Re: [asterisk-users] Real Time, sip.conf, [general]

[Olle E Johansson]

4 mar 2007 kl. 21.45 skrev André Santos:


Hi,

I am implementing de Real Time architecture, I would like to know  
if, their is any problem in putting the section [general] of the  
sip.conf file in the table of sippeers.


You can't mix the general section with sippeers for realtime.

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RE: [asterisk-users] Back to back E1 - asterisk <=> toshiba pbx - Calldroping issue

[Steve Totaro]

As these problems are very time sensitive and frustrating, I suggest you
document each change you make and do them one at a time so you can
actually know what the problem was and not introduce new problems in the
process.  

 

Find someone who is on the phone quite a bit and will give you an honest
evaluation of the call dropping situation (unless you yourself are
experiencing this issue too).  Some people are so quick to say, "It is
still happening" without starting the evaluation from a clean slate
after each change.

 

You may want to check your Asterisk log for more insight.
/var/log/asterisk/full.  Also you can turn on debugging on one span at a
time and see if you can find something there

 

Do you have a resetinterval set in zapata.conf?  If you can isolate the
dropped calls to the reset interval (watch the console, it will scroll
with each channel being reset) then set resetinterval=never.  If there
is no entry for resetinterval, add it and set it to never since it is
defaulted to on.

 

Also, try changing your second span timing from span=2,2,0,ccs,hdb3,crc4
to span=2,0,0,ccs,hdb3,crc4.  This in combination with your first span
should accept timing from the Telco and then supply it to your Toshiba,
I would actually try this first.



Another thought, It seems you have quite a lot of hardware in that box.
I am not sure how much is too much, but that would probably just rear
it's ugly head as poor audio.

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
  

  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vidura
Senadeera
Sent: Wednesday, March 07, 2007 2:15 AM
To: [EMAIL PROTECTED]
Cc: asterisk-users@lists.digium.com
Subject: [asterisk-users] Back to back E1 - asterisk <=> toshiba pbx -
Calldroping issue

 

 

Hi Team,

 

I have integrated asterisk with Toshiba analog PBX. NOw the live setup
is going.

 

Now I am facing call droping problem. It's happening ample time. 10-20
calls are droping every day.

 

What could be the reason. I attached latest zapata.conf file for your
information.

 

 

 

This is being a huge issue.

 

Highly appreciate your help on this regard.
 

Thanks & Regards,

Vidura Senadeera.


 

On 1/26/07, Vidura Senadeera <[EMAIL PROTECTED] > wrote: 

Dear Marco,

 

There is a huge problem i'm facing.

 

My asterisk server included with TDM2451E and 2 TE110p cards. One E1 i
conected to the telco. other E1 port i'm using to cros-connection with
toshiba pbx. My telco E1 d channels communicating well. but toshiba pbx
E1 not getting. d-channels are not getting up. 

what could be the issue. i'm using asterisk -1.2.14 and zaptel 1.2.12.

 

notes - if i put, zap show channels in asterisk cli. its only showing
the first 31 channels. but with ztcfg -vvv it showing al the channels.

 

my configs are

 

# Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" HDB3/CCS/CRC4 RED

#  Suntel E1 connection ==

span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16

# Span 2: WCT1/1 "Digium Wildcard TE110P T1/E1 Card 1"
#  Legacy PBX E1 connection ===

span=2,2,0,ccs,hdb3,crc4
bchan=32-46,48-62
dchan=47

# Span 3: WCTDM/0 "Wildcard TDM2400P Prototype Board 1"
fxoks=63
fxoks=64
fxoks=65
fxoks=66
fxoks=67
fxoks=68
fxoks=69
fxoks=70
fxoks=71
fxoks=72
fxoks=73
fxoks=74
fxoks=75 
fxoks=76
fxoks=77
fxoks=78
fxoks=79
fxoks=80
fxoks=81
fxoks=82
fxsks=83
fxsks=84
fxsks=85
fxsks=86

# Global data

loadzone= us
defaultzone = us

 Regards,

vidura

 



-- 
Thanks & Regards,
Vidura B. Senadeera. 




-- 
Thanks & Regards,
Vidura B. Senadeera. 




-- 
Thanks & Regards,
Vidura B. Senadeera. 

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[asterisk-users] Asterisk 1.4.1 - Calling problem

[Thomas Deillon]

Hi all,

 

I install the Asterisk 1.4.1 in order to use the T.38 pass-through, but
for the moment, I cannot even make call 

I have this WARNING: 

 

[Mar  7 11:32:09] WARNING[13395]: chan_sip.c:12290 handle_response:
Remote host can't match request BYE to call
'[EMAIL PROTECTED]'. Giving up. 

 

Do you know what is this error and what can I do to solve it ?

 

Thanks a lot for your help,

 

Thomas

 

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[asterisk-users] capi installation problem

[Giedrius Augys]

Hello,
I have problem with capi, I can't install it. I have putted all info what I
did and what I get :). I think you can suggest me how to solve this
problem.and thanking you in anticipation. I have ISDN Frtiz!Card PCI
v2.1and I want  to install it to my ubuntu box (kernel:
2.6.17-10-server). Using command "lspci -vv" , I can see that kernel finds
this card:
*02:0d.0 Network controller: AVM Audiovisuelles MKTG & Computer System GmbH
Fritz!PCI v2.0 ISDN (rev 02)
   Subsystem: AVM Audiovisuelles MKTG & Computer System GmbH Fritz!PCI
v2.0 ISDN
   Control: I/O+ Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr-
Stepping- SERR+ FastB2B-
   Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort-
SERR- http://www.asteriskguru.com/tutorials/avm_b1.html .I
have installed these packages:
*ii  capiutils  3.9.20060704-1  Utilities for
CAPI-capable ISDN cards
ii  libcapi20-33.9.20060704-1  libraries for
CAPI support
ii  libcapi20-dev  3.9.20060704-1  libraries for
CAPI support
ii  avm-fritz-firmware-2.6.17-10   3.11+2.6.17.7-10.1  Firmware for AVM
Fritz! ISDN hardware*

and downloaded firmaware from *
ftp://ftp.in-berlin.de/pub/capi4linux/firmware/b1/ .*My capi.conf is:
*b1pci   /usr/share/isdn/b1.t4   DSS1-   -
-   -   P2P*

Then I exec command capiinit start, I have noting on output, but it load
modules:
[EMAIL PROTECTED]:~# lsmod
Module  Size  Used by
b1pci  10624  0
b1dma  17412  1 b1pci
b1 25856  2 b1pci,b1dma
capi   19392  0
kernelcapi 49664  4 b1pci,b1dma,b1,capi
capifs  7176  2 capi
ipv6  271136  12
lp 12964  0
mISDN_l2   44288  0
mISDN_l1   13192  0
avmfritz   25740  0
mISDN_isac 17280  1 avmfritz
mISDN_core 75648  4 mISDN_l2,mISDN_l1,avmfritz,mISDN_isac*

But when I execute command cappinfo, I get :
[EMAIL PROTECTED]:~# capiinfo
capi not installed - No such device or address (6).
*
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Re: [asterisk-users] Building a new voicemail system... Testers needed!

[John Marvin]

Olle E Johansson wrote:

Friends in the Asterisk community,

One thing I avoided working with for a long time is the Asterisk 
voicemail code. One module in
Asterisk I've constantly been naming as one of the worst parts is 
voicemail. One part of

Asterisk that I've been kind of avoiding during my trainings is voicemail.


...

I've wanted to do exactly this for quite some time, but I haven't had 
the bandwidth to be able to do any work on it. This idea makes perfect 
sense. I've always felt that the current voicemail implementation was 
too restricted and didn't allow enough customization.


Anyway, I don't have any time to help or test right now (perhaps I will 
in a couple of months), but I just wanted to at least thank you for 
doing what you have done already, and encourage you and others to 
continue this work.


Thanks!

John
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Re: [asterisk-users] capi installation problem

[Armin Schindler]

I think you are mixing something here. The FritzCard is not a "B1", so you 
don't need the b1 modules, the firmware and the /etc/capi.conf.

You can either use the FritzCard driver (binary modules from AVM), or you 
use mISDN (which is also already loaded according to your lsmod).

When using mISDN, you can either use the mISDN-CAPI to really provide a CAPI
interface, or just don't use CAPI and use chan_misdn instead.

Armin

On Wed, 7 Mar 2007, Giedrius Augys wrote:
> Hello,
> I have problem with capi, I can't install it. I have putted all info what I
> did and what I get :). I think you can suggest me how to solve this
> problem.and thanking you in anticipation. I have ISDN Frtiz!Card PCI
> v2.1and I want  to install it to my ubuntu box (kernel:
> 2.6.17-10-server). Using command "lspci -vv" , I can see that kernel finds
> this card:
> *02:0d.0 Network controller: AVM Audiovisuelles MKTG & Computer System GmbH
> Fritz!PCI v2.0 ISDN (rev 02)
>Subsystem: AVM Audiovisuelles MKTG & Computer System GmbH Fritz!PCI
> v2.0 ISDN
>Control: I/O+ Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr-
> Stepping- SERR+ FastB2B-
>Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort-
> SERR-  Interrupt: pin A routed to IRQ 201
> Region 0: Memory at ff9fec00 (32-bit, non-prefetchable) [size=32]
> Region 1: I/O ports at df80 [size=32]
> Capabilities: [40] Power Management version 2
>Flags: PMEClk- DSI- D1- D2+ AuxCurrent=55mA
> PME(D0-,D1-,D2+,D3hot+,D3cold+)
> Status: D0 PME-Enable- DSel=0 DScale=0 PME-*
> 
> 
> 
>  I found tutorial in http://www.asteriskguru.com/tutorials/avm_b1.html .I
> have installed these packages:
> *ii  capiutils  3.9.20060704-1  Utilities for
> CAPI-capable ISDN cards
> ii  libcapi20-33.9.20060704-1  libraries for
> CAPI support
> ii  libcapi20-dev  3.9.20060704-1  libraries for
> CAPI support
> ii  avm-fritz-firmware-2.6.17-10   3.11+2.6.17.7-10.1  Firmware for AVM
> Fritz! ISDN hardware*
> 
> and downloaded firmaware from *
> ftp://ftp.in-berlin.de/pub/capi4linux/firmware/b1/ .*My capi.conf is:
> *b1pci   /usr/share/isdn/b1.t4   DSS1-   -
> -   -   P2P*
> 
> Then I exec command capiinit start, I have noting on output, but it load
> modules:
> [EMAIL PROTECTED]:~# lsmod
> Module  Size  Used by
> b1pci  10624  0
> b1dma  17412  1 b1pci
> b1 25856  2 b1pci,b1dma
> capi   19392  0
> kernelcapi 49664  4 b1pci,b1dma,b1,capi
> capifs  7176  2 capi
> ipv6  271136  12
> lp 12964  0
> mISDN_l2   44288  0
> mISDN_l1   13192  0
> avmfritz   25740  0
> mISDN_isac 17280  1 avmfritz
> mISDN_core 75648  4 mISDN_l2,mISDN_l1,avmfritz,mISDN_isac*
> 
> But when I execute command cappinfo, I get :
> [EMAIL PROTECTED]:~# capiinfo
> capi not installed - No such device or address (6).
> *
> 
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[asterisk-users] VoIP over Alvarion Wireless

[Matt]

Hi,
This question isn't specifically asterisk related, but perhaps someone here
can shed some light or offer some insight.
Is anyone else here running VoIP over Alvarion wireless?If yes, do you
have any suggestions for what you've done to make it "work"?   It seems that
no amount of traffic shaping, checking installs for error rates, lowering
error rates, or setting contention windows makes VoIP work. It just
plain blows over Alvarion wireless gear.   Cable and DSL?  Yup.. it works
great.Wireless?  Forget it, pickup your cell phone.

Anyone have any better news, or suggestions?
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[asterisk-users] Background / Invalid Extension through cell phone

[John Congdon]

Hi,

I have had an issue for a long time, and really just can't solve it.

My boss and others seem to have a problem when they call into our 
asterisk phone system.  It often takes 3-4 tries of entering an 
extension before the system gets it right.


Below is my context that the call comes into, and some debugging from 
the asterisk console.


[corporate]
;;exten => fax,1,Macro(faxreceive)
#include "local_ext.conf"

exten => s, 1, answer
exten => s, 2, Wait(2)
exten => s, 3, Background(menu_ty)
exten => s, 4, Background(menu_ext)
exten => s, 5, WaitExten(30)
exten => s, 6, Voicemail(u100)
exten => s, 7, Hangup()

exten => t, 1, Hangup
exten => i, 1, goto(s|4)


Below, you will see the call come into Zap/53.  He is trying to dial 
extension 104.  The first few times, only the 4 is recognized.  Then 
14.  But if you look at the cellphone screen, you see 104104104104104104.


   -- Executing BackGround("Zap/53-1", "menu_ext") in new stack
   -- Playing 'menu_ext' (language 'en')
   -- Invalid extension '4' in context 'corporate' on Zap/53-1
 == CDR updated on Zap/53-1
   -- Executing Goto("Zap/53-1", "s|4") in new stack
   -- Goto (corporate,s,4)
   -- Executing BackGround("Zap/53-1", "menu_ext") in new stack
   -- Playing 'menu_ext' (language 'en')
   -- Invalid extension '4' in context 'corporate' on Zap/53-1
 == CDR updated on Zap/53-1
   -- Executing Goto("Zap/53-1", "s|4") in new stack
   -- Goto (corporate,s,4)
   -- Executing BackGround("Zap/53-1", "menu_ext") in new stack
   -- Playing 'menu_ext' (language 'en')
   -- Playing 'vm-intro' (language 'en')
pbx*CLI>
   -- Invalid extension '4' in context 'corporate' on Zap/53-1
 == CDR updated on Zap/53-1
   -- Executing Goto("Zap/53-1", "s|4") in new stack
   -- Goto (corporate,s,4)
   -- Executing BackGround("Zap/53-1", "menu_ext") in new stack
   -- Playing 'menu_ext' (language 'en')
pbx*CLI>
   -- Invalid extension '14' in context 'corporate' on Zap/53-1
 == CDR updated on Zap/53-1
   -- Executing Goto("Zap/53-1", "s|4") in new stack
   -- Goto (corporate,s,4)
   -- Executing BackGround("Zap/53-1", "menu_ext") in new stack
   -- Playing 'menu_ext' (language 'en')
pbx*CLI>
pbx*CLI>
 == Spawn extension (ebay, s, 2) exited non-zero on 'Zap/56-1'
   -- Hungup 'Zap/56-1'
pbx*CLI>
 == CDR updated on Zap/53-1
   -- Executing Macro("Zap/53-1", "stdexten|104|IAX2/congdonj") in new 
stack

   -- Executing Dial("Zap/53-1", "IAX2/congdonj|20|tw") in new stack

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Re: [asterisk-users] Background / Invalid Extension through cell phone

[Doug Lytle]

John Congdon wrote:

Hi,

I have had an issue for a long time, and really just can't solve it.

My boss and others seem to have a problem when they call into our 
asterisk phone system.  It often takes 3-4 tries of entering an 
extension before the system gets it right.


Below is my context that the call comes into, and some debugging from 
the asterisk console.



You may want to add the following to the zapata.conf

; If you are having trouble with DTMF detection, you can relax the DTMF
; detection parameters.  Relaxing them may make the DTMF detector more 
likely

; to have "talkoff" where DTMF is detected when it shouldn't be.
;
;relaxdtmf=yes

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] VoIP over Alvarion Wireless

[Gordon Henderson]

On Wed, 7 Mar 2007, Matt wrote:


Hi,
This question isn't specifically asterisk related, but perhaps someone here
can shed some light or offer some insight.
Is anyone else here running VoIP over Alvarion wireless?If yes, do you
have any suggestions for what you've done to make it "work"?   It seems that
no amount of traffic shaping, checking installs for error rates, lowering
error rates, or setting contention windows makes VoIP work. It just
plain blows over Alvarion wireless gear.   Cable and DSL?  Yup.. it works
great.Wireless?  Forget it, pickup your cell phone.


Many many moons ago I was involved with community WiFi broadband systems 
and we used a variety of wireless systems for "backhaul" I think we 
trialled Alvarion but never used it - we did use Orthogon, WiLAN and 
Apperto kit though.


The biggest issues we had (with both the consumer facing WiFi and the more 
industrial "backhaul" kit) at the time were to do with the packet size and 
the units ability to cope with small packets. At the nuts & bolts level, 
all the low-end stuff is really half-duplex, so there is a link 
turn-around-time, and if that exceeds the packet send time, then you're in 
trouble - VoIP typically requires a full duplex link with small packets 
going in both directions - you can emulate full duplex with a half duplex 
link if you can do the turn-around quick enough and/or accept a lower 
overall data rate, higher latency, etc., but I'm guessing here that the 
Alvarion kit can't cope - they are typically optimised to stream large 
quantities of data in full sized MTU packets back to back. (so streaming 
video and audio seems great, but interactive audio is less-so).


Increasing the packet size at the end points may help, if possible - but 
I'm not sure where to start - Grandstream phones have a "Voice Frames per 
TX:" parameter, but I've no idea whether that (or it's equivalent 
elsewhere) would help, or just make things like latency worse...



Things may have changed in the 3 years since I was actively involved in 
this though, but we spent a lot of time playing with iperf and various 
makes of links and so on. (and daisy-chaining links back to back, which we 
found was a really bad thing to do with the 802.11b kit we were using - 
each hop would halve the throughput )-: So one kiddy running a video phone 
application killed everything else, or another kiddy doing huge uploads 
would kill it for people trying to do downloads. Traffic shaping helped, 
but it really needs to be done at the access point level, and not 2-3 hops 
further up the network where our routers were typically positioned.


If you want good PtP backhaul kit, then get Motorola Canopy units, but I 
used to work for them (before the company was bought by motorola) so I may 
be biased ;-)


Gordon
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Re: [asterisk-users] Background / Invalid Extension through cell phone

[Eric \"ManxPower\" Wieling]

Doug Lytle wrote:

John Congdon wrote:

Hi,

I have had an issue for a long time, and really just can't solve it.

My boss and others seem to have a problem when they call into our 
asterisk phone system.  It often takes 3-4 tries of entering an 
extension before the system gets it right.


Below is my context that the call comes into, and some debugging from 
the asterisk console.



You may want to add the following to the zapata.conf

; If you are having trouble with DTMF detection, you can relax the DTMF
; detection parameters.  Relaxing them may make the DTMF detector more 
likely

; to have "talkoff" where DTMF is detected when it shouldn't be.
;
;relaxdtmf=yes


I have found that relaxdtmf=yes has caused more problems than it fixes. 
 In my experience problems with detecting DTMF on an FXO port can 
usually be fixed by playing with rxgain and txgain.

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Re: [asterisk-users] Back to back E1 - asterisk <=> toshiba pbx - Call droping issue

[Eric \"ManxPower\" Wieling]

Vidura Senadeera wrote:

Hi Team,

I have integrated asterisk with Toshiba analog PBX. NOw the live setup is
going.

Now I am facing call droping problem. It's happening ample time. 10-20 
calls

are droping every day.

What could be the reason. I attached latest zapata.conf file for your
information.



This is being a huge issue.

Highly appreciate your help on this regard.


I didn't see your /etc/asterisk/zapata.conf.  If you have 
callprogress=yes then remove it.  If you have busydetect=yes then add 
busycount=8.  Look at the output of "cat /proc/interrupts".  If you see 
any of the Digium cards sharing an IRQ, you need to make the Digium 
cards not share IRQs.  This can be done in the BIOS on some systems, on 
other systems you need to move the boards to a different PCI slot.


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RE: [asterisk-users] TC400B

[Wai Wu]

Anyone? 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wai Wu
Sent: Monday, March 05, 2007 10:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] TC400B

 
Anyone tried the digium TC400B transcoding card? What are your opinions?

Thnx
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Re: [asterisk-users] Background / Invalid Extension through cell phone

[Steve Totaro]

Eric "ManxPower" Wieling wrote:

Doug Lytle wrote:

John Congdon wrote:

Hi,

I have had an issue for a long time, and really just can't solve it.

My boss and others seem to have a problem when they call into our 
asterisk phone system.  It often takes 3-4 tries of entering an 
extension before the system gets it right.


Below is my context that the call comes into, and some debugging 
from the asterisk console.



You may want to add the following to the zapata.conf

; If you are having trouble with DTMF detection, you can relax the DTMF
; detection parameters.  Relaxing them may make the DTMF detector 
more likely

; to have "talkoff" where DTMF is detected when it shouldn't be.
;
;relaxdtmf=yes


I have found that relaxdtmf=yes has caused more problems than it 
fixes.  In my experience problems with detecting DTMF on an FXO port 
can usually be fixed by playing with rxgain and txgain.


What sort of problems have you seen it cause?  I guess I could see 
hitting the wrong extension in rare cases.  Anyways, relaxdtmf has 
worked wonders for me over T1s and analog lines (always seems to be cell 
phones that have issues, probably because of the GSM and radio distorts 
beyond the specs).


Thanks,
Steve

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[asterisk-users] Realtime Extensions and "Include"

[Peder @ NetworkOblivion]

Is it possible to use the "include" command to include other contexts if 
you are using realtime for extensions?  I've searched voip-info and some 
people were asking about it, but I didn't find a real answer anywhere.


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Re: [asterisk-users] Background / Invalid Extension through cell phone

[Eric \"ManxPower\" Wieling]

Steve Totaro wrote:

Eric "ManxPower" Wieling wrote:

Doug Lytle wrote:

John Congdon wrote:

Hi,

I have had an issue for a long time, and really just can't solve it.

My boss and others seem to have a problem when they call into our 
asterisk phone system.  It often takes 3-4 tries of entering an 
extension before the system gets it right.


Below is my context that the call comes into, and some debugging 
from the asterisk console.



You may want to add the following to the zapata.conf

; If you are having trouble with DTMF detection, you can relax the DTMF
; detection parameters.  Relaxing them may make the DTMF detector 
more likely

; to have "talkoff" where DTMF is detected when it shouldn't be.
;
;relaxdtmf=yes


I have found that relaxdtmf=yes has caused more problems than it 
fixes.  In my experience problems with detecting DTMF on an FXO port 
can usually be fixed by playing with rxgain and txgain.


What sort of problems have you seen it cause?  I guess I could see 
hitting the wrong extension in rare cases.  Anyways, relaxdtmf has 
worked wonders for me over T1s and analog lines (always seems to be cell 
phones that have issues, probably because of the GSM and radio distorts 
beyond the specs).


It caused asterisk to see a single digit when two of the same digits 
were dialed in a row.  So a user dialed 4415 and Asterisk saw 415. 
Remember that on all cell phones (except the analog ones) DTMF from the 
phone is sent out of band and so should not be distorted.

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[asterisk-users] asterisk and ssl

[nik600]

what is the support in asterisk for ssl voip protocols?

I am looking for a solutions to grant the possibility to some users to
use an asterisk server as a proxy voice, for talking each them in a
safe and secure mode on internet.

Is it possible?

thanks
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Re: [asterisk-users] Realtime Extensions and "Include"

[Rob Schall]

Not sure if this is what you mean But we have includes in our
sip,extensions and voicemail files.
;#include sip.inc

We keep them commented out only because they are a copy of what is
running in realtime. Every night the include files are generated and put
in /etc/asterisk. If MySQL were to ever fail, we could just uncomment
those 3 includes and reload asterisk. It wouldn't have the same dynamic
nature to it, but it would bring functionality back online.

Rob


Peder @ NetworkOblivion wrote:
> Is it possible to use the "include" command to include other contexts
> if you are using realtime for extensions?  I've searched voip-info and
> some people were asking about it, but I didn't find a real answer
> anywhere.
>
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[asterisk-users] OT Vonage V-Phone Adapter (Possible Hack)

[Steve Totaro]

It would be cool to get one of these and see if it can be hacked and
loaded with your favorite SIP or IAX softphone.  Looking at the pic, it
looks like the dongle is both a soundcard and memory stick.  Heck, I
would be glad to have it if I could get the soundcard to work.

Might as well since it is free after rebate.  

http://www.circuitcity.com/ssm/Accessories-for-Vonage-V-Phone-VPHONE/sem
/rpsm/oid/162059/rpem/ccd/productDetailAccessory.do#tabs


Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
 


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RE: [asterisk-users] asterisk and ssl

[Dean Collins]

Hi Nik,
Do some googling on Phil Zimmermanns Zphone.

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of nik600
> Sent: Wednesday, 7 March 2007 9:38 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] asterisk and ssl
> 
> what is the support in asterisk for ssl voip protocols?
> 
> I am looking for a solutions to grant the possibility to some users to
> use an asterisk server as a proxy voice, for talking each them in a
> safe and secure mode on internet.
> 
> Is it possible?
> 
> thanks
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RE: [asterisk-users] Background / Invalid Extension through cell phone

[Steve Totaro]

> >> I have found that relaxdtmf=yes has caused more problems than it
> >> fixes.  In my experience problems with detecting DTMF on an FXO
port
> >> can usually be fixed by playing with rxgain and txgain.
> >>
> > What sort of problems have you seen it cause?  I guess I could see
> > hitting the wrong extension in rare cases.  Anyways, relaxdtmf has
> > worked wonders for me over T1s and analog lines (always seems to be
cell
> > phones that have issues, probably because of the GSM and radio
distorts
> > beyond the specs).
> 
> It caused asterisk to see a single digit when two of the same digits
> were dialed in a row.  So a user dialed 4415 and Asterisk saw 415.
> Remember that on all cell phones (except the analog ones) DTMF from
the
> phone is sent out of band and so should not be distorted.

Good points.  So I wonder why callers with DTMF issues are usually cell
callers.  The only person I know with an analog phone anymore is my
mother :-)

Thanks,
Steve

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Re: [asterisk-users] asterisk and ssl

[Olle E Johansson]

7 mar 2007 kl. 15.38 skrev nik600:


what is the support in asterisk for ssl voip protocols?

I am looking for a solutions to grant the possibility to some users to
use an asterisk server as a proxy voice, for talking each them in a
safe and secure mode on internet.

Is it possible?


No.

/O
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RE: [asterisk-users] OT Vonage V-Phone Adapter (Possible Hack)

[Steve Totaro]

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Steve Totaro
> Sent: Wednesday, March 07, 2007 9:58 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] OT Vonage V-Phone Adapter (Possible Hack)
> 
> It would be cool to get one of these and see if it can be hacked and
> loaded with your favorite SIP or IAX softphone.  Looking at the pic,
it
> looks like the dongle is both a soundcard and memory stick.  Heck, I
> would be glad to have it if I could get the soundcard to work.
> 
> Might as well since it is free after rebate.
> 
>
http://www.circuitcity.com/ssm/Accessories-for-Vonage-V-Phone-VPHONE/sem
> /rpsm/oid/162059/rpem/ccd/productDetailAccessory.do#tabs
> 
> 
> Thanks,
> Steve Totaro
> http://www.asteriskhelpdesk.com
> KB3OPB

Hate to reply to my own thread, but the rebate is only available with
new activations.  I am still in for one to play with.

Thanks,
Steve

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RE: [asterisk-users] asterisk and ssl

[Steve Totaro]

If you just want to secure between Asterisk servers and clients that you
control, OpenVPN rules in simplicity and transparency.

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
 

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Dean Collins
> Sent: Wednesday, March 07, 2007 9:56 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] asterisk and ssl
> 
> Hi Nik,
> Do some googling on Phil Zimmermanns Zphone.
> 
> 
> 
> Regards,
> 
> Dean Collins
> Cognation Pty Ltd
> [EMAIL PROTECTED]
> +1-212-203-4357 Ph
> 
> 
> > -Original Message-
> > From: [EMAIL PROTECTED]
[mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of nik600
> > Sent: Wednesday, 7 March 2007 9:38 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: [asterisk-users] asterisk and ssl
> >
> > what is the support in asterisk for ssl voip protocols?
> >
> > I am looking for a solutions to grant the possibility to some users
to
> > use an asterisk server as a proxy voice, for talking each them in a
> > safe and secure mode on internet.
> >
> > Is it possible?
> >
> > thanks
> > ___
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> >http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] OT Vonage V-Phone Adapter (Possible Hack)

[cb]

On Mar 7, 2007, at 9:58 AM, Steve Totaro wrote:


Might as well since it is free after rebate.


Just as a heads up, that rebate, like most of the others for Vonage  
based items, requires Vonage activation in order to actually get the  
rebate.


-chris



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Re: [asterisk-users] Background / Invalid Extension through cell phone

[Jorge Mendoza]

Eric

Eric "ManxPower" Wieling wrote:

Steve Totaro wrote:

Eric "ManxPower" Wieling wrote:

Doug Lytle wrote:

John Congdon wrote:

Hi,

I have had an issue for a long time, and really just can't solve it.

My boss and others seem to have a problem when they call into our 
asterisk phone system.  It often takes 3-4 tries of entering an 
extension before the system gets it right.


Below is my context that the call comes into, and some debugging 
from the asterisk console.



You may want to add the following to the zapata.conf

; If you are having trouble with DTMF detection, you can relax the 
DTMF
; detection parameters.  Relaxing them may make the DTMF detector 
more likely

; to have "talkoff" where DTMF is detected when it shouldn't be.
;
;relaxdtmf=yes


I have found that relaxdtmf=yes has caused more problems than it 
fixes.  In my experience problems with detecting DTMF on an FXO port 
can usually be fixed by playing with rxgain and txgain.


What sort of problems have you seen it cause?  I guess I could see 
hitting the wrong extension in rare cases.  Anyways, relaxdtmf has 
worked wonders for me over T1s and analog lines (always seems to be 
cell phones that have issues, probably because of the GSM and radio 
distorts beyond the specs).


It caused asterisk to see a single digit when two of the same digits 
were dialed in a row.  So a user dialed 4415 and Asterisk saw 415. 
Remember that on all cell phones (except the analog ones) DTMF from 
the phone is sent out of band and so should not be distorted.


Are you sure about that?. I think that the DTMF digits are send out of 
band before the answer supervision. After that, the DTMF digits are send 
in band if they are dialled from the keypad. When I call my IVR, the 
system answer and I dial other DTMF digits, only around 20% of calls 
succeed. However if I store the DTMF sequence in the cell phone (digits, 
pause, send, digits, etc.) 100% of calls succeed.


Jorge Mendoza
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[asterisk-users] queue information in mySQL

[Thomas Winter]

Hi,

is it possible to have the information stored in 

/var/log/asterisk/queue_log

realtime in mySQL?

thanks

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[asterisk-users] Asterisk queue and agents

[Hall, Eric M.]

Have a question for the group

 

If I have an agent is on the phone outside of the queue should that
person still get queue calls ?

 

Doing a show agents online I see Available however show hints I see
inuse.

 

Any ideas

 

 

 

Eric Hall
Vice-president
Amaxx, Inc.
"Customized IT Solutions"
5925B Wilcox Place
Dublin OH 43016
614.923.6652 - Direct
614.486.3481 - Office
614.923.6652 - eFax

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you are hereby notified that any dissemination, distribution or copying
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[asterisk-users] Asterisk Auto-dial out

[Phil Menico]

I am using the * auto-dial out feature but don't want to have to specify
a channel (Zap/G2/) to connect to the extension.

Current file I use:

Channel: Zap/G2/12127778866   #<<  I have to specify a specific
channel 
MaxRetries: 1
RetryTime: 60
WaitTime: 30
#
# Assuming that your outgoing call logic is kept in the
#  context called [line1out]
#
Context: line1out
Extension: 7632
Priority: 1

Is there a way that I can just put in the number and have the system
decide the channel to use for calling it?

What I would like to do:

Channel:   #<<=== This number could be 
   #  7645 in which case go via SIP/7645 
   #  68001 which should go to CiscoSIP/68001
   #  12127778866 which would go via
Zap/G2/12127778866
MaxRetries: 1
RetryTime: 60
WaitTime: 30
#
# Assuming that your outgoing call logic is kept in the
#  context called [line1out]
#
Context: line1out
Extension: 7632
Priority: 1

Based on dialing plan the system should be able to route the call to
whatever channel supports dialing that number.

Thank you.

Phil Menico 

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[asterisk-users] AsteriskNow Beta 4 - zaptel.conf / zapata.conf problems

[mark.d.rowe]

I have recently installed Asterisk Now Beta 4 (32-bit) with three Digium
cards - 2 Wildcards TE400P (one with 4 FXS modules and one with 4 FXO
modules) and a TE205P dual E1/T1 card. The problem is with the TE205P
and configuring the /etc/zaptel.conf and /etc/asterisk/zapata.conf
files. I go in and add the required lines to configure the TE205P and
restart Asterisk. 

All is fine until I reboot the server. When the server reboots all
changes to the zaptel.conf and zapata.conf files have disappeared and
the files are back in their original state. I've tried installing
AsteriskNow again from scratch but the same problem keeps occurring. Can
anyone suggest a reason for this and how it can be overcome?

Regards,

Mark.
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Re: [asterisk-users] queue information in mySQL

[Philipp Kempgen]

Thomas Winter wrote:

> is it possible to have the information stored in 
> 
> /var/log/asterisk/queue_log
> 
> realtime in mySQL?

No. You need to write a custom script.
See
http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL

William Lloyd has a Perl script for that:
http://lists.digium.com/pipermail/asterisk-users/2005-July/109892.html

And QueueMetrics (free demo) comes with a Perl script as well:
http://queuemetrics.loway.it/

But there's one big disadvantage of replacing the queue_log file
by a FIFO: should your script ever fail to read from the pipe
Asterisk will stop working as the FIFO is blocking.

So you might want to use logrotate which has an option to truncate
the file in place. Disadvantage: there's a very small chance that
some log entries get lost during the rotation although that is not
very likely.

Regards,
  Philipp

-- 
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] AsteriskNow Beta 4 - zaptel.conf / zapata.conf problems

[demuel]

U have edited the wrong files. Can u tell us here the zaptel and the zapata 
files being used by
asteriskNow 1.4-beta?

>
> I have recently installed Asterisk Now Beta 4 (32-bit) with three Digium
> cards - 2 Wildcards TE400P (one with 4 FXS modules and one with 4 FXO
> modules) and a TE205P dual E1/T1 card. The problem is with the TE205P
> and configuring the /etc/zaptel.conf and /etc/asterisk/zapata.conf
> files. I go in and add the required lines to configure the TE205P and
> restart Asterisk.
>
> All is fine until I reboot the server. When the server reboots all
> changes to the zaptel.conf and zapata.conf files have disappeared and
> the files are back in their original state. I've tried installing
> AsteriskNow again from scratch but the same problem keeps occurring. Can
> anyone suggest a reason for this and how it can be overcome?
>
> Regards,
>
> Mark.
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Re: [asterisk-users] Realtime Extensions and "Include"

[Peder @ NetworkOblivion]

Not really what I mean.  I have customer contexts, say customer1 and 
customer2.  I also have a LD, Local and Intl context.  To allow 
customer1 to dial LD, I include the LD context within the customer1 
context.  I want to skip text files and move to realtime for extensions 
and I want to know if I can include other contexts in the realtime 
mechanism like I do with the text files.


Rob Schall wrote:

Not sure if this is what you mean But we have includes in our
sip,extensions and voicemail files.
;#include sip.inc

We keep them commented out only because they are a copy of what is
running in realtime. Every night the include files are generated and put
in /etc/asterisk. If MySQL were to ever fail, we could just uncomment
those 3 includes and reload asterisk. It wouldn't have the same dynamic
nature to it, but it would bring functionality back online.

Rob


Peder @ NetworkOblivion wrote:

Is it possible to use the "include" command to include other contexts
if you are using realtime for extensions?  I've searched voip-info and
some people were asking about it, but I didn't find a real answer
anywhere.

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--

Network stuff you didn't know
http://www.networkoblivion.com

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RE: [asterisk-users] queue information in mySQL

[Steve Totaro]

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Philipp Kempgen
> Sent: Wednesday, March 07, 2007 11:08 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] queue information in mySQL
> 
> Thomas Winter wrote:
> 
> > is it possible to have the information stored in
> >
> > /var/log/asterisk/queue_log
> >
> > realtime in mySQL?
> 
> No. You need to write a custom script.
> See
> http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL
> 
> William Lloyd has a Perl script for that:
> http://lists.digium.com/pipermail/asterisk-users/2005-July/109892.html
> 
> And QueueMetrics (free demo) comes with a Perl script as well:
> http://queuemetrics.loway.it/
> 
> But there's one big disadvantage of replacing the queue_log file
> by a FIFO: should your script ever fail to read from the pipe
> Asterisk will stop working as the FIFO is blocking.
> 
> So you might want to use logrotate which has an option to truncate
> the file in place. Disadvantage: there's a very small chance that
> some log entries get lost during the rotation although that is not
> very likely.
> 
> Regards,
>   Philipp
> 
> --
> 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
> 

Queuemetrics has two different scripts that work great.  One will parse
your log files and input the data into MySQL, the second one just tails
the file and puts the data into the MySQL DB on the fly as it is written
to queue_log.  Neither one will modify the way your original queue_log
files are written or operate.

Thanks,
Steve

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[asterisk-users] mobility with asterisk

[Alvaro Pacho]

Hello,

I´m working testing every feature of asterisk in a lab.  Now I am very
interested in asterisk over network mobility environment. For example : when
somebody is talking with his ip-phone ) and moving around a big enterprise,
needing to change the ip-address (other AP) would it be possible in the
minimum time to avoid loosing quality in the current call? I read this test
http://lists.digium.com/pipermail/asterisk-dev/2006-December/025263.html
but it´was written in December of 2006!! Were this ideas implemented? If you
can help me with information about that please write me and I´ll test and
give you my end result.
Does anybody knows something about which is the best Cisco router to this
mobility environment?

Best regards,

  Pacho
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[asterisk-users] Problem HandyTone 488 does not call transfer

[Pablo Almido]

Hi

I have a analog phone connected to my Gateway Handytone and registered to
Asterisk 1.4 I have configured my HandyTone 488
(in the section FXS Port) for make and receive calls, however I can
not transfer a call when it come via PSTN. But, when a call come from via IP
I can transfer it.


[phoneanalog]

type=friend
secret=XXX
context=local
nat=no
qualify=yes
host=dynamic
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=gsm
allow=alaw
callerid=Krix Altrust <2041>
[EMAIL PROTECTED]
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Re: [asterisk-users] mobility with asterisk

[Alex Robar]

Hi Alvaro,

There was a discussion about this a little while ago. Andrew Joakimsen had
some good ideas with how to allow a wifi sip phone to roam between APs
seemlessly. His post pretty much said the following:

 - Set all APs to the same channel and same SSID.
 - Make sure all APs are connected to the same LAN (no NAT on the AP).
 - Play with the settings on the phone with regards to roaming deltas and
receive levels. Suggested settings:
- RxLevel: -60
- PreRoaming: Enable
- RxLevel: -75
- Try Over TxErrcnt: 15
- Try Over RxErrorcnt: 10

Playing with the pre-roaming settings will help you, but you may see a drop
in battery life.

Cheers,
Alex

On 3/7/07, Alvaro Pacho <[EMAIL PROTECTED]> wrote:


Hello,

I´m working testing every feature of asterisk in a lab.  Now I am very
interested in asterisk over network mobility environment. For example : when
somebody is talking with his ip-phone ) and moving around a big enterprise,
needing to change the ip-address (other AP) would it be possible in the
minimum time to avoid loosing quality in the current call? I read this test
http://lists.digium.com/pipermail/asterisk-dev/2006-December/025263.html
but it´was written in December of 2006!! Were this ideas implemented? If you
can help me with information about that please write me and I´ll test and
give you my end result.
Does anybody knows something about which is the best Cisco router to this
mobility environment?

Best regards,

   Pacho

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--
Alex Robar
[EMAIL PROTECTED]
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Re: [asterisk-users] Asterisk queue and agents

[Octavio Ruiz (Ta^3)]

>Have a question for the group
>If I have an agent is on the phone outside of the queue should that person
>still get queue calls ?
>Doing a show agents online I see Available however show hints I see inuse.

There is a ringinuse feature for SIP devices on 1.4.X which is what you are 
looking for.

-- 
Octavio Ruiz Cervera
Neocenter, SA. de CV.
http://www.neocenter.com/
Soluciones para Centros de Contacto y Telefonía IP
Tel.: (+52 55) 8590-9000 Ext. 9016
Cel.: (+55 55) 5514-087790
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Re: [asterisk-users] VoIP over Alvarion Wireless

[Andrew Niemantsverdriet]

What Alvarion stuff are you using? Alvarion is the top rated wireless
manufactor when it comes to VOIP on there new stuff (VL 4.0 series and
B100 backhaul series). Some of there older stuff like the BrezeeAccess
FHSS will not work well because the QoS that the radio provides is
spotty at best.

On 3/7/07, Matt <[EMAIL PROTECTED]> wrote:

Hi,
This question isn't specifically asterisk related, but perhaps someone here
can shed some light or offer some insight.
Is anyone else here running VoIP over Alvarion wireless?If yes, do you
have any suggestions for what you've done to make it "work"?   It seems that
no amount of traffic shaping, checking installs for error rates, lowering
error rates, or setting contention windows makes VoIP work. It just
plain blows over Alvarion wireless gear.   Cable and DSL?  Yup.. it works
great.Wireless?  Forget it, pickup your cell phone.

Anyone have any better news, or suggestions?

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Re: [asterisk-users] mobility with asterisk

[Jorge Mendoza]

Take a look at Alberto Pastor's mail on this list at 2007/02/15: moving 
WiFi phone.


+jm

Alvaro Pacho wrote:

Hello,

I´m working testing every feature of asterisk in a lab.  Now I am very 
interested in asterisk over network mobility environment. For example 
: when somebody is talking with his ip-phone ) and moving around a big 
enterprise, needing to change the ip-address (other AP) would it be 
possible in the minimum time to avoid loosing quality in the current 
call? I read this test 
http://lists.digium.com/pipermail/asterisk-dev/2006-December/025263.html  
but it´was written in December of 2006!! Were this ideas implemented? 
If you can help me with information about that please write me and 
I´ll test and give you my end result.
Does anybody knows something about which is the best Cisco router to 
this mobility environment?


Best regards,

   Pacho


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RE: [asterisk-users] gtalk2voip and Asteris

[DID 4ME]

What kinds of problems were you having? I'm on 1.4.0 and chan_gtalk.so
simply doesn't load. Of the 146 files in the /usr/lib/asterisk/modules/
directory, asterisk loads 144 of them, omitting only chan_gtalk.so and
res_jabber.so.


Connected to Asterisk 1.4.1 currently running on monkey (pid = 9371)
Verbosity is at least 3
foo*CLI> module load chan_gtalk.so
[Mar  7 10:23:07] WARNING[9407]: loader.c:362 load_dynamic_module: Error
loading module 'chan_gtalk.so': libiksemel.so.3: cannot open shared object
file: No such file or directory
[Mar  7 10:23:07] WARNING[9407]: loader.c:614 load_resource: Module
'chan_gtalk.so' could not be loaded.
foo*CLI>

[EMAIL PROTECTED]:/etc/asterisk# ls /usr/lib/asterisk/modules/ | grep gtalk
chan_gtalk.so
[EMAIL PROTECTED]:/etc/asterisk#

[EMAIL PROTECTED]:/etc/asterisk# ls /etc/asterisk/ | grep gtalk
gtalk.conf
gtalk.conf.old
[EMAIL PROTECTED]:/etc/asterisk#

[EMAIL PROTECTED]:/etc/asterisk# ls /usr/local/lib/ | grep libiks
libiksemel.a
libiksemel.la
libiksemel.so
libiksemel.so.3
libiksemel.so.3.0.0
[EMAIL PROTECTED]:/etc/asterisk#

Is it possible I am having an issue with libiksemel?  I can't figure out
what the problem can be with Asterisk.

-Nick

-Original Message-

From: [EMAIL PROTECTED] [*
mailto:[EMAIL PROTECTED]<[EMAIL PROTECTED]>]
On Behalf Of Klaverstyn, David C

Sent: Tuesday, March 06, 2007 8:44 PM

To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: RE: [asterisk-users] gtalk2voip and Asterisk

After upgrading to Asterisk 1.4.1 from 1.4.0 it just worked for me.

There must have been a bug in 1.4.0. I have successfully connected to a
Gmail and MSN instant message client.

-Original Message-

From: [EMAIL PROTECTED]

[*mailto:[EMAIL PROTECTED]<[EMAIL PROTECTED]>]
On Behalf Of Mani Sridhar

Sent: Saturday, 3 March 2007 8:44 PM

To: asterisk-users@lists.digium.com

Subject: Re: [asterisk-users] gtalk2voip and Asterisk

hi,

i was able to get this working with google talk.

i entered [EMAIL PROTECTED] using the gtalk2voip.com website's "invite"

box, and as a result, saw a request from [EMAIL PROTECTED] to be added
as a buddy in my google talk contact list. i accepted the request.

in my asterisk dialplan, i have this entry...

exten => 3501, 1, Dial(SIP/[EMAIL PROTECTED])

this allows any extension in my asterisk box to dial 3501, and
[EMAIL PROTECTED] receives the call on the google talk client. the call
is established, and voice quality is good.

to allow a call from google talk client for [EMAIL PROTECTED], i opened a
chat window to the buddy [EMAIL PROTECTED] and typed "call
[EMAIL PROTECTED]", and this made extension ABC on my asterisk box start
ringing. again, the call was established, and audio was ok.

as far as asterisk is concerned, this is a SIP call. bottomline - it's a

good alternative to using the native jabber/jingle library in asterisk

1.4 .

in fact, i haven't been able to get asterisk to successfully set up a call
to googletalk using the chan_gtalk module . i am inside a NAT-ed LAN, and
audio works in one direction only for the asterisk (SIP) - gtalk call.

anyone else got asterisk - googletalk using chan_gtalk working?






Message: 10



Date: Fri, 02 Mar 2007 19:07:41 +0200



From: Cosmin Prund <[EMAIL PROTECTED]>



Subject: Re: [asterisk-users] gtalktovoip and Asteirsk



To: Asterisk Users Mailing List - Non-Commercial Discussion







Message-ID: <[EMAIL PROTECTED]>



Content-Type: text/plain; charset="iso-8859-1"







I don't think it works. I tried calling my own yahoo messenger ID with



no success: it rings a number of times and then it goes to some sort of



voice mail.



And I did "invite" the user they specified to my yahoo list, I also



entered my yahoo id into the registration form on the site.



I used a extensions.conf command like this for the try:







exten => 641,1,Dial(SIP/[EMAIL PROTECTED])







(and yes, that's one of the yahoo ID I tryed with, and I don't think it



exists! )







Klaverstyn, David C wrote:



>



> Has anyone managed to get gtalktovoip working at all? If so please



> explain.



>



>



>



> *http://www.gtalk2voip.com/faq.shtml*



>



>



>



>



>



> *2. Q: Ok, how can I call Google Talk, MSN or Yahoo users from SIP


?*


>



> A: This is a major feature of our gateway and it is very easy.



>



> o GTalk: [EMAIL PROTECTED] can be reached by calling to



> sip:[EMAIL PROTECTED]



>



> o MSN: [EMAIL PROTECTED] can be reached by calling to



> sip:[EMAIL PROTECTED]



>



> o Yahoo: [EMAIL PROTECTED] can be reached by calling to



> sip:[EMAIL PROTECTED]



>



>



>



>





>



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>


_

Get Married in 2007. Join Shaadi.com

*http://www.shaadi.

Re: [asterisk-users] VoIP over Alvarion Wireless

[Matt]

We are running Alvarion VL 4.0.   We know it doesn't work well over BA or
900.   However, with 60 customers, all we error rates below 10% VoIP is
aweful.

On 3/7/07, Andrew Niemantsverdriet <[EMAIL PROTECTED]> wrote:


What Alvarion stuff are you using? Alvarion is the top rated wireless
manufactor when it comes to VOIP on there new stuff (VL 4.0 series and
B100 backhaul series). Some of there older stuff like the BrezeeAccess
FHSS will not work well because the QoS that the radio provides is
spotty at best.

On 3/7/07, Matt <[EMAIL PROTECTED]> wrote:
> Hi,
> This question isn't specifically asterisk related, but perhaps someone
here
> can shed some light or offer some insight.
> Is anyone else here running VoIP over Alvarion wireless?If yes, do
you
> have any suggestions for what you've done to make it "work"?   It seems
that
> no amount of traffic shaping, checking installs for error rates,
lowering
> error rates, or setting contention windows makes VoIP work. It just
> plain blows over Alvarion wireless gear.   Cable and DSL?  Yup.. it
works
> great.Wireless?  Forget it, pickup your cell phone.
>
> Anyone have any better news, or suggestions?
>
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>
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
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RE: [asterisk-users] Voicemail question

[Race Vanderdecken]

I have done per user context programming for asterisk in the recent
past.

I am a professional contract programmer but my rates are reasonable and
you will get code that works.

Please contact me if you would like my help on this.

-Race

Race Vanderdecken
Code Tyrant, Inc.
[EMAIL PROTECTED]
828 221 2636 vonage
Somewhere near Asheville, NC.
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olle E
Johansson
Sent: Wednesday, March 07, 2007 2:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voicemail question

Everything can be done with a certain amount of coding... :-) No,  
it's not possible
in Asterisk today.

Check the configuration templates to make life easier when  
configuring voicemail.
It's documented in doc/configuration.txt in your asterisk source code  
directory, or
here:
http://svn.digium.com/view/asterisk/branches/1.4/doc/configuration.txt

/Olle

6 mar 2007 kl. 00.42 skrev Hall, Eric M.:

> Group
>
>  In voicemail.conf I would like to having the following setup per  
> context not per-mailbox settings
>
>
>
> serveremail
>
> userscontext
>
> fromstring
>
> usedirectory
>
> emailbody
>
> pagerfromstring
>
> dialout
>
> sendvoicemail
>
> callback
>
> review
>
> operator
>
>  volgain
>
> nextaftercmd
>
> forcename
>
> forcegreetings
>
> tempgreetwarn

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RE: [asterisk-users] Asterisk Auto-dial out

[Michael Collins]

> I am using the * auto-dial out feature but don't want to have to
specify
> a channel (Zap/G2/) to connect to the extension.
> 
> Current file I use:
> 
> Channel: Zap/G2/12127778866   #<<  I have to specify a specific
> channel
> MaxRetries: 1
> RetryTime: 60
> WaitTime: 30
> #
> # Assuming that your outgoing call logic is kept in the
> #  context called [line1out]
> #
> Context: line1out
> Extension: 7632
> Priority: 1
> 
> Is there a way that I can just put in the number and have the system
> decide the channel to use for calling it?
> 
> What I would like to do:
> 
> Channel:   #<<=== This number could be
>#  7645 in which case go via SIP/7645
>#  68001 which should go to CiscoSIP/68001
>#  12127778866 which would go via
> Zap/G2/12127778866
> MaxRetries: 1
> RetryTime: 60
> WaitTime: 30
> #
> # Assuming that your outgoing call logic is kept in the
> #  context called [line1out]
> #
> Context: line1out
> Extension: 7632
> Priority: 1
> 
> Based on dialing plan the system should be able to route the call to
> whatever channel supports dialing that number.

You probably want to use the Local channel.  Definitely hit the wiki and
check it out:
http://www.voip-info.org/wiki/view/Asterisk+local+channels

The idea behind the local channel is that you can, in effect, drop a
call right into a specific part of the dialplan.  From there, your
dialplan can handle the logic of figuring out which technology and
channel to use.

-MC
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Re: [asterisk-users] Problem HandyTone 488 does not call transfer

[Drew Gibson]

Pablo Almido wrote:

Hi

I have a analog phone connected to my Gateway Handytone and registered 
to Asterisk 1.4 I have configured my HandyTone 488 
(in the section FXS Port) for make and receive calls, however I can 
not transfer a call when it come via PSTN. But, when a call come from 
via IP I can transfer it.


I had this issue when I connected the PSTN to the "failover" port on an 
HT 386.
The HT will pick up the call from the PSTN directly, before Asterisk can 
get to it. Check the Asterisk console and logs for an incoming call to 
see what Asterisk sees.


According to Grandstream tech support there is no way around this 
(Working As Designed), at least on the HT386.


regards,

Drew

--
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com

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[asterisk-users] Asterisk Registering to other SIP servers.

[Bala Neelakantan]

Hello,

 

I am trying to REGISTER asterisk to a SIP server, which is listening on Port
6060 (not 5060).  

 

The sip.conf file contains

 

register=1847420:[EMAIL PROTECTED]:6060/1847420

maxexpirey=3600

defaultexpirey=120

 

But the REGISTER message is sent to Port 6060, but the Request-URI still
contains, 5060.  This is being rejected by SIP server.

 

REGISTER sip:192.168.2.94 SIP/2.0

 

The request line instead should be like this:

 

REGISTER sip:192.168.2.94:6060 SIP/2.0

 

See the attached debug logs.

 

 

SIP Debugging enabled

*CLI> reload

Mar  7 13:23:12 NOTICE[17748]: cdr.c:1193 do_reload: CDR simple logging
enabled.

Mar  7 13:23:12 NOTICE[17748]: indications.c:505
ast_unregister_indication_country: Removed default indication country 'us'

Mar  7 13:23:12 WARNING[17748]: chan_zap.c:11029 setup_zap: Ignoring
signalling

Mar  7 13:23:12 WARNING[17748]: chan_zap.c:11029 setup_zap: Ignoring
signalling

*CLI> Mar  7 13:23:12 NOTICE[17759]: chan_sip.c:5465 sip_reregister:--
Re-registration for  [EMAIL PROTECTED]

REGISTER 12 headers, 0 lines

Reliably Transmitting (no NAT) to 192.168.2.94:6060:

REGISTER sip:192.168.2.94 SIP/2.0

Via: SIP/2.0/UDP 192.168.2.96:5060;branch=z9hG4bK1d8de05f;rport

From: ;tag=as40ac92ca

To: 

Call-ID: [EMAIL PROTECTED]

CSeq: 102 REGISTER

User-Agent: Asterisk PBX

Max-Forwards: 70

Expires: 120

Contact: 

Event: registration

Content-Length: 0

 

Thanks,

Neel

 

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Re: [asterisk-users] Asterisk Realtime

[Mike Hammett]

[EMAIL PROTECTED] asterisk]# cat res_mysql.conf
;
; Sample configuration for res_config_mysql.c
;
; The value of dbhost may be either a hostname or an IP address.
; If dbhost is commented out or the string "localhost", a connection
; to the local host is assumed and dbsock is used instead of TCP/IP
; to connect to the server.
;
[general]
;dbhost = 127.0.0.1
dbname = asterisk
dbuser = asterisk
dbpass = yourpassword
;dbport = 3306
dbsock =  /var/lib/mysql/mysql.sock

[EMAIL PROTECTED] asterisk]# cat extconfig.conf
;
; Static and realtime external configuration
; engine configuration
;
; Please read doc/extconfig.txt for basic table
; formatting information.
;
[settings]
;
; Static configuration files:
;
; file.conf => driver,database[,table]
;
; maps a particular configuration file to the given
; database driver, database and table (or uses the
; name of the file as the table if not specified)
;
;uncomment to load queues.conf via the odbc engine.
;
;queues.conf => odbc,asterisk,ast_config
;
; The following files CANNOT be loaded from Realtime storage:
;   asterisk.conf
;   extconfig.conf (this file)
;   logger.conf
;
; Additionally, the following files cannot be loaded from
; Realtime storage unless the storage driver is loaded
; early using 'preload' statements in modules.conf:
;   manager.conf
;   cdr.conf
;   rtp.conf
;
;
; Realtime configuration engine
;
; maps a particular family of realtime
; configuration to a given database driver,
; database and table (or uses the name of
; the family if the table is not specified
;
;example => odbc,asterisk,alttable
;example2 => ldap,"dc=oxymium,dc=net",example2
;
;iaxusers => odbc,asterisk
;iaxpeers => odbc,asterisk
;sipusers => odbc,asterisk
;sippeers => odbc,asterisk
;voicemail => odbc,asterisk
;extensions => odbc,asterisk
;queues => odbc,asterisk
;queue_members => odbc,asterisk

sippeers => mysql,asterisk,sip_peers
sipusers => mysql,asterisk,sip_users
iaxpeers => mysql,asterisk,iax_peers
iaxusers => mysql,asterisk,iax_users
queues => mysql,asterisk,queue_table
queue_members => mysql,asterisk,queue_member_table
voicemail => mysql,asterisk



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[asterisk-users] Re: Asterisk Realtime

[Mike Hammett]

  == Parsing '/etc/asterisk/res_mysql.conf': [Mar  7 14:12:37] DEBUG[4380]:
config.c:844 config_text_file_load: Parsing /etc/asterisk/res_mysql.conf
Found
[Mar  7 14:12:37] WARNING[4380]: res_config_mysql.c:555 parse_config: MySQL
RealTime: No database host found, using localhost via socket.
[Mar  7 14:12:37] WARNING[4380]: res_config_mysql.c:555 parse_config: MySQL
RealTime: No database host found, using localhost via socket.
[Mar  7 14:12:37] WARNING[4380]: res_config_mysql.c:569 parse_config: MySQL
RealTime: No database port found, using 3306 as default.
[Mar  7 14:12:37] WARNING[4380]: res_config_mysql.c:569 parse_config: MySQL
RealTime: No database port found, using 3306 as default.
[Mar  7 14:12:37] DEBUG[4380]: res_config_mysql.c:585 parse_config: MySQL
RealTime Host:
[Mar  7 14:12:37] DEBUG[4380]: res_config_mysql.c:586 parse_config: MySQL
RealTime Port: 3306
[Mar  7 14:12:37] DEBUG[4380]: res_config_mysql.c:590 parse_config: MySQL
RealTime User: asterisk
[Mar  7 14:12:37] DEBUG[4380]: res_config_mysql.c:591 parse_config: MySQL
RealTime Password: yourpassword
[Mar  7 14:12:37] DEBUG[4380]: res_config_mysql.c:623 mysql_reconnect: MySQL
RealTime: Successfully connected to database.
[Mar  7 14:12:37] NOTICE[4380]: config.c:1174 ast_config_engine_register:
Registered Config Engine mysql
[Mar  7 14:12:37] NOTICE[4380]: config.c:1174 ast_config_engine_register:
Registered Config Engine mysql
MySQL RealTime driver loaded.
res_config_mysql.so => (MySQL RealTime Configuration Driver)


--

Message: 5
Date: Thu, 01 Mar 2007 20:58:37 +0100
From: Philipp Kempgen <[EMAIL PROTECTED]>
Subject: Re: [asterisk-users] Asterisk Realtime
To: Asterisk Users Mailing List - Non-Commercial Discussion

Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-15

Brian Capouch wrote:
> Mike Hammett wrote:
>> Could someone provide some steps for troubleshooting Realtime?  I can't 
>> see any signs that it's working.  I followed and double-checked a few 
>> different guides around the net, but haven't been able to figure it out.
> 
> You don't say which version you're running.
> 
> I *think* the syntax is the same for both:
> 
> realtime  status
> 
> will show you the status.  For postgres it's pgsql for driver name 
> (that's what I use).  I think the other driver ids are mysql and odbc.
> 
> If you don't see yourself connected, that's where to start.

Or put
console => notice,warning,error,verbose,debug

in logger.conf
/ run asterisk -vvvdddc  :)

This will give you all MySQL queries and warnings.

Regards,
  Philipp

-- 
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

Geschdftsf|hrer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998


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RE: [asterisk-users] Asterisk queue and agents

[Hall, Eric M.]

I think that is already set. Here is my queue.conf



[general]
persistentmembers = yes
autofill = yes
monitor-type = MixMonitor


[support]
musicclass = default
strategy = fewestcalls
timeout = 10
retry = 5
autofill=yes
autopause=yes
setinterfacevar=no
announce-frequency = 90 
periodic-announce-frequency=60
announce-holdtime = yes
announce-round-seconds = 10
queue-youarenext = queue-youarenext
queue-thereare = queue-thereare
queue-callswaiting = queue-callswaiting
queue-holdtime = queue-holdtime
queue-minutes = queue-minutes
queue-seconds = queue-seconds
queue-thankyou = queue-thankyou
queue-lessthan = queue-less-than
queue-reporthold = queue-reporthold
;periodic-announce = queue-periodic-announce
joinempty = yes
leavewhenempty = no
eventwhencalled = vars
QueueMemberStatus=yes
eventmemberstatus = yes
reportholdtime = no
ringinuse = no
memberdelay = 1

member => agent/56416
member => agent/56420
member => agent/56421
member => agent/56423
member => agent/56405
member => agent/564221
member => agent/564321



-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Octavio Ruiz 
(Ta^3)
Sent: Wednesday, March 07, 2007 1:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk queue and agents

>Have a question for the group
>If I have an agent is on the phone outside of the queue should that person
>still get queue calls ?
>Doing a show agents online I see Available however show hints I see inuse.

There is a ringinuse feature for SIP devices on 1.4.X which is what you are 
looking for.

-- 
Octavio Ruiz Cervera
Neocenter, SA. de CV.
http://www.neocenter.com/
Soluciones para Centros de Contacto y Telefonía IP
Tel.: (+52 55) 8590-9000 Ext. 9016
Cel.: (+55 55) 5514-087790
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Re: [asterisk-users] VoIP over Alvarion Wireless

[Andrew Niemantsverdriet]

Something must be setup wrong on your radios. Alavarion is pretty good
when it comes to tech support. I would contact the distributer that
you bought it from and ask for help.

On 3/7/07, Matt <[EMAIL PROTECTED]> wrote:

We are running Alvarion VL 4.0.   We know it doesn't work well over BA or
900.   However, with 60 customers, all we error rates below 10% VoIP is
aweful.


On 3/7/07, Andrew Niemantsverdriet <[EMAIL PROTECTED]> wrote:
> What Alvarion stuff are you using? Alvarion is the top rated wireless
> manufactor when it comes to VOIP on there new stuff (VL 4.0 series and
> B100 backhaul series). Some of there older stuff like the BrezeeAccess
> FHSS will not work well because the QoS that the radio provides is
> spotty at best.
>
> On 3/7/07, Matt <[EMAIL PROTECTED]> wrote:
> > Hi,
> > This question isn't specifically asterisk related, but perhaps someone
here
> > can shed some light or offer some insight.
> > Is anyone else here running VoIP over Alvarion wireless?If yes, do
you
> > have any suggestions for what you've done to make it "work"?   It seems
that
> > no amount of traffic shaping, checking installs for error rates,
lowering
> > error rates, or setting contention windows makes VoIP work. It just
> > plain blows over Alvarion wireless gear.   Cable and DSL?  Yup.. it
works
> > great.Wireless?  Forget it, pickup your cell phone.
> >
> > Anyone have any better news, or suggestions?
> >
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> >
> >
http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
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Re: [asterisk-users] Re: Asterisk Realtime

[Brian Capouch]

Mike Hammett wrote:


[Mar  7 14:12:37] DEBUG[4380]: res_config_mysql.c:623 mysql_reconnect: MySQL
RealTime: Successfully connected to database.
[Mar  7 14:12:37] NOTICE[4380]: config.c:1174 ast_config_engine_register:
Registered Config Engine mysql
[Mar  7 14:12:37] NOTICE[4380]: config.c:1174 ast_config_engine_register:
Registered Config Engine mysql
MySQL RealTime driver loaded.
res_config_mysql.so => (MySQL RealTime Configuration Driver)



All that looks fine.

What do you get when you do "realtime mysql status?"

The next areas to look at would be your DB configs, and debug status 
when you actually try to use one of the entries in your DB. . .


I only use it for iaxpeers/users and extensions, so I can't comment much 
on its use with SIP or voicemail.


B.

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Re: [asterisk-users] Asterisk Registering to other SIP servers.

[Olle E Johansson]

7 mar 2007 kl. 20.48 skrev Bala Neelakantan:


Hello,


I am trying to REGISTER asterisk to a SIP server, which is  
listening on Port 6060 (not 5060).



The sip.conf file contains


register=1847420:[EMAIL PROTECTED]:6060/1847420

maxexpirey=3600

defaultexpirey=120


But the REGISTER message is sent to Port 6060, but the Request-URI  
still contains, 5060.  This is being rejected by SIP server.

REGISTER sip:192.168.2.94 SIP/2.0
I don't see "5060" in the request URI ?

I don't see 6060 either, which should be there...

Please file a bug report at bugs.digium.com

Thanks
/O



REGISTER sip:192.168.2.94 SIP/2.0


The request line instead should be like this:


REGISTER sip:192.168.2.94:6060 SIP/2.0


See the attached debug logs.



SIP Debugging enabled

*CLI> reload

Mar  7 13:23:12 NOTICE[17748]: cdr.c:1193 do_reload: CDR simple  
logging enabled.


Mar  7 13:23:12 NOTICE[17748]: indications.c:505  
ast_unregister_indication_country: Removed default indication  
country 'us'


Mar  7 13:23:12 WARNING[17748]: chan_zap.c:11029 setup_zap:  
Ignoring signalling


Mar  7 13:23:12 WARNING[17748]: chan_zap.c:11029 setup_zap:  
Ignoring signalling


*CLI> Mar  7 13:23:12 NOTICE[17759]: chan_sip.c:5465  
sip_reregister:-- Re-registration for  [EMAIL PROTECTED]


REGISTER 12 headers, 0 lines

Reliably Transmitting (no NAT) to 192.168.2.94:6060:

REGISTER sip:192.168.2.94 SIP/2.0

Via: SIP/2.0/UDP 192.168.2.96:5060;branch=z9hG4bK1d8de05f;rport

From: ;tag=as40ac92ca

To: 

Call-ID: [EMAIL PROTECTED]

CSeq: 102 REGISTER

User-Agent: Asterisk PBX

Max-Forwards: 70

Expires: 120

Contact: 

Event: registration

Content-Length: 0


Thanks,

Neel


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---
* Olle E Johansson - [EMAIL PROTECTED]
* Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden



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[asterisk-users] auto dialer

[Hall, Eric M.]

Not able to get the auto dialer part of asterisk to work with the zap
channel. It works great with the sip channel. Here is the call file and
the CLI output

 

Call File 

Channel: ZAP/G1/6144994925

MaxRetries: 3

RetryTime: 40

WaitTime: 2

Context: amaxx

Extension: 36652 

Priority: 1

 

 

 

 

 

 

CLI Output

Connected to Asterisk SVN-branch-1.4-r57207 currently running on
VoIP-PBX (pid = 8002)

Verbosity is at least 3

-- Attempting call on ZAP/G1/6144994925 for [EMAIL PROTECTED]:1 (Retry 1)

-- Requested transfer capability: 0x00 - SPEECH

-- Hungup 'Zap/23-1'

[Mar  7 15:46:29] NOTICE[10159]: pbx_spool.c:341 attempt_thread: Call
failed to go through, reason 0

VoIP-PBX*CLI> 

 

 

 

 

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Re: [asterisk-users] VoIP over Alvarion Wireless

[Matt]

Oh, We've been in contact with them.They said we need to purchase WLP..
which we are in the process of doing.. but then we found some other
interesting info out.I'll post a follow up here in a few days if I
remember.

On 3/7/07, Andrew Niemantsverdriet <[EMAIL PROTECTED]> wrote:


Something must be setup wrong on your radios. Alavarion is pretty good
when it comes to tech support. I would contact the distributer that
you bought it from and ask for help.

On 3/7/07, Matt <[EMAIL PROTECTED]> wrote:
> We are running Alvarion VL 4.0.   We know it doesn't work well over BA
or
> 900.   However, with 60 customers, all we error rates below 10% VoIP is
> aweful.
>
>
> On 3/7/07, Andrew Niemantsverdriet <[EMAIL PROTECTED]> wrote:
> > What Alvarion stuff are you using? Alvarion is the top rated wireless
> > manufactor when it comes to VOIP on there new stuff (VL 4.0 series and
> > B100 backhaul series). Some of there older stuff like the BrezeeAccess
> > FHSS will not work well because the QoS that the radio provides is
> > spotty at best.
> >
> > On 3/7/07, Matt <[EMAIL PROTECTED]> wrote:
> > > Hi,
> > > This question isn't specifically asterisk related, but perhaps
someone
> here
> > > can shed some light or offer some insight.
> > > Is anyone else here running VoIP over Alvarion wireless?If yes,
do
> you
> > > have any suggestions for what you've done to make it "work"?   It
seems
> that
> > > no amount of traffic shaping, checking installs for error rates,
> lowering
> > > error rates, or setting contention windows makes VoIP work. It
just
> > > plain blows over Alvarion wireless gear.   Cable and DSL?  Yup.. it
> works
> > > great.Wireless?  Forget it, pickup your cell phone.
> > >
> > > Anyone have any better news, or suggestions?
> > >
> > > ___
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> > >
> > > asterisk-users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > >
> > >
> http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > >
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> >
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> >
>
>
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>
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Re: [asterisk-users] auto dialer

[Melcon Moraes]

WaitTime stands for how long to wait until the call is considered "NO
ANSWERED"

Who can pickup a phone in 2 seconds, if not a robot? Try switch values
between Retrytime and WaitTime.

[]'s
MM

 -Original Message-
From:   "Hall, Eric M." <[EMAIL PROTECTED]>
To: 
Cc: 
Sent:  Wed, 7 Mar 2007 15:53:23 -0500
Delivered:  Wed,  07 Mar 2007 17:45:35 
Subject:[asterisk-users] auto dialer

Not able to get the auto dialer part of asterisk to workwith the zap 
channel. It works great with the sip channel. Here is the callfile and the CLI 
output
 
Call File 
Channel: ZAP/G1/6144994925
MaxRetries: 3
RetryTime: 40
WaitTime: 2
Context: amaxx
Extension: 36652 
Priority: 1
 
 
 
 
 
 
CLI Output
Connected to Asterisk SVN-branch-1.4-r57207 currentlyrunning on VoIP-PBX (pid = 
8002)
Verbosity is at least 3
-- Attempting call on ZAP/G1/6144994925 [EMAIL PROTECTED]:1 (Retry 1)
-- Requested transfer capability: 0x00 - SPEECH
-- Hungup 'Zap/23-1'
[Mar  7 15:46:29] NOTICE[10159]: pbx_spool.c:341attempt_thread: Call failed to 
go through, reason 0
VoIP-PBX*CLI> 
 
 
 
 


E-mail classificado pelo Identificador de Spam Inteligente.
Para alterar a categoria classificada, visite o 
Terra
 Mail

 --Original Message Ends--

-- 
Melcon Moraes <[EMAIL PROTECTED]>

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RE: [asterisk-users] auto dialer

[Hall, Eric M.]

OK now I fell like a a$$... Thanks for that kick in the butt !!



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Melcon
Moraes
Sent: Wednesday, March 07, 2007 4:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] auto dialer

WaitTime stands for how long to wait until the call is considered "NO
ANSWERED"

Who can pickup a phone in 2 seconds, if not a robot? Try switch values
between Retrytime and WaitTime.

[]'s
MM

 -Original Message-
From:   "Hall, Eric M." <[EMAIL PROTECTED]>
To: 
Cc: 
Sent:  Wed, 7 Mar 2007 15:53:23 -0500
Delivered:  Wed,  07 Mar 2007 17:45:35 
Subject:[asterisk-users] auto dialer

Not able to get the auto dialer part of asterisk to workwith the zap
channel. It works great with the sip channel. Here is the callfile and
the CLI output
 
Call File 
Channel: ZAP/G1/6144994925
MaxRetries: 3
RetryTime: 40
WaitTime: 2
Context: amaxx
Extension: 36652 
Priority: 1
 
 
 
 
 
 
CLI Output
Connected to Asterisk SVN-branch-1.4-r57207 currentlyrunning on VoIP-PBX
(pid = 8002)
Verbosity is at least 3
-- Attempting call on ZAP/G1/6144994925 [EMAIL PROTECTED]:1 (Retry 1)
-- Requested transfer capability: 0x00 - SPEECH
-- Hungup 'Zap/23-1'
[Mar  7 15:46:29] NOTICE[10159]: pbx_spool.c:341attempt_thread: Call
failed to go through, reason 0
VoIP-PBX*CLI> 
 
 
 
 


E-mail classificado pelo Identificador de Spam Inteligente.
Para alterar a categoria classificada, visite o
Ter
ra Mail

 --Original Message Ends--

-- 
Melcon Moraes <[EMAIL PROTECTED]>

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RE: [asterisk-users] Asterisk queue and agents

[Hall, Eric M.]

Looks like it's a bug 

http://bugs.digium.com/view.php?id=9172&nbn=3

I have update to Asterisk SVN-branch-1.4-r58243 and will test it tomorrow and 
report back to the list.



Eric Hall


-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Octavio Ruiz 
(Ta^3)
Sent: Wednesday, March 07, 2007 1:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk queue and agents

>Have a question for the group
>If I have an agent is on the phone outside of the queue should that person
>still get queue calls ?
>Doing a show agents online I see Available however show hints I see inuse.

There is a ringinuse feature for SIP devices on 1.4.X which is what you are 
looking for.

-- 
Octavio Ruiz Cervera
Neocenter, SA. de CV.
http://www.neocenter.com/
Soluciones para Centros de Contacto y Telefonía IP
Tel.: (+52 55) 8590-9000 Ext. 9016
Cel.: (+55 55) 5514-087790
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[asterisk-users] Problems with Zaptel Drivers

[Lutgring, Sam]

Has anyone run into the problem of having your FXO ports show up as FXS?
I believe that my drivers are loading because I see the channels appear
under /dev/zap. I am using Fedora Core 6 with the latest Zaptel 1.4.0
drivers and a 8 port FXO Xorcom Astribank. All of my indicator lights on
the Astribank are solid red. 

My XPD-0 summary file under proc reads: 
XPD-0 (FXS ,card present, span registeres) SYNC MASTER 
and so on 

My /proc/zaptel/1 file reads: 
SPAN 1: XBUS-0/XPD-0 "Xorcom XPD #0/0: FXS" 

 1 XPP_FXS/0/0/0 FXOKS (In Use) 
 2 XPP_FXS/0/0/1 FXOKS (In Use) 
 3 XPP_FXS/0/0/2 FXOKS (In Use) 
 4 XPP_FXS/0/0/3 FXOKS (In Use) 
 5 XPP_FXS/0/0/4 FXOKS (In Use) 
 6 XPP_FXS/0/0/5 FXOKS (In Use) 
 7 XPP_FXS/0/0/6 FXOKS (In Use) 
 8 XPP_FXS/0/0/7 FXOKS (In Use) 
 9 XPP_OUT/0/0/8 FXOKS 
10 XPP_OUT/0/0/9 FXOKS 
11 XPP_OUT/0/0/10 FXOKS 
12 XPP_OUT/0/0/11 FXOKS 
13 XPP_OUT/0/0/12 FXOKS 
14 XPP_OUT/0/0/13 FXOKS

This should be showing up as an FXO ports.  Any suggestions would be
greatly appreciated. 
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Re: [asterisk-users] Problems with Zaptel Drivers

[Tzafrir Cohen]

On Wed, Mar 07, 2007 at 08:39:36PM -0500, Lutgring, Sam wrote:
> Has anyone run into the problem of having your FXO ports show up as FXS?

FXS channels have FXO signalling. Yeah, this is confusing.

An FXS port is the one to which you connect phones. FXO ports emulate
phones, and can connect to a standard telephone-company socket.

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

> I believe that my drivers are loading because I see the channels appear
> under /dev/zap. I am using Fedora Core 6 with the latest Zaptel 1.4.0
> drivers and a 8 port FXO Xorcom Astribank. All of my indicator lights on
> the Astribank are solid red. 
> 
> My XPD-0 summary file under proc reads: 
> XPD-0 (FXS ,card present, span registeres) SYNC MASTER 
> and so on 
> 
> My /proc/zaptel/1 file reads: 
> SPAN 1: XBUS-0/XPD-0 "Xorcom XPD #0/0: FXS" 
> 
>  1 XPP_FXS/0/0/0 FXOKS (In Use) 

 port (FXS)signalling (FXO)

>  2 XPP_FXS/0/0/1 FXOKS (In Use) 
>  3 XPP_FXS/0/0/2 FXOKS (In Use) 
>  4 XPP_FXS/0/0/3 FXOKS (In Use) 
>  5 XPP_FXS/0/0/4 FXOKS (In Use) 
>  6 XPP_FXS/0/0/5 FXOKS (In Use) 
>  7 XPP_FXS/0/0/6 FXOKS (In Use) 
>  8 XPP_FXS/0/0/7 FXOKS (In Use) 
>  9 XPP_OUT/0/0/8 FXOKS 
> 10 XPP_OUT/0/0/9 FXOKS 
> 11 XPP_OUT/0/0/10 FXOKS 
> 12 XPP_OUT/0/0/11 FXOKS 
> 13 XPP_OUT/0/0/12 FXOKS 
> 14 XPP_OUT/0/0/13 FXOKS

Huh? Is this an exact paste? The last four ports should be inputs:

One of the servers here has:

 [snip]
   9 XPP_OUT/0/0/8 FXOLS (In use) 
  10 XPP_OUT/0/0/9 FXOLS (In use) 
  11 XPP_IN/0/0/10 FXOLS (In use) 
  12 XPP_IN/0/0/11 FXOLS (In use) 
  13 XPP_IN/0/0/12 FXOLS (In use) 
  14 XPP_IN/0/0/13 FXOLS (In use) 


> 
> This should be showing up as an FXO ports.  Any suggestions would be
> greatly appreciated. 



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

[Bill Gibbs]

Behavior on Asterisk 1.2.12, 1.2.15, 1.2.16

"sip show channels"

 

Always tends to show 100+ lines such as

192.168.1.241(None)  2e2872da-1d  00101/21507  unkn  No
Rx: REGISTER   

 

Never seem to go away

 

198 total peers on this server

All devices are behind NAT

Registration expirations between 30secs to 2 minutes to help keep NAT
open

 

Should I extend the registration expiration to maybe 5 minutes?  Will
that keep the NAT holes on the remote router open long enough to receive
calls?

 

Bill

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Re: [asterisk-users] Asterisk queue and agents

[BJ Weschke]

I don't think this is a bug.

From UPGRADE.txt:

* Queues depend on the channel driver reporting the proper state
 for each member of the queue. To get proper signalling on
 queue members that use the SIP channel driver, you need to
 enable a call limit (could be set to a high value so it
 is not put into action) and also make sure that both inbound
 and outbound calls are accounted for.

 Example:

  [general]
  limitonpeer = yes

  [peername]
  type=friend
  call-limit=10



Please test with that and report your findings, and if it's still not
working find us on IRC as we'd like to take a further look and see
what might be wrong.

BJ

On 3/7/07, Hall, Eric M. <[EMAIL PROTECTED]> wrote:

Looks like it's a bug

http://bugs.digium.com/view.php?id=9172&nbn=3

I have update to Asterisk SVN-branch-1.4-r58243 and will test it tomorrow and 
report back to the list.



Eric Hall


-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Octavio Ruiz 
(Ta^3)
Sent: Wednesday, March 07, 2007 1:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk queue and agents

>Have a question for the group
>If I have an agent is on the phone outside of the queue should that person
>still get queue calls ?
>Doing a show agents online I see Available however show hints I see inuse.

There is a ringinuse feature for SIP devices on 1.4.X which is what you are 
looking for.

--
Octavio Ruiz Cervera
Neocenter, SA. de CV.
http://www.neocenter.com/
Soluciones para Centros de Contacto y Telefonía IP
Tel.: (+52 55) 8590-9000 Ext. 9016
Cel.: (+55 55) 5514-087790
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RE: [asterisk-users] Asterisk queue and agents

[Hall, Eric M.]

BJ
 Here is the sip.conf file. Hints work great. The only problem is the queue is 
sending calls to an agent that's on the phone.


[general]
rtcachefriends=yes
videosupport=yes
port=5060 ; Port to bind to (SIP is 5060)
bindaddr=0.0.0.0  ; Address to bind to (all addresses on machine)
context=sip ; Send unknown SIP callers to this context
allow=g729
allow=h263 ; H.263 is our video codec
allow=h263p ; H.263p is the enhanced video codec
;allow=g711
;allow=all
;allow=ulaw
;allow=gsm
nat=1
host=dynamic
type=peer
qualify=yes
notifyringing=yes
Subscribecontext=sip
call-limit=300
notifyhold = yes
limitonpeer = yes
notifyringing = yes; Notify subscriptions on RINGING state 
(default: no)
notifyhold = yes


[56405] ;Eric Test
type=friend   ; "friend" means this device takes and makes calls
username=1 ; Username on device
callerid=Eric Test Phone  <56405>
secret=56405; Password for device
host=dynamic  ; This host is not on the same IP addr every time
context=sip ; Inbound calls from this host go here
[EMAIL PROTECTED]; Activate the message waiting light if this
canreinvite=no; Leave this alone for now; see archives for details
nat=1
qualify=yes
Subscribecontext=sip
notifyringing=yes
call-limit=300



-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke
Sent: Wednesday, March 07, 2007 10:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk queue and agents

 I don't think this is a bug.

 From UPGRADE.txt:

* Queues depend on the channel driver reporting the proper state
  for each member of the queue. To get proper signalling on
  queue members that use the SIP channel driver, you need to
  enable a call limit (could be set to a high value so it
  is not put into action) and also make sure that both inbound
  and outbound calls are accounted for.

  Example:

   [general]
   limitonpeer = yes

   [peername]
   type=friend
   call-limit=10



 Please test with that and report your findings, and if it's still not
working find us on IRC as we'd like to take a further look and see
what might be wrong.

 BJ

On 3/7/07, Hall, Eric M. <[EMAIL PROTECTED]> wrote:
> Looks like it's a bug
>
> http://bugs.digium.com/view.php?id=9172&nbn=3
>
> I have update to Asterisk SVN-branch-1.4-r58243 and will test it tomorrow and 
> report back to the list.
>
>
>
> Eric Hall
>
>
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Octavio Ruiz 
> (Ta^3)
> Sent: Wednesday, March 07, 2007 1:05 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Asterisk queue and agents
>
> >Have a question for the group
> >If I have an agent is on the phone outside of the queue should that 
> > person
> >still get queue calls ?
> >Doing a show agents online I see Available however show hints I see 
> > inuse.
>
> There is a ringinuse feature for SIP devices on 1.4.X which is what you are 
> looking for.
>
> --
> Octavio Ruiz Cervera
> Neocenter, SA. de CV.
> http://www.neocenter.com/
> Soluciones para Centros de Contacto y Telefonía IP
> Tel.: (+52 55) 8590-9000 Ext. 9016
> Cel.: (+55 55) 5514-087790
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[asterisk-users] Number of SIP messages per minute

[Mark Davies]

Hi all,

 

I've just been told from an ex workmate that my VSP (who I used to work
for) has put an anti flooding limit of 80 SIP messages per IP per minute
in place.

 

I run the phone system for a facility that has a lot of extensions, but
would rarely have more than 4 or 5 simultaneous external calls.  Am I in
danger of tripping over this limit?

 

It sounds dangerously low to me.

 

Thanks in advance,

 

 

Mark.

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Re: [asterisk-users] Asterisk 1.4.1 Released

[voiplist]

Um, is it possible to patch 1.2.4? We have some pretty busy
production systems and are not exactly excited about having to upgrade
from this version.

Is there no other way to protect our systems from this hole?



On 3/3/07, Thomas Kenyon <[EMAIL PROTECTED]> wrote:

Michelle Dupuis wrote:
> Can one do an in-place update to 1.4.1 from 1.4.0 ?  (Just compile and
> install)
>
./configure
make update worked here.
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[asterisk-users] SIP remote crash bug

[voiplist]

We run 1.2.4, do we have to upgrade? Is there a patch for this version?

At this point we REALLY don't want to upgrade and potentially
introduce a bunch of new issues and problems including our AGI's all
breaking due to SET and SETVAR etc..

Thanks in advance for any insight on this issue.
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[asterisk-users] Re: Back to back E1 - asterisk <=> toshiba pbx - Call droping

[Vidura Senadeera]

Hi steve and All,

I'm attaching cat proc/interrupts out put, lspci -bv out put, zapa.conf,
zaptel.conf for your information

Thanks so much for the feedback and I do accordingly. Hope to get rid off
this isue any how.
To day also reported 10 call drops within 2 hours of period.

fook forward to have your support on this regard.


Thanks & Regards,

Vidura Senadeera,

Network Engineer,

Debug Solutions

Sri Lanka.

Tel - +94114520036

Mobile - +9466596

Web - www.debug.lk



Message: 16

Date: Wed, 7 Mar 2007 05:05:36 -0500
From: "Steve Totaro" <[EMAIL PROTECTED]>
Subject: RE: [asterisk-users] Back to back E1 - asterisk <=> toshiba
   pbx -   Calldroping issue
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
   
Message-ID:
   <
[EMAIL PROTECTED]>

Content-Type: text/plain; charset="us-ascii"

As these problems are very time sensitive and frustrating, I suggest you
document each change you make and do them one at a time so you can
actually know what the problem was and not introduce new problems in the
process.



Find someone who is on the phone quite a bit and will give you an honest
evaluation of the call dropping situation (unless you yourself are
experiencing this issue too).  Some people are so quick to say, "It is
still happening" without starting the evaluation from a clean slate
after each change.



You may want to check your Asterisk log for more insight.
/var/log/asterisk/full.  Also you can turn on debugging on one span at a
time and see if you can find something there



Do you have a resetinterval set in zapata.conf?  If you can isolate the
dropped calls to the reset interval (watch the console, it will scroll
with each channel being reset) then set resetinterval=never.  If there
is no entry for resetinterval, add it and set it to never since it is
defaulted to on.



Also, try changing your second span timing from span=2,2,0,ccs,hdb3,crc4
to span=2,0,0,ccs,hdb3,crc4.  This in combination with your first span
should accept timing from the Telco and then supply it to your Toshiba,
I would actually try this first.



Another thought, It seems you have quite a lot of hardware in that box.
I am not sure how much is too much, but that would probably just rear
it's ugly head as poor audio.

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com


_

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vidura
Senadeera
Sent: Wednesday, March 07, 2007 2:15 AM
To: [EMAIL PROTECTED]
Cc: asterisk-users@lists.digium.com
Subject: [asterisk-users] Back to back E1 - asterisk <=> toshiba pbx -
Calldroping issue





Hi Team,



I have integrated asterisk with Toshiba analog PBX. NOw the live setup
is going.



Now I am facing call droping problem. It's happening ample time. 10-20
calls are droping every day.



What could be the reason. I attached latest zapata.conf file for your
information.







This is being a huge issue.



Highly appreciate your help on this regard.


Thanks & Regards,

Vidura Senadeera.




On 1/26/07, Vidura Senadeera <[EMAIL PROTECTED] > wrote:

Dear Marco,



There is a huge problem i'm facing.



My asterisk server included with TDM2451E and 2 TE110p cards. One E1 i
conected to the telco. other E1 port i'm using to cros-connection with
toshiba pbx. My telco E1 d channels communicating well. but toshiba pbx
E1 not getting. d-channels are not getting up.

what could be the issue. i'm using asterisk -1.2.14 and zaptel 1.2.12.



notes - if i put, zap show channels in asterisk cli. its only showing
the first 31 channels. but with ztcfg -vvv it showing al the channels.



my configs are



# Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" HDB3/CCS/CRC4 RED

#  Suntel E1 connection ==

span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16

# Span 2: WCT1/1 "Digium Wildcard TE110P T1/E1 Card 1"
#  Legacy PBX E1 connection ===

span=2,2,0,ccs,hdb3,crc4
bchan=32-46,48-62
dchan=47

# Span 3: WCTDM/0 "Wildcard TDM2400P Prototype Board 1"
fxoks=63
fxoks=64
fxoks=65
fxoks=66
fxoks=67
fxoks=68
fxoks=69
fxoks=70
fxoks=71
fxoks=72
fxoks=73
fxoks=74
fxoks=75
fxoks=76
fxoks=77
fxoks=78
fxoks=79
fxoks=80
fxoks=81
fxoks=82
fxsks=83
fxsks=84
fxsks=85
fxsks=86

# Global data

loadzone= us
defaultzone = us

Regards,

vidura





--
Thanks & Regards,
Vidura B. Senadeera.




--
Thanks & Regards,
Vidura B. Senadeera.




--
Thanks & Regards,
Vidura B. Senadeera.

-- next part --
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--

Message: 17
Date: Wed, 7 Mar 2007 11:17:07 +0100
From: "Thomas Deillon" <[EMAIL PROTECTED]>
Subject: [asterisk-users] Asterisk 1.4.1 - Calling problem
To: "Asterisk Users 

[asterisk-users] Fwd: Back to back E1 - asterisk <=> toshiba pbx - Call droping

[Vidura Senadeera]

-- Forwarded message --
From: Vidura Senadeera <[EMAIL PROTECTED]>
Date: Mar 8, 2007 11:27 AM
Subject: Re: Back to back E1 - asterisk <=> toshiba pbx - Call droping
To: asterisk-users@lists.digium.com


Hi steve and All,

I'm attaching cat proc/interrupts out put, lspci -bv out put, zapa.conf,
zaptel.conf for your information

Thanks so much for the feedback and I do accordingly. Hope to get rid off
this isue any how.
To day also reported 10 call drops within 2 hours of period.

fook forward to have your support on this regard.


Thanks & Regards,

Vidura Senadeera,

Network Engineer,

Debug Solutions

Sri Lanka.

Tel - +94114520036

Mobile - +9466596

Web - www.debug.lk



Message: 16

Date: Wed, 7 Mar 2007 05:05:36 -0500
From: "Steve Totaro" < [EMAIL PROTECTED]>
Subject: RE: [asterisk-users] Back to back E1 - asterisk <=> toshiba
   pbx -   Calldroping issue
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
   
Message-ID:
   <[EMAIL PROTECTED]
>

Content-Type: text/plain; charset="us-ascii"

As these problems are very time sensitive and frustrating, I suggest you
document each change you make and do them one at a time so you can
actually know what the problem was and not introduce new problems in the
process.



Find someone who is on the phone quite a bit and will give you an honest
evaluation of the call dropping situation (unless you yourself are
experiencing this issue too).  Some people are so quick to say, "It is
still happening" without starting the evaluation from a clean slate
after each change.



You may want to check your Asterisk log for more insight.
/var/log/asterisk/full.  Also you can turn on debugging on one span at a
time and see if you can find something there



Do you have a resetinterval set in zapata.conf?  If you can isolate the
dropped calls to the reset interval (watch the console, it will scroll
with each channel being reset) then set resetinterval=never.  If there
is no entry for resetinterval, add it and set it to never since it is
defaulted to on.



Also, try changing your second span timing from span=2,2,0,ccs,hdb3,crc4
to span=2,0,0,ccs,hdb3,crc4.  This in combination with your first span
should accept timing from the Telco and then supply it to your Toshiba,
I would actually try this first.



Another thought, It seems you have quite a lot of hardware in that box.
I am not sure how much is too much, but that would probably just rear
it's ugly head as poor audio.

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com


_

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vidura
Senadeera
Sent: Wednesday, March 07, 2007 2:15 AM
To: [EMAIL PROTECTED]
Cc: asterisk-users@lists.digium.com
Subject: [asterisk-users] Back to back E1 - asterisk <=> toshiba pbx -
Calldroping issue





Hi Team,



I have integrated asterisk with Toshiba analog PBX. NOw the live setup
is going.



Now I am facing call droping problem. It's happening ample time. 10-20
calls are droping every day.



What could be the reason. I attached latest zapata.conf file for your
information.







This is being a huge issue.



Highly appreciate your help on this regard.


Thanks & Regards,

Vidura Senadeera.




On 1/26/07, Vidura Senadeera <[EMAIL PROTECTED] > wrote:

Dear Marco,



There is a huge problem i'm facing.



My asterisk server included with TDM2451E and 2 TE110p cards. One E1 i
conected to the telco. other E1 port i'm using to cros-connection with
toshiba pbx. My telco E1 d channels communicating well. but toshiba pbx
E1 not getting. d-channels are not getting up.

what could be the issue. i'm using asterisk -1.2.14 and zaptel 1.2.12.



notes - if i put, zap show channels in asterisk cli. its only showing
the first 31 channels. but with ztcfg -vvv it showing al the channels.



my configs are



# Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" HDB3/CCS/CRC4 RED

#  Suntel E1 connection ==

span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16

# Span 2: WCT1/1 "Digium Wildcard TE110P T1/E1 Card 1"
#  Legacy PBX E1 connection ===

span=2,2,0,ccs,hdb3,crc4
bchan=32-46,48-62
dchan=47

# Span 3: WCTDM/0 "Wildcard TDM2400P Prototype Board 1"
fxoks=63
fxoks=64
fxoks=65
fxoks=66
fxoks=67
fxoks=68
fxoks=69
fxoks=70
fxoks=71
fxoks=72
fxoks=73
fxoks=74
fxoks=75
fxoks=76
fxoks=77
fxoks=78
fxoks=79
fxoks=80
fxoks=81
fxoks=82
fxsks=83
fxsks=84
fxsks=85
fxsks=86

# Global data

loadzone= us
defaultzone = us

Regards,

vidura





--
Thanks & Regards,
Vidura B. Senadeera.




--
Thanks & Regards,
Vidura B. Senadeera.




--
Thanks & Regards,
Vidura B. Senadeera.

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[asterisk-users] Call recording and archiving

[Voip Asterisk]

Does anyone have a good suggestion for a automated solution to record calls
on certain interfaces and easily archiving them in a way which is easily
matched against CDRs?  Also can someone suggest the appropriate protocol to
archive the recording when the conversations are transpiring in ulaw.
Basically a nice cost effective trade off between CPU and disk space for
medium call load.

Thanks

Miles
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Re: [asterisk-users] sip show channels

[Olle E Johansson]

8 mar 2007 kl. 04.16 skrev Bill Gibbs:


Behavior on Asterisk 1.2.12, 1.2.15, 1.2.16

“sip show channels”



Always tends to show 100+ lines such as

192.168.1.241(None)  2e2872da-1d  00101/21507  unkn   
No   Rx: REGISTER




Never seem to go away



198 total peers on this server

All devices are behind NAT

Registration expirations between 30secs to 2 minutes to help keep  
NAT open




Should I extend the registration expiration to maybe 5 minutes?   
Will that keep the NAT holes on the remote router open long enough  
to receive calls?



If you have that short an expiry time, there's no reason to question  
the 100+ open dialogs at any time.
"sip show channel" doesn't only show calls, it shows all kinds of SIP  
dialogs except the subscriptions.


The dialogs are kept open for about 30 secs to make sure we handle  
retransmits properly. The network
may have dropped our 200 OK and if the REGISTER comes back in  
retransmit, we need to handle it.
Within that time, your devices send a re-register and the dialog  
stays open.


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Re: [asterisk-users] Asterisk 1.4.1 Released

[Olle E Johansson]

8 mar 2007 kl. 05.33 skrev voiplist:


Um, is it possible to patch 1.2.4? We have some pretty busy
production systems and are not exactly excited about having to upgrade
from this version.

Is there no other way to protect our systems from this hole?

You can apply the patch that was applied to 1.2, found by following  
this URL:


http://tinyurl.com/2soujh

Remember that we released a new version of 1.2, 1.4 and updated svn  
trunk.


1.0 is no longer maintained, since it is too old.

/O
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Re: [asterisk-users] Re: queue information into db

[nik600]

https://sourceforge.net/projects/ccmanager/

please note that it is a beta version, i'd like to improve it but i'm
busy with work and university.

take a look and let me know.

nik

On 3/6/07, nik600 <[EMAIL PROTECTED]> wrote:

i've submittet the project to SF.

I have to wait 2 business days for their validation.

The project is in a beta release and will be released on GPL.

Bye

On 3/2/07, Tomislav Parcina <[EMAIL PROTECTED]> wrote:
> nik600 wrote:
> > i'm sorry but due to some problem the software will be released not
> > first than Wednesday 7/02/2007. i'll post a message .
>
> This should be Wednesday 7/3/2007. right?
>
>
> --
> Tomislav Parcina
> [EMAIL PROTECTED]
>
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