
 problem. External
  faxing from the rest of the world via our voip provider is
 working
  great. One strange thing that I noticed is that we can not
 fax to our
  iaxmodem, ATA --- iaxmodem, but works perfect ATA ---
 rx_fax. Not sure
  why either.
 
  That does not make sense. I can any one of these ata's or
 phones and
  connect them to the public ip side and they work fine.
 
  It can make sense or not make sense, but you cannot have echo
 on a pure
  VoIP call unless the endpoints introduce it.
 
 
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 http://www.api-digital.com--
 
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  To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
 http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
 
  i have seen this when the  headset volume is too high and simply
  lowering the volume addressed the problem
 
  as others have said an echo is simply not possible
 
 
 
 
 
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 http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
 
 
 
 
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-- 
Jason

Re: [asterisk-users] Voicemail playback on iPhone

2007-10-24 Thread Eric Chamberlain

Convert the voicemail to a mp3 file.

As of firmware version 1.1.1, the iPhone mail application will recognize, but 
not play wav attachments.  But the mail application does, recognize and play 
mp3 file attachments.

--
Eric Chamberlain, CISSP
Chief Technical Officer
Voxilla - http://voxilla.com/

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Jason Lixfeld
 Sent: Wednesday, October 24, 2007 7:46 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Voicemail playback on iPhone
 
 Sorry, it's clear my question was far too vague.
 
 To clarify, is there a recipe to make * record voicemail in a format
 that allows playback on iPhone's media/music player playback for
 voicemails that are received say, in an email message.
 
 It seems the * voicemail defaults don't work.  This link seems to
 describe codecs that do work, however I haven't seen any indications
 as to whether * voicemail can be tweaked to record in any of the
 supported formats:  http://www.kehlet.cx/
 
 Any success out there?
 
 On 22-Oct-07, at 7:38 PM, Ron Stephan wrote:
 
 
  Trick question I assume?
 
  It was mind numbingly simple on my iPhone...(though none of the
  voice mail worked when London a few weeks ago).
 
  - tap voice mail -
  - tap speaker (upper right) until it turns blue (is activate)
  - tap the message you want to playback
  - use assorted  controls to delete - replay etc.
 
 
  Now...if the question is ... how do you get asterisk voice mail to
  show up on an iPhone...I am all ears.  Groovy concept - if
  anybody has a hack - I'd love to see it.
 
 
 
  Elvis
 
 
 
 
 
 
 
  -Original Message-
  From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED]
  ] On Behalf Of Jason Lixfeld
  Sent: Monday, October 22, 2007 4:16 PM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: [asterisk-users] Voicemail playback on iPhone
 
  Anyone managed to get this to work?  What's the recipe?
 
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  __ NOD32 2607 (20071022) Information __
 
  This message was checked by NOD32 antivirus system.
  http://www.eset.com
 
 
 
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[asterisk-users] How to get TCP access to CDR Master.csv

2007-10-24 Thread asterisk-users

Hi.

 

I'd like to get access to the CDR's generated by Asterisk (1.4) in real-time
from a remote connection coming in on TCP. Basically what I have is a
Windows application that is used to process incoming, outgoing and missed
call records putting them into a database for some analysing etc. This app
can connect to a TCP server and read from this connection the CDR's as they
are coming in (being generated).

 

I can't find this as a feature of the standard Asterisk... but maybe I'm
missing something? The closest I could get is something around the manager
api but it's not really what I'm after. I'd like to access the CDR's them
selves.

 

Being a (more or less) novice Linux user the only thing I can think of is
trying to do this using Perl scripts where it would set up a listening
socket and when connection is received it would do something like (in
princip, not managed to do this properly yet):

 

...

print $connection `tail -f /var/log/asterisk/cdr-custom/Master.csv`

...

 

But even this is full of issues to solve. Things like only one connection at
a time (which I can live with) from the remote computer. The fact that tail
will not write to the socket (yeah, a major issue probably) which I'm
thinking of trying to solve by reading line by line somehow and writing back
to the socket... not even sure if this is possible.

 

So basically I'm hoping someone has a nice solution for this. With or witout
scripting, external programs of some sort (runnin ubuntu 7.04 or 6.06) or
whatever works. I'd really appreciate your input here.

 

Sincerely, Baldvin

 

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Re: [asterisk-users] How to get TCP access to CDR Master.csv

2007-10-24 Thread John Hass





Hello,

I am not sure if I totally understand the question but if your looking
to stream the connection you could create a simple bash script like this


#!/bin/bash
while true; do
 tail -f /var/log/asterisk/cdr-custom/Master.csv | nc -p 1024 -l
done

There probably is a better solution then this, but this will get you
going

>From any machine you should be able to type `telnet ip.of.machine 1024`


--John


[EMAIL PROTECTED] wrote:

  
  
  

  
  Hi.
   
  I‘d like to get access to the CDR‘s generated by
Asterisk (1.4) in real-time from a remote connection coming in on TCP.
Basically what I have is a Windows application that is used to process
incoming, outgoing and missed call records putting them into a database
for
some analysing etc. This app can connect to a TCP server and read from
this
connection the CDR‘s as they  are coming in (being generated).
   
  I can‘t find this as a „feature“ of the
standard Asterisk... but maybe I‘m missing something? The closest I
could
get is something around the manager api but it‘s not really what I‘m
after. I‘d like to access the CDR‘s them selves.
   
  Being a (more or less) novice Linux user the
only thing I
can think of is trying to do this using Perl scripts where it would set
up a
listening socket and when connection is received it would do something
like (in
princip, not managed to do this properly yet):
   
  ...
  print $connection `tail –f
/var/log/asterisk/cdr-custom/Master.csv`
  ...
   
  But even this is full of issues to solve. Things
like only
one connection at a time (which I can live with) from the remote
computer. The
fact that tail will not write to the socket (yeah, a major issue
probably)
which I‘m thinking of trying to solve by reading line by line somehow
and
writing back to the socket... not even sure if this is possible.
   
  So basically I‘m hoping someone has a nice
solution
for this. With or witout scripting, external programs of some sort
(runnin
ubuntu 7.04 or 6.06) or whatever works. I‘d really appreciate your
input
here.
   
  Sincerely, Baldvin
   
  
  

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Re: [asterisk-users] How to get TCP access to CDR Master.csv

2007-10-24 Thread Klaverstyn, David C

I’m no expert in this field bit I would have though logging the calls to MySQL 
and then queering the MySQL database would be the best not to mention the 
easiest way to get the details you are looking for.

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Hass
Sent: Thursday, 25 October 2007 8:39 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to get TCP access to CDR Master.csv

Hello,

I am not sure if I totally understand the question but if your looking to 
stream the connection you could create a simple bash script like this

#!/bin/bash
while true; do
 tail -f /var/log/asterisk/cdr-custom/Master.csv | nc -p 1024 -l
done

There probably is a better solution then this, but this will get you going

From any machine you should be able to type `telnet ip.of.machine 1024`

--John

[EMAIL PROTECTED] wrote: 

Hi.

I‘d like to get access to the CDR‘s generated by Asterisk (1.4) in real-time 
from a remote connection coming in on TCP. Basically what I have is a Windows 
application that is used to process incoming, outgoing and missed call records 
putting them into a database for some analysing etc. This app can connect to a 
TCP server and read from this connection the CDR‘s as they  are coming in 
(being generated).

I can‘t find this as a „feature“ of the standard Asterisk... but maybe I‘m 
missing something? The closest I could get is something around the manager api 
but it‘s not really what I‘m after. I‘d like to access the CDR‘s them selves.

Being a (more or less) novice Linux user the only thing I can think of is 
trying to do this using Perl scripts where it would set up a listening socket 
and when connection is received it would do something like (in princip, not 
managed to do this properly yet):

...

print $connection `tail –f /var/log/asterisk/cdr-custom/Master.csv`

...

But even this is full of issues to solve. Things like only one connection at a 
time (which I can live with) from the remote computer. The fact that tail will 
not write to the socket (yeah, a major issue probably) which I‘m thinking of 
trying to solve by reading line by line somehow and writing back to the 
socket... not even sure if this is possible.

So basically I‘m hoping someone has a nice solution for this. With or witout 
scripting, external programs of some sort (runnin ubuntu 7.04 or 6.06) or 
whatever works. I‘d really appreciate your input here.

Sincerely, Baldvin

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Re: [asterisk-users] Voicemail playback on iPhone

2007-10-24 Thread Eric Chamberlain

I tested this again, and wav files do play as attachments with firmware 1.1.1.

--
Eric Chamberlain, CISSP
Chief Technical Officer
Voxilla - http://voxilla.com/

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Eric Chamberlain
 Sent: Wednesday, October 24, 2007 3:29 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Voicemail playback on iPhone
 
 Convert the voicemail to a mp3 file.
 
 As of firmware version 1.1.1, the iPhone mail application will recognize,
 but not play wav attachments.  But the mail application does, recognize
 and play mp3 file attachments.
 
 --
 Eric Chamberlain, CISSP
 Chief Technical Officer
 Voxilla - http://voxilla.com/
 
  -Original Message-
  From: [EMAIL PROTECTED] [mailto:asterisk-users-
  [EMAIL PROTECTED] On Behalf Of Jason Lixfeld
  Sent: Wednesday, October 24, 2007 7:46 AM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [asterisk-users] Voicemail playback on iPhone
 
  Sorry, it's clear my question was far too vague.
 
  To clarify, is there a recipe to make * record voicemail in a format
  that allows playback on iPhone's media/music player playback for
  voicemails that are received say, in an email message.
 
  It seems the * voicemail defaults don't work.  This link seems to
  describe codecs that do work, however I haven't seen any indications
  as to whether * voicemail can be tweaked to record in any of the
  supported formats:  http://www.kehlet.cx/
 
  Any success out there?
 
  On 22-Oct-07, at 7:38 PM, Ron Stephan wrote:
 
  
   Trick question I assume?
  
   It was mind numbingly simple on my iPhone...(though none of the
   voice mail worked when London a few weeks ago).
  
   - tap voice mail -
   - tap speaker (upper right) until it turns blue (is activate)
   - tap the message you want to playback
   - use assorted  controls to delete - replay etc.
  
  
   Now...if the question is ... how do you get asterisk voice mail to
   show up on an iPhone...I am all ears.  Groovy concept - if
   anybody has a hack - I'd love to see it.
  
  
  
   Elvis
  
  
  
  
  
  
  
   -Original Message-
   From: [EMAIL PROTECTED] [mailto:asterisk-users-
  [EMAIL PROTECTED]
   ] On Behalf Of Jason Lixfeld
   Sent: Monday, October 22, 2007 4:16 PM
   To: Asterisk Users Mailing List - Non-Commercial Discussion
   Subject: [asterisk-users] Voicemail playback on iPhone
  
   Anyone managed to get this to work?  What's the recipe?
  
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Re: [asterisk-users] [Fwd: Internal LAN echo problem]

2007-10-24 Thread Jonn R Taylor

Jason Parker wrote:
 See response in-random-lined.
 
 David Gomillion wrote:

 On 10/24/07, *David Gomillion* [EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED] wrote:

 On 10/24/07, *Steve Totaro* [EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED] wrote:

 Let me screw this thread up by top posting now.

 Could echo be caused by late packets if jitterbuffer is on or
 something
 or would that just cause lag?

 Thanks,
 Steve



 So, does this qualify as an in-line reply, or a top post? Maybe it's
 a medium post ;)

 If both calls were in the LAN, chances are good that the phones will
 have re-invited to go around the SIP server. If that's the case,
 then it shouldn't be a problem.

 Now, if dial options, recording, or SIP settings prevent reinvites,
 then this might be part of the problem. 



 Sorry, I need to clarify my own post. By part of the problem, I mean
 magnifying the effect. The real problem is the handset leaking, probably
 too much sidetone.

 Anyway, the more the delay, the more noticeable this echo will usually be.

 kevin bergner wrote:
  On 10/24/07, Eric ManxPower Wieling  [EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED] wrote:
 
  Jonn Taylor wrote:
 
  Eric ManxPower Wieling wrote:
 
  Any echo you hear on pure IP calls is caused by the endpoint
 phone.  You
  cannot do ANYTHING about it on Asterisk.
 
 
  Jonn Taylor wrote:
 
 
  Any ideas ?
 
  Jonn
 
   Original Message 
  Subject:[asterisk-users] Internal LAN echo problem
  Date:   Wed, 24 Oct 2007 08:34:32 -0500
  From:   Jonn R Taylor [EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED]
 
 
 Will the madness never end?
 
 
  Reply-To:   Asterisk Users Mailing List - Non-Commercial
 Discussion
  asterisk-users@lists.digium.com
 mailto:asterisk-users@lists.digium.com
  To: Asterisk Users Mailing List - Non-Commercial
 Discussion
   asterisk-users@lists.digium.com
 mailto:asterisk-users@lists.digium.com
 
 
 
  Hi all,
 
  I have an internal echo problem on my LAN only. I replaced
 the LAN
  switch with a new linksys 2024 with QOS and seemed to help
 but not fix
  the problem. Any ideas? Here in my setup - Dell PE6400 Dual
 700,
  Asterisk 1.2.24 /FreePBX, 2-NIC cards, one with a public ip
 and one with
  an internal ip, one PAP2, one SPA3102 and 2 BT101. I know
 that bt's are
  cheap that are known for echo problem in the handset. I
 have one remote
  user that never has a problem. I have a remote test server
 at home
  connect via IAX with no problems, also a PAP2 with no
 problem. External
  faxing from the rest of the world via our voip provider is
 working
  great. One strange thing that I noticed is that we can not
 fax to our
  iaxmodem, ATA --- iaxmodem, but works perfect ATA ---
 rx_fax. Not sure
  why either.
 
  That does not make sense. I can any one of these ata's or
 phones and
  connect them to the public ip side and they work fine.
 
  It can make sense or not make sense, but you cannot have echo
 on a pure
  VoIP call unless the endpoints introduce it.
 
 
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  i have seen this when the  headset volume is too high and simply
  lowering the volume addressed the problem
 
  as others have said an echo is simply not possible
 
 
 


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Re: [asterisk-users] How to get TCP access to CDR Master.csv

On Wed, Oct 24, 2007 at 10:29:41PM -, [EMAIL PROTECTED] wrote:
 Hi.
 
 I'd like to get access to the CDR's generated by Asterisk (1.4) in real-time
 from a remote connection coming in on TCP. Basically what I have is a
 Windows application that is used to process incoming, outgoing and missed
 call records putting them into a database for some analysing etc. This app
 can connect to a TCP server and read from this connection the CDR's as they
 are coming in (being generated).

CDR also generates manager events in real time. An Asterisk manager
listener can get them and notify your application.

-- 
   Tzafrir Cohen   
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] How to get TCP access to CDR Master.csv

2007-10-24 Thread asterisk-users

#!/bin/bash
while true; do
 tail -f /var/log/asterisk/cdr-custom/Master.csv | nc -p 1024 -l
done 

 

Thank you John, this bash script is exactly what I was looking for. Very 
simple, yet works.

 

As for doing this with insert into database and then polling for it... well I 
don‘t like polling. It‘s a good idea, but in the end, for this solution/in this 
case, the system reading the socket will in fact file the data (post 
processing) in a sql database for storing and querying.

 

tnx,

Baldvin

 

From: John Hass [mailto:[EMAIL PROTECTED] 
Sent: 24. október 2007 22:39
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to get TCP access to CDR Master.csv

 

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Re: [asterisk-users] [Fwd: Internal LAN echo problem]