[asterisk-users] Asterisk both behind a NAT and outside at the same time

[Brad Templeton]

I've read a lot of the descriptions of handling NAT with Asterisk,
and the use of both the nat and canreinvite flags.  I am very
familiar with Sip and NAT but have not seen an answer to the following
question.


My Asterisk server runs on a machine with two ethernets.  One is
an external net, with exposed IP addresses.   The other is an internal
net with natted IP addresses.   Thus the server has two addresses. 

The server is _not_ the NAT gateway.  That's a linksys box which has
its own external IP to gateway traffic from the internal natwork.

The phones are on the internal NATwork.   Asterisk talks to them over
it.   Outside peers, such as SIP termination providers etc. talk
to the Asterisk server via its outside address, which is as you
would expect.

However, from time to time I get the famous one-way audio because
Asterisk has decided to do a native bridge between a natted SIP
phone and an external SIP peer.   It sends the internal IP of
the SIP phone in the SDP and of course the outside service can't
send packets to that.

I could just turn off reinvites on the internal phones, but this
would cause them to route all traffic through the asterisk box,
even on internal calls between phones on the same ethernet, which
seems foolish to me.   I don't want to turn off reinvites to the
external peers -- if a call comes in from a SIP originator for example,
and is send back out to a SIP terminator (call forwarding) I want
a native bridge for sure.(Handling the internal traffic is not
so much of a burden though sometimes I hear latency because of it, but
routing external traffic through the asterisk box is a bad thing.)

So what I want is for Asterisk to use native bridges when connecting
two channels behind the NAT, or two channels on the real internet, but
not to do so when connecting an internal and external channel.

It should be able to see the IP addresses, and know the difference between
natted and external ones and know they can't talk to one another.
(The ICE protocol would handle this someday.)

Is IAX smarter about this?

Of course I might even want to get smarter about this.  Is it possible,
typically by configuring stun in the phones, to have them be aware of their
external IP and tell Asterisk about it?  With a full cone NAT, it would
work to do a native bridge between the internal and external devices
so long as the external device is given the right address and port of
the NAT box, not the internal address of the phone.   However, we don't
want to do this on internal to internal calls -- many NATs can't hairpin.


I would think this would be a common situation (though perhaps more
commonly the asterisk server IS the firewall/NAT.)   Is there a
solution that does the right thing most of the time?
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[asterisk-users] Fedora Core 6 (FC6) and Asterisk-1.2.13 and Zaptel-1.2.10 compile problems

[Michael J. Tubby G8TIC]

All,

I have upgraded by home machine from Fedora Core 5 
(FC5) to the recent FC6 and am struggling to build Zaptel-1.2.10 and 
Asterisk-1.2.13 on the box... which is an Intep P4 2.8GHz HT processor box with 
845 chipset, hence the kernel installed is 2.6.18-1.2798.fc6-i686so we hve 
this:

 [EMAIL PROTECTED] zaptel-1.2.10]# uname 
-a Linux gate.tubby.org 2.6.18-1.2798.fc6 #1 SMP Mon Oct 
16 14:54:20 EDT 2006 i686 i686 i386 GNU/Linux
I have ensured that the kernel-devel package and 
glibc-kernel headers are installed.

When I attempt to compile Zaptel-1.2.10 I get the 
following:

 make[1]: Entering directory 
`/usr/src/kernels/2.6.18-1.2798.fc6-i686'  CC 
[M] /root/asterisk/zaptel-1.2.10/zaptel.o In file 
included from /root/asterisk/zaptel-1.2.10/zaptel.c:40: 
/root/asterisk/zaptel-1.2.10/zconfig.h:9:26: error: linux/config.h: No such file 
or directory make[2]: *** 
[/root/asterisk/zaptel-1.2.10/zaptel.o] Error 1 make[1]: 
*** [_module_/root/asterisk/zaptel-1.2.10] Error 2 
make[1]: Leaving directory 
`/usr/src/kernels/2.6.18-1.2798.fc6-i686' make: *** 
[linux26] Error 2

and can confirm that the file does not exist in the 
path to the kernel source:

 
/usr/src/kernels/2.6.18-1.2798.fc6-i686/include/linux/

Grabbing the 2.6.18 source from ftp.kernel.org and unpacking it finds the following 
contents:

 #ifndef 
_LINUX_CONFIG_H #define 
_LINUX_CONFIG_H /* This file is no longer in use and kept 
only for backward compatibility. * autoconf.h is now 
included via -imacros on the commandline 
*/ #include linux/autoconf.h
 
 #endif

but it doesn't exist in the FC6 kernel tree for 
2.6.18-1.2798.fc6 ...


Likewise, attempting to build Asterisk-1.2.13 
fails thusly:

 chan_phone.c:41:29: error: 
linux/compiler.h: No such file or directory make[1]: *** 
[chan_phone.o] Error 1 make[1]: Leaving directory 
`/root/asterisk/asterisk-1.2.13/channels' make: *** 
[subdirs] Error 1 [EMAIL PROTECTED] 
asterisk-1.2.13]#

and the path:

 /usr/include/linux

doesn't have a compiler.h :o(

Hacking at the Makefile I added:

 
INCLUDE+=-I/usr/src/kernels/2.6.18-1.2798.fc6-i686

which gets us a compiler.h and the build process 
continues somewhat further but then blows up due to a lack of zaptel.h (back to 
first problem)


My box is acient, having been RedHat 9, then FC2, 
FC3, FC5 and now FC6 ... so just to prove I'm not going mad (or possibly more 
likely the progressive upgrades left something broken) I did a clean install of 
FC6 to a spare box and tried to build with the same results.


So, to a bit more hacking... I copied "config.h" from the clean 2.6.18 
kernel source tree to the FC6 tree in inlcude/linux and now I can compile and 
install the zaptel stuff, but I can't load it:

 [EMAIL PROTECTED] zaptel-1.2.10]# modprobe 
zaptel FATAL: Error inserting zaptel 
(/lib/modules/2.6.18-1.2798.fc6/misc/zaptel.ko): Invalid module format

I've copied the compiler.h across to /usr/inlcude/linux and now I can 
compile and link Asterisk-1.2.13 along with Asterisk-Addons-1.2.5 so I am able 
to get my system back online but only by removing the zaptel configuration 
:o(


Is Asterisk using deprecated kernel header files 
that some distros have decided to stop supporting/shipping? What do I need to do 
to get Zaptel drivers correctly compiled for my FC6 box?



Mike

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[asterisk-users] Asterisk does not bridge zap channels on outgoing calls

[Alexandru Voinescu]

Hello... I have a big problem with asterisk. Every time i make a call 
asterisk does not bridge the zap channels. The zap channel from which 
i'm calling remains in state:ring and applicaton:dial and the zap 
channel with the external line configured remains in state:dialling an 
Application:AppDial.

Zap/20-1 agentie s 1 Dialing AppDial (Outgoing Line) 09399 (None)
Zap/9-1 int_omg 09399 5 Ring Dial Zap/g5/9399||T 00:07:58 (None)

Obviously, this is a big problem for us...
Below are my zapata.conf, zaptel.conf and extensions.conf:

--
zapata.conf
--
[channels]

usecallerid=yes
hidecallerid=no
echocancel=yes
musiconhold=service
busydetect=yes
;callprogress=yes
busycount=3
flash=20
rxflash=40
transfer=yes
threewaycalling=yes
;rxgain=100%
;txgain=1.0
;relaxdtmf=yes



;--
context=int_soft
;--
group=1
callgroup=1
pickupgroup=1,6,7
threewaycalling=yes
transfer=yes
;useincomingcalleridonzaptransfer=yes
callwaitingcallerid=yes
;echocancelwhenbriged=yes
immediate=no

rxgain=-2.0
txgain=2.2

signalling=fxo_ks

callerid=Soft 1
channel=1

callerid=Soft 2
channel=2

callerid=Soft 3
channel=3

callerid=Soft 4
channel=4

callerid=Soft 5
channel=5

callerid=Soft 6
channel=6

callerid=Soft 7
channel=7

;--
context=int_omg
;--
group=2
callgroup=2
pickupgroup=2,5,6
threewaycalling=yes
transfer=yes
;useincomingcalleridonzaptransfer=yes
callwaitingcallerid=yes
;echocancelwhenbriged=yes
immediate=no

rxgain=-2.0
txgain=2.2

callerid=OMG 28
channel=8

callerid=OMG 29
channel=9

callerid=OMG 30
channel=10

callerid=OMG 31
channel=11

callerid=OMG 32
channel=12

callerid=OMG 33
channel=13

callerid=OMG 34
channel=14

callerid=OMG 35
channel=15

callerid=OMG 36
channel=16

;---
;Placa TDM24XXP - 24 DE INTERIOARE; CONTEXT NOU!!
;---

;--
context=int_agentie
;--
group=3
callgroup=3
pickupgroup=3,4,5,10
threewaycalling=yes
transfer=yes
;useincomingcalleridonzaptransfer=yes
callwaitingcallerid=yes
;echocancelwhenbriged=yes
immediate=no

rxgain=-2.0
txgain=2.2

callerid=Agentie 45
channel=25

callerid=Agentie 46
channel=26

callerid=Agentie 47
channel=27

callerid=Agentie 48
channel=28

callerid=Agentie 49
channel=29

callerid=Agentie 50
channel=30

callerid=Agentie 51
channel=31

callerid=Agentie 52
channel=32

callerid=Agentie 53
channel=33

callerid=Agentie 54
channel=34

callerid=Agentie 55
channel=35

callerid=Agentie 56
channel=36

callerid=Agentie 57
channel=37

callerid=Agentie 58
channel=38

callerid=Agentie 59
channel=39

callerid=Agentie 60
channel=40

callerid=Agentie 61
channel=41

callerid=Agentie 62
channel=42

callerid=Agentie 63
channel=43

callerid=Agentie 64
channel=44

callerid=Clopotel 65
channel=45

callerid=Clopotel 66
channel=46

callerid=Clopotel 67
channel=47

callerid=Clopotel 68
channel=48

;--
context=cap_hunting
;--
group=4
callgroup=4
callerid=asreceived
cidsignalling=v23
cidstart=ring
transfer=yes
threewaycalling=yes
immediate=no
useincomingcalleridonzaptransfer=yes
;echocancelwhenbriged=yes
;musiconhold=guitar
sendcalleridafter=2

rxgain=10.2
txgain=1.8

signalling=fxs_ks
channel=17

;--
context=omegasoft
;--
group=10
callgroup=10
callerid=asreceived
cidsignalling=v23
cidstart=ring
transfer=yes
;threewaycalling=yes
immediate=no
useincomingcalleridonzaptransfer=yes
;echocancelwhenbriged=yes
;musiconhold=guitar
sendcalleridafter=2

rxgain=10.2
txgain=1.8

signalling=fxs_ks
channel=18

;--
context=agentie
;--
group=5
callgroup=5
callerid=asreceived
cidsignalling=v23
cidstart=ring
transfer=yes
threewaycalling=yes
;immediate=no
;useincomingcalleridonzaptransfer=yes
;echocancelwhenbriged=yes
;musiconhold=guitar
sendcalleridafter=2

rxgain=10.2
txgain=1.8

signalling=fxs_ks
channel=19-24

;--
context=tehnic
;--
group=6
callgroup=6
callerid=asreceived
cidsignalling=v23
cidstart=ring
transfer=yes
threewaycalling=yes
;immediate=no
useincomingcalleridonzaptransfer=yes
;callprogres=yes
;echocancelwhenbriged=yes

rxgain=10.2
txgain=1.8

signalling=fxs_ks
channel=80

;--
context=service
;--
group=7
callgroup=7
;echocancelwhenbriged=yes
callerid=asreceived
answeronpolarityswitch=yes
hanguponpolarityswitch=yes
;cidstart=ring
;cidsignalling=v23
sendcalleridafter=2

rxgain=5.5
txgain=7.2

signalling=fxs_ks
channel=81

;--
context=service_out
;--
group=7
callgroup=7
;echocancelwhenbriged=yes
callerid=asreceived
;cidstart=ring
;cidsignalling=v23
answeronpolarityswitch=yes
hanguponpolarityswitch=yes
sendcalleridafter=2

rxgain=4.5
txgain=5.2

[asterisk-users] Bridging Video Calls using Zap

[Steffen Weinreich]

Hi!

For demonstration purposes I try to bridge an incoming video call from a
3G mobile handset to another 3G mobile handset using asterisk as switch.

On the incoming call leg I see all expected bearer capabilities
(Digital, 64k Transparent, G.7xx 384k video) but on the outgoing call
leg the bearer capability G.7xx 384k video get lost and therefore the
call is dropped from the mobile operator.

What I have tried so far ist to use SetTransferCapability(VIDEO) but
this does not change the behavior.

Is there a way to set or preserve the bearer capability for the outgoing
call leg?

cheerio
Steve


--- environment

pbx-test*CLI show version
Asterisk 1.2.9.1-BRIstuffed-0.3.0-PRE-1q built by root @
pbx-test.bb.ic3s.de on a i686 running Linux on 2006-06-16 10:17:00 UTC

with A Quad ZAP Pri Card

 dialplan

  ;exten = 297,1,SetTransferCapability(VIDEO)
  exten = 297,1,Noop()
  exten = 297,2,Dial(${TRUNK}/0175234567)
  exten = 297,3,Hangup
  exten = 297,104,SetVar(PRI_CAUSE=17) ; Indicate Busy
  exten = 297,105,Hangup

 Log
pbx-test*CLI
  == Primary D-Channel on span 2 down
Oct 31 09:05:03 WARNING[6771]: chan_zap.c:2506 pri_find_dchan: No
D-channels available!  Using Primary channel 47 as D-channel anyway!
1  Protocol Discriminator: Q.931 (8)  len=45
1  Call Ref: len= 2 (reference 25880/0x6518) (Originator)
1  Message type: SETUP (5)
1  [1 a11 ]
1  Sending Complete (len= 1)
1  [1 041  031  881  901  a61 ]
1  Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer
capability: Unrestricted digital information (8)
1   Ext: 1  Trans mode/rate: 64kbps,
circuit-mode (16)
1   Ext: 1  User information layer 1: G.7xx
384k Video (38)
1  [1 181  031  a11  831  871 ]
1  Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0,
Preferred Dchan: 0
1 ChanSel: Reserved
1Ext: 1  Coding: 0   Number Specified   Channel
Type: 3
1Ext: 1  Channel: 7 ]
1  [1 6c1  0c1  211  831  311  371  351  351  381  361  341  371  381
311 ]
1  Calling Number (len=14) [ Ext: 0  TON: National Number (2)  NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
1Presentation: Presentation allowed of
network provided number (3) '1755864781' ]
1  [1 701  081  c11  351  351  351  361  321  391  371 ]
1  Called Number (len=10) [ Ext: 1  TON: Subscriber Number (4)  NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '5556297' ]
1  [1 7c1  031  881  901  a61 ]
1  Low-layer compatibilty (len= 5) [ 1 0x88 1 0x90 1 0xA6 1  ]
1 -- Making new call for cr 25880
1 -- Processing Q.931 Call Setup
1 -- Processing IE 161 (cs0, Sending Complete)
1 -- Processing IE 4 (cs0, Bearer Capability)
1 -- Processing IE 24 (cs0, Channel Identification)
1 -- Processing IE 108 (cs0, Calling Party Number)
1 -- Processing IE 112 (cs0, Called Party Number)
1 -- Processing IE 124 (cs0, Low-layer Compatibility)
Oct 31 09:05:07 WARNING[6770]: chan_zap.c:8503 zt_pri_error: 1 copying 5
bytes LLC
1  Protocol Discriminator: Q.931 (8)  len=10
1  Call Ref: len= 2 (reference 25880/0x6518) (Terminator)
1  Message type: CALL PROCEEDING (2)
1  [1 181  031  a91  831  871 ]
1  Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0,
Exclusive Dchan: 0
1 ChanSel: Reserved
1Ext: 1  Coding: 0   Number Specified   Channel
Type: 3
1Ext: 1  Channel: 7 ]
-- Accepting data call from '1755864781' to '5556297' on channel
0/7, span 1
-- Executing Macro(Zap/7-1, handle-callerid) in new stack
-- Executing NoOp(Zap/7-1, 01751234567) in new stack
-- Executing GotoIf(Zap/7-1, 0?3:5) in new stack
-- Goto (macro-handle-callerid,s,5)
-- Executing SetCallerID(Zap/7-1, 001751234567) in new stack
-- Executing LookupCIDName(Zap/7-1, ) in new stack
-- Changed Caller*ID name to Testi Tester
-- Executing NoOp(Zap/7-1, Testi Tester 001751234567 / Testi
Tester / 001751234567) in new stack
-- Executing Goto(Zap/7-1, external-call|297|1) in new stack
-- Goto (external-call,297,1)
-- Executing Dial(Zap/7-1, Zap/r1/01752345678) in new stack
1 -- Making new call for cr 32777
-- Requested transfer capability: 0x08 - DIGITAL
1  Protocol Discriminator: Q.931 (8)  len=44
1  Call Ref: len= 2 (reference 9/0x9) (Originator)
1  Message type: SETUP (5)
1  [1 041  021  881  901 ]
1  Bearer Capability (len= 4) [ Ext: 1  Q.931 Std: 0  Info transfer
capability: Unrestricted digital information (8)
1   Ext: 1  Trans mode/rate: 64kbps,
circuit-mode (16)
1   Ext: 0  User information layer 1:
Unknown (24)
1  [1 181  031  a91  831  881 ]
1  Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0,
Exclusive Dchan: 0
1 ChanSel: Reserved
1Ext: 1  Coding: 0   Number Specified   Channel
Type: 3
1Ext: 1  Channel: 8 ]
1  [1 6c1 

[asterisk-users] Read cmd

[Dovid B]

Hi list,
I have been using the read command and I have 
noticed that it behaves more like the playback command and not background. Is 
there any way to set it up that I can enter a selection before Asterisk finishes 
playing the file ?
Thanks.

Dovid
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[asterisk-users] Setting up UTStarcom F300

[Dovid B]

Hi List,
I have a F300. I have not yet been able to set it 
up. Can anyone email me the exact configs on how they set it up to work with 
Asterisk ?

Thanks.

Dovid
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Re: [asterisk-users] light web user interface

[Dovid B]

What attributes are you talking about ? Depending 
on what they are it may be real simple to set something up.

  - Original Message - 
  From: 
  Curt Shaffer 
  
  To: 'Asterisk Users Mailing List - 
  Non-Commercial Discussion' 
  Sent: Monday, October 30, 2006 9:51 
  PM
  Subject: [asterisk-users] light web user 
  interface
  
  
  Does anyone know of a really 
  lightweight web interface that allows users to log in and modify attributes of 
  their extension only?
  
  Thanks
  
  Curt
  
  

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Re: [asterisk-users] Re: Architecture for Asterisk

[Olle E Johansson]

31 okt 2006 kl. 01.37 skrev je .:

Thanks for the diagram. Is it possible to get a more detailed  
diagram. I'm looking for something a little more technical. In  
other words, where does Asterisk stand when inviting a user, when  
hanging up, when canceling an invitation etc.. Does it go direct  
from user to user or does it go to Asterisk?




Asterisk is not a SIP proxy, we're a back-to-back-SIP-ua, b2bua.  
Everything ends in Asterisk and Asterisk,

being a PBX, decides what to do next.

In some cases we re-invite the media to go p2p, but SIP signalling  
ends in Asterisk.


In SIP terminology: We're a SIP registrar/location server, SIP  
outbound proxy (or can be configured as one to

be more correct) and a SIP b2bua, but not a SIP proxy.

/O
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Re: [asterisk-users] Architecture for Asterisk

[Olle E Johansson]

31 okt 2006 kl. 06.34 skrev je .:

Thank you for the link. Chapter 8 was most useful in explaining the  
different types of connections (user/peer/friend) as well as the  
register function such that users may know how to contact another  
user. However I'm looking for something more specific. For  
instance, for a normal session termination (i.e. BYE), user1 would  
send msg BYE to proxy who would then forward it to user2 who would  
then close the connection. For a cancel request the following  
messages are exchanged:


u1 - proxy: invite
proxy - u2: arp request


In this scenario for instance (where user 2 closes the connection),  
where does Asterisk fit in? If at all? Does Asterisk behave as the  
proxy?


In an Asterisk scenario, you would have two different calls, one from  
u1 to Asterisk, one from Asterisk to u2.
We never forward any SIP messages, like a SIP proxy. If u1 hangs up,  
we decide to hang up our call to u2,

but those are two different SIP dialogs.

Remember that Asterisk is a multiprotocol PBX. The connection to U2  
might be using a different signalling

protocol, like ISDN PRI.

/O

---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/
* Next training: Stockholm, Sweden, November 13th!



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Re: [asterisk-users] Multiple dial macros at the same time

[Dovid B]

When user A is bridged with the client (right before it is done) set a 
variable that the call was taken, this way when he presses 1 to accpet the 
system checks to see what the value is, if its in taken mode he gets the 
message and then gets hung up on.


- Original Message - 
From: Graham Mainwaring [EMAIL PROTECTED]

To: asterisk-users@lists.digium.com
Sent: Monday, October 30, 2006 3:38 AM
Subject: [asterisk-users] Multiple dial macros at the same time


I am setting up an after-hours on-call system. Someone calls in and 
requests service, and while they listen to music on hold, we dial out to 
several people's cell phones and home phones. We don't know if they will be 
answered by the employee, or by voicemail or a spouse/relative/child/pet. 
So we play a message that says press 1 to accept the call and ask 
employees to train their spouse/relative/child/pets not to press 1.


The following extract from my dialplan shows how I have this feature set 
up. This is with Asterisk 1.2.13.


 [macro-screen]
 exten = s,1,Set(MACRO_RESULT=CONTINUE)
 exten = s,2,Read(ACCEPT|press-1-to-accept|1|skip|3|1)
 exten = s,3,GotoIf($[${ACCEPT}=1]?5:4)
 exten = s,4,MacroExit
 exten = s,5,Set(MACRO_RESULT=)
 exten = s,6,Playback(please-say-hello)

 [menu]
 exten = _FOLS1NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],20,oM(screen))
 exten = s,1,Playback(welcome)
 exten = s,2,Dial(LOCAL/FOLS19195551000LOCAL/FOLS19195552000,,tm)
 exten = s,3,Voicemail(u301)
 exten = s,4,Hangup

In order for this to work, I needed the ability to restore MACRO_RESULT 
back to an unset state. For now I just hacked the Set application so that 
after removing the variable from the context, it only re-creates it if the 
value provided is greater than zero length. In the future I will probably 
write an UnSet application to handle this more gracefully, unless someone 
knows a better way to unset a variable.


This all works fine, with one small problem that is driving me batty. I 
would appreciate any insight or ideas on how to solve this. Here's the 
scenario:


 1. Caller dials the number and hears the welcome message,
then music on hold.
 2. Simultaneous calls are made to Employee A at 555-1000
and Employee B at 555-2000 (per above).
 3. Both of them answer the phone.
 4. Employee A presses one and hears you will now be connected,
please say hello to the caller.
 5. Employee A is bridged to the caller, says hello, and begins
working with them.
 6. A few seconds later, the Employee B also presses 1. He also
hears you will now be connected.
 7. Employee B fails to bridge, and is hung up on.

The problem is, these are pretty urgent calls and employees are highly 
motivated to make sure they get answered. Employee B doesn't know whether 
the call dropped because someone else got it, or because of a phone system 
problem of some sort. He is now obligated to figure out what's up with the 
call and make sure someone got it. What I want instead is for Employee B 
to hear an alternate message that says someone else got the call. This 
gives positive confirmation that it's not his problem, so he can roll over 
and go back to sleep.


I can see two ways of doing this.

1. Write a function called BridgedChannel that takes a channel ID and 
returns its bridge peer channel ID, if any. This would allow me to set a 
variable __PARENTCHANNEL with the channel ID of the incoming call, before 
the Dial command executes. The macro, at priority 6, can then check 
BridgedChannel(${__PARENTCHANNEL}). If it has a value then the call is 
already bridged and we can tell the employee not to worry.


2. Have a MySQL database with a single table with two fields, varchar 
channel-ID and boolean answered. When the call starts do update table set 
answered=false where channel-ID=${__PARENTCHANNEL}. When an employee dials 
1, retrieve the value of answered for __PARENTCHANNEL and also set it to 
true in a single transaction. If the returned value was false, tell them 
to answer the call and bridge; if the returned value was true, tell them 
to go back to sleep and hang up.


Solution #1 requires me to write a whole new function, and solution #2 
requires a MySQL database, which is pretty big dependency for such a 
simple function.


Does anyone see a simpler way of doing this, or have any ideas for other 
avenues to pursue?


Thanks in advance,

-Graham
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[asterisk-users] best gui

[Altus Snyman]

Good day

Im look at

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

And I see there are a few GUI for asterisk

What do you guys prefer?

What is the best and simplest? Id like something that give me
access to backend for a little bit of customization

Thanks for you help and time








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[asterisk-users] Re: IAX2 show peers - description

[Tomislav Parčina]

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
 Hi,
 
 I think the (T) is for Trunk.
 
 Regards
 Fred

Hi Fred!

I believe that T is for trunk. Thank you.


--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)270248
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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Re: [asterisk-users] Asterisk both behind a NAT and outside at the same time

[Leo Ann Boon]

Brad Templeton wrote:

I've read a lot of the descriptions of handling NAT with Asterisk,
and the use of both the nat and canreinvite flags.  I am very
familiar with Sip and NAT but have not seen an answer to the following
question.


My Asterisk server runs on a machine with two ethernets.  One is
an external net, with exposed IP addresses.   The other is an internal
net with natted IP addresses.   Thus the server has two addresses. 


The server is _not_ the NAT gateway.  That's a linksys box which has
its own external IP to gateway traffic from the internal natwork.

The phones are on the internal NATwork.   Asterisk talks to them over
it.   Outside peers, such as SIP termination providers etc. talk
to the Asterisk server via its outside address, which is as you
would expect.

However, from time to time I get the famous one-way audio because
Asterisk has decided to do a native bridge between a natted SIP
phone and an external SIP peer.   It sends the internal IP of
the SIP phone in the SDP and of course the outside service can't
send packets to that.

I could just turn off reinvites on the internal phones, but this
would cause them to route all traffic through the asterisk box,
even on internal calls between phones on the same ethernet, which
seems foolish to me.   I don't want to turn off reinvites to the
external peers -- if a call comes in from a SIP originator for example,
and is send back out to a SIP terminator (call forwarding) I want
a native bridge for sure.(Handling the internal traffic is not
so much of a burden though sometimes I hear latency because of it, but
routing external traffic through the asterisk box is a bad thing.)

So what I want is for Asterisk to use native bridges when connecting
two channels behind the NAT, or two channels on the real internet, but
not to do so when connecting an internal and external channel.

It should be able to see the IP addresses, and know the difference between
natted and external ones and know they can't talk to one another.
(The ICE protocol would handle this someday.)

  

Have you tried setting the externalip and localnet parameters?

Leo

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Re: [asterisk-users] S(x) - Hang up the call after 'x' seconds - Not working from queue

[Leo Ann Boon]

Rajkumar S wrote:


   -- Executing Queue(SIP/1002-74e9, Auth-Enq|t) in new stack
   -- Started music on hold, class 'default', on channel 'SIP/1002-74e9'
   -- outgoing agentcall, to agent '1001', on 
'Local/[EMAIL PROTECTED],1'

   -- Executing Dial(Local/[EMAIL PROTECTED],2,
SIP/1001||tS(30)) in new stack
   -- Setting call duration limit to 30 seconds.
   -- Called 1001
   -- Called Agent/1001
   -- SIP/1001-d43c is ringing
   -- Agent/1001 is ringing
   -- SIP/1001-d43c answered Local/[EMAIL PROTECTED],2
   -- Agent/1001 answered SIP/1002-74e9
   -- Stopped music on hold on SIP/1002-74e9
 == Spawn extension (from-sip, 1001, 1) exited non-zero on
'Local/[EMAIL PROTECTED],2'
 == Spawn extension (from-sip, 99, 1) exited non-zero on 'SIP/1002-74e9'

Someone correct me if I'm wrong: The Dial string is missing a '/n' 
parameter for the Local channel. Without /n, Asterisk will do a native 
transfer to SIP/1001 and lose the timeout value defined earlier.


Leo.

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[asterisk-users] Dropping extra frame of G.729 since we already have a VAD frame at the end

[laurent schweizer]

Hello,

I have a problem with a new provider and perhaps you can help me .

whenI send traffic to this new provider i have this error :

Dropping extra frame of G.729 since we already have a VAD frame at the end

and the quality of the voice is bad because some parts of words are dropped.


Laurent

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Re: [asterisk-users] Diva server 4bri and Portuguese BRI

[Armin Schindler]

On Tue, 31 Oct 2006, Pedro Silva wrote:
 Hello,
 
 One problem is solved and another appears... :(
 I cannot receive incoming calls on trixbox. I defined one incoming
 route (any DID/any CID) and forwading these calls to a SIP extension.
 With capi and sip debug in asterisk -r console i dont detect any
 incoming activity...

Did you use
  set verbose 5
  capi debug
?
If not, you should see anything there. But if you don't see activity with 
this verbose level too, this call is not signaled through capi. In that case
you should create traces with
  divactrl ditrace
(or the trace wizard) to get capi activity too.

Armin

 In xlog console i have the following debug:
  0:1898:127 - SIG-R(034) 08 01 0D 05 A1 04 03 80 90 A3 18 01 81 6C 0B
 00 83 39 36 33 30 34 35 37 32 33 70 02 81 30 7D 02 91 81
 Q.931  CR0d SETUP
 Sending complete
 Bearer Capability 80 90 a3
 Channel Id 81
 Calling Party Number 00 83 '963045723'
 Called Party Number 81 '0'
 HLC 91 81
 0:1898:127 - SIG-S 0-6 e:805
 0:1898:130 - CREATEID ok: context:1f assigned Id:3 freeIds=ec
 0:1898:130 - alloc cr in use =4
 0:1898:133 - SIG-X(008) 08 01 8D 45 08 02 80 95
 Q.931  CR8d DISC
Cause 80 95 'Call rejected'
 0:1898:133 - SIG-x(008) 08 01 8D 5A 08 02 80 D8
 Q.931  CR8d REL_COM
Cause 80 d8 'Incompatible destination'
 0:1898:133 - SIG-S 6-0 e:8c5
 0:1898:134 - D-X(012) 00 01 14 16 08 01 8D 5A 08 02 80 D8
 0:1898:135 - free cr in use =3
 0:1898:135 - DELETEID ok: deleted Id:4 freeIds=ec
 0:1898:155 - D-R(004) 00 01 01 16
 
 So the problem appears to be Incompatible destination... but is
 problem in asterisk or is before asterisk, on diva card...?
 
 Tanks by any possible help!
 Best regards,
 PS.
 
 2006/10/29, Pedro Silva [EMAIL PROTECTED]:
  Finally this works!!! :)
  Tanks to Alberto and Marco by your help!
  The problems are:
  - the cable was connected to the wong card port... :(
  - the card config needs to be: ETSI; TE; Point-to-Point (I thought
  that was point-to-multipoint).
  
  Best regards,
  PS.
  
  2006/10/29, Pedro Silva [EMAIL PROTECTED]:
   Hello again Alberto!
   
Anyway, to get more info, try to open a second shell
and run /usr/lib/eicon/divas/xlog
then on the first shell redo the telsampl test, then
post the output of xlog off the list to my address
(alberto at msoft-italia.com)
   
   This is the xlog output:
   4:1736:074 - CREATEID ok: context:1f assigned Id:6 freeIds=eb
   4:1736:074 - alloc cr in use =4
   4:1736:076 - free cr in use =3
   4:1736:077 - DELETEID ok: deleted Id:6 freeIds=eb
   4:1736:078 - CREATEID ok: context:1f assigned Id:6 freeIds=eb
   4:1736:078 - alloc cr in use =4
   4:1736:080 - free cr in use =3
   4:1736:080 - DELETEID ok: deleted Id:6 freeIds=eb
   4:1736:081 - CREATEID ok: context:1f assigned Id:6 freeIds=eb
   4:1736:081 - alloc cr in use =4
   4:1736:083 - [1,0] dsp_assign 0016, 0, 2048
   4:1736:083 - [1,0] CAI[00] 16 00 00 00 00 08
   4:1736:084 - [1,0] Download 532 requested
   4:1736:084 - MORE
   4:1736:085 - SIG-X(029) 08 01 36 05 A1 04 03 80 90 A3 18 01 83
   1E
   02 80 83 70 0A 80 39 36 33 30 34 35 37 32 33
   Q.931  CR36 SETUP
   Sending complete
   Bearer Capability 80 90 a3
   Channel Id 83
   Progress Indicator 80 83
   Called Party Number 80 '963045723'
   4:1736:085 - SIG-S 0-1 e:885
   4:1736:087 - ACTIVATION_REQ
   4:1744:147 - L1_DOWN
   4:1744:150 - SIG-EVENT  08
   
   4:1744:150 - SIG-EVENT  08
   
   4:1744:150 - EVENT: Call failed in State 'Call initiated'
   Link disconnected, Layer-1 error (cable or NT)
   4:1744:150 - SIG-S 1-0 e:
   4:1744:151 - [1,0] dsp_release
   4:1744:155 - free cr in use =3
   4:1744:156 - DELETEID ok: deleted Id:6 freeIds=eb
   
   I disconnect the rj45 cable from alcatel pbx and connect that to the
   diva card (with alcatel pbx i can make calls normally). The green
   led
   of the diva card is activated when i connect the cable. So i dont
   understand why the message  Link disconnected, Layer-1 error (cable
   or NT)...
   This debug is th same if the cable is connected to the NT or not.
   Any ideas...? Thanks!
   PS.
   
  
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Re: [asterisk-users] Diva server 4bri and Portuguese BRI

[Marco Mouta]

Hi Pedro,

pls post your capi.conf! I'm not used with CAPI, but should have something like:

[interfaces]
incomingmsn=*  ; Here you match MSNs arriving from telco, for debug
let it ' * '
controller=1
softdtmf=1
accountcode=
context=demo   ;Set this to ext-did that should be the context
TRIXBOX will handle


By the way, Trybox probably has an extensions_custom.conf

In this file try this:

[ext-did-custom]
exten=_X.,1,Answer
exten= _X.,n,Noop(Debugging MSNs from Telco: ${EXTEN})
exten= _X.,n,wait(1)
exten= _X.,n,playback(tt-monkeys)
exten=_X.,n,hangup



Hope this helps.

Pls give some feedback


On 10/31/06, Armin Schindler [EMAIL PROTECTED] wrote:

On Tue, 31 Oct 2006, Pedro Silva wrote:
 Hello,

 One problem is solved and another appears... :(
 I cannot receive incoming calls on trixbox. I defined one incoming
 route (any DID/any CID) and forwading these calls to a SIP extension.
 With capi and sip debug in asterisk -r console i dont detect any
 incoming activity...

Did you use
  set verbose 5
  capi debug
?
If not, you should see anything there. But if you don't see activity with
this verbose level too, this call is not signaled through capi. In that case
you should create traces with
  divactrl ditrace
(or the trace wizard) to get capi activity too.

Armin

 In xlog console i have the following debug:
  0:1898:127 - SIG-R(034) 08 01 0D 05 A1 04 03 80 90 A3 18 01 81 6C 0B
 00 83 39 36 33 30 34 35 37 32 33 70 02 81 30 7D 02 91 81
 Q.931  CR0d SETUP
 Sending complete
 Bearer Capability 80 90 a3
 Channel Id 81
 Calling Party Number 00 83 '963045723'
 Called Party Number 81 '0'
 HLC 91 81
 0:1898:127 - SIG-S 0-6 e:805
 0:1898:130 - CREATEID ok: context:1f assigned Id:3 freeIds=ec
 0:1898:130 - alloc cr in use =4
 0:1898:133 - SIG-X(008) 08 01 8D 45 08 02 80 95
 Q.931  CR8d DISC
Cause 80 95 'Call rejected'
 0:1898:133 - SIG-x(008) 08 01 8D 5A 08 02 80 D8
 Q.931  CR8d REL_COM
Cause 80 d8 'Incompatible destination'
 0:1898:133 - SIG-S 6-0 e:8c5
 0:1898:134 - D-X(012) 00 01 14 16 08 01 8D 5A 08 02 80 D8
 0:1898:135 - free cr in use =3
 0:1898:135 - DELETEID ok: deleted Id:4 freeIds=ec
 0:1898:155 - D-R(004) 00 01 01 16

 So the problem appears to be Incompatible destination... but is
 problem in asterisk or is before asterisk, on diva card...?

 Tanks by any possible help!
 Best regards,
 PS.

 2006/10/29, Pedro Silva [EMAIL PROTECTED]:
  Finally this works!!! :)
  Tanks to Alberto and Marco by your help!
  The problems are:
  - the cable was connected to the wong card port... :(
  - the card config needs to be: ETSI; TE; Point-to-Point (I thought
  that was point-to-multipoint).
 
  Best regards,
  PS.
 
  2006/10/29, Pedro Silva [EMAIL PROTECTED]:
   Hello again Alberto!
  
Anyway, to get more info, try to open a second shell
and run /usr/lib/eicon/divas/xlog
then on the first shell redo the telsampl test, then
post the output of xlog off the list to my address
(alberto at msoft-italia.com)
  
   This is the xlog output:
   4:1736:074 - CREATEID ok: context:1f assigned Id:6 freeIds=eb
   4:1736:074 - alloc cr in use =4
   4:1736:076 - free cr in use =3
   4:1736:077 - DELETEID ok: deleted Id:6 freeIds=eb
   4:1736:078 - CREATEID ok: context:1f assigned Id:6 freeIds=eb
   4:1736:078 - alloc cr in use =4
   4:1736:080 - free cr in use =3
   4:1736:080 - DELETEID ok: deleted Id:6 freeIds=eb
   4:1736:081 - CREATEID ok: context:1f assigned Id:6 freeIds=eb
   4:1736:081 - alloc cr in use =4
   4:1736:083 - [1,0] dsp_assign 0016, 0, 2048
   4:1736:083 - [1,0] CAI[00] 16 00 00 00 00 08
   4:1736:084 - [1,0] Download 532 requested
   4:1736:084 - MORE
   4:1736:085 - SIG-X(029) 08 01 36 05 A1 04 03 80 90 A3 18 01 83
   1E
   02 80 83 70 0A 80 39 36 33 30 34 35 37 32 33
   Q.931  CR36 SETUP
   Sending complete
   Bearer Capability 80 90 a3
   Channel Id 83
   Progress Indicator 80 83
   Called Party Number 80 '963045723'
   4:1736:085 - SIG-S 0-1 e:885
   4:1736:087 - ACTIVATION_REQ
   4:1744:147 - L1_DOWN
   4:1744:150 - SIG-EVENT  08
  
   4:1744:150 - SIG-EVENT  08
  
   4:1744:150 - EVENT: Call failed in State 'Call initiated'
   Link disconnected, Layer-1 error (cable or NT)
   4:1744:150 - SIG-S 1-0 e:
   4:1744:151 - [1,0] dsp_release
   4:1744:155 - free cr in use =3
   4:1744:156 - DELETEID ok: deleted Id:6 freeIds=eb
  
   I disconnect the rj45 cable from alcatel pbx and connect that to the
   diva card (with alcatel pbx i can make calls normally). The green
   led
   of the diva card is activated when i connect the cable. So i dont
   understand why the message  Link disconnected, Layer-1 error (cable
   or NT)...
   This debug is th same if the cable is connected to the NT or not.
   Any ideas...? Thanks!
   PS.
  
 
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[asterisk-users] sip realtime broken?

[Don]

Can someone confirm if sip realtime is broken in 
1.2.13 and if so when was the last release it wasn't? 
heh
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Re: [asterisk-users] sip realtime broken?

[Marco Mouta]

i've been using it sucessfully

On 10/31/06, Don [EMAIL PROTECTED] wrote:



Can someone confirm if sip realtime is broken in 1.2.13 and if so when was
the last release it wasn't? heh
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--
Com os melhores cumprimentos,

Marco Mouta
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[asterisk-users] dial D option with w for wait?

[BerkHolz, Steven]

From WIKI:
D(digits): After the called party answers, send digits as a DTMF stream,
then connect the call to the originating channel. (You can also use 'w'
to produce .5 second pauses.) 

When I use the D option to send a call to my paging system and pick a
zone, the Tone is too early.
I have tried the 'w' option, but it does not appear to work.

No matter how many 'w's I use, the tone is still immediately on answer.

Is this a known issue?
Is there a work around?

My current workaround is to send both channel to a meetme that runs a
macro to play the tone.
This is way to much overhead to play a single tone after .5 or 1
seconds.

Please advise.

 

Thank You,

Steven BerkHolz

Board member of
www.glimasoutheast.org


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Re: [asterisk-users] Bridging Video Calls using Zap

[Tristan]

Hi,

You have to modify the setup sent for the call outgoing...


Regards,
Tristan


Steffen Weinreich a écrit :

Hi!

For demonstration purposes I try to bridge an incoming video call from a
3G mobile handset to another 3G mobile handset using asterisk as switch.

On the incoming call leg I see all expected bearer capabilities
(Digital, 64k Transparent, G.7xx 384k video) but on the outgoing call
leg the bearer capability G.7xx 384k video get lost and therefore the
call is dropped from the mobile operator.

What I have tried so far ist to use SetTransferCapability(VIDEO) but
this does not change the behavior.

Is there a way to set or preserve the bearer capability for the outgoing
call leg?

cheerio
Steve


--- environment

pbx-test*CLI show version
Asterisk 1.2.9.1-BRIstuffed-0.3.0-PRE-1q built by root @
pbx-test.bb.ic3s.de on a i686 running Linux on 2006-06-16 10:17:00 UTC

with A Quad ZAP Pri Card

 dialplan

  ;exten = 297,1,SetTransferCapability(VIDEO)
  exten = 297,1,Noop()
  exten = 297,2,Dial(${TRUNK}/0175234567)
  exten = 297,3,Hangup
  exten = 297,104,SetVar(PRI_CAUSE=17) ; Indicate Busy
  exten = 297,105,Hangup

 Log
pbx-test*CLI
  == Primary D-Channel on span 2 down
Oct 31 09:05:03 WARNING[6771]: chan_zap.c:2506 pri_find_dchan: No
D-channels available!  Using Primary channel 47 as D-channel anyway!
1  Protocol Discriminator: Q.931 (8)  len=45
1  Call Ref: len= 2 (reference 25880/0x6518) (Originator)
1  Message type: SETUP (5)
1  [1 a11 ]
1  Sending Complete (len= 1)
1  [1 041  031  881  901  a61 ]
1  Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer
capability: Unrestricted digital information (8)
1   Ext: 1  Trans mode/rate: 64kbps,
circuit-mode (16)
1   Ext: 1  User information layer 1: G.7xx
384k Video (38)
1  [1 181  031  a11  831  871 ]
1  Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0,
Preferred Dchan: 0
1 ChanSel: Reserved
1Ext: 1  Coding: 0   Number Specified   Channel
Type: 3
1Ext: 1  Channel: 7 ]
1  [1 6c1  0c1  211  831  311  371  351  351  381  361  341  371  381
311 ]
1  Calling Number (len=14) [ Ext: 0  TON: National Number (2)  NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
1Presentation: Presentation allowed of
network provided number (3) '1755864781' ]
1  [1 701  081  c11  351  351  351  361  321  391  371 ]
1  Called Number (len=10) [ Ext: 1  TON: Subscriber Number (4)  NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '5556297' ]
1  [1 7c1  031  881  901  a61 ]
1  Low-layer compatibilty (len= 5) [ 1 0x88 1 0x90 1 0xA6 1  ]
1 -- Making new call for cr 25880
1 -- Processing Q.931 Call Setup
1 -- Processing IE 161 (cs0, Sending Complete)
1 -- Processing IE 4 (cs0, Bearer Capability)
1 -- Processing IE 24 (cs0, Channel Identification)
1 -- Processing IE 108 (cs0, Calling Party Number)
1 -- Processing IE 112 (cs0, Called Party Number)
1 -- Processing IE 124 (cs0, Low-layer Compatibility)
Oct 31 09:05:07 WARNING[6770]: chan_zap.c:8503 zt_pri_error: 1 copying 5
bytes LLC
1  Protocol Discriminator: Q.931 (8)  len=10
1  Call Ref: len= 2 (reference 25880/0x6518) (Terminator)
1  Message type: CALL PROCEEDING (2)
1  [1 181  031  a91  831  871 ]
1  Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0,
Exclusive Dchan: 0
1 ChanSel: Reserved
1Ext: 1  Coding: 0   Number Specified   Channel
Type: 3
1Ext: 1  Channel: 7 ]
-- Accepting data call from '1755864781' to '5556297' on channel
0/7, span 1
-- Executing Macro(Zap/7-1, handle-callerid) in new stack
-- Executing NoOp(Zap/7-1, 01751234567) in new stack
-- Executing GotoIf(Zap/7-1, 0?3:5) in new stack
-- Goto (macro-handle-callerid,s,5)
-- Executing SetCallerID(Zap/7-1, 001751234567) in new stack
-- Executing LookupCIDName(Zap/7-1, ) in new stack
-- Changed Caller*ID name to Testi Tester
-- Executing NoOp(Zap/7-1, Testi Tester 001751234567 / Testi
Tester / 001751234567) in new stack
-- Executing Goto(Zap/7-1, external-call|297|1) in new stack
-- Goto (external-call,297,1)
-- Executing Dial(Zap/7-1, Zap/r1/01752345678) in new stack
1 -- Making new call for cr 32777
-- Requested transfer capability: 0x08 - DIGITAL
1  Protocol Discriminator: Q.931 (8)  len=44
1  Call Ref: len= 2 (reference 9/0x9) (Originator)
1  Message type: SETUP (5)
1  [1 041  021  881  901 ]
1  Bearer Capability (len= 4) [ Ext: 1  Q.931 Std: 0  Info transfer
capability: Unrestricted digital information (8)
1   Ext: 1  Trans mode/rate: 64kbps,
circuit-mode (16)
1   Ext: 0  User information layer 1:
Unknown (24)
1  [1 181  031  a91  831  881 ]
1  Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0,
Exclusive Dchan: 0
1 ChanSel: Reserved
1

[asterisk-users] Re: Forwarding recorded calls to Voicemail

[Tomislav Parčina]

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
 I was wondering if anyone has implemented a feature that would allow a user
 to
 record a phone call and once the call has ended, the call is forwarded to
 his voicemail?

Hi Tom!

I was looking for something like this, but I was unable to find anything 
useful. Hopefully someone will answer your mail. 


--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)270248
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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Re: [asterisk-users] sip realtime broken?

[Don]

not the sip.conf static realtime but sipusers etc?

- Original Message - 
From: Marco Mouta [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: Tuesday, October 31, 2006 7:26 AM
Subject: Re: [asterisk-users] sip realtime broken?



i've been using it sucessfully

On 10/31/06, Don [EMAIL PROTECTED] wrote:



Can someone confirm if sip realtime is broken in 1.2.13 and if so when 
was

the last release it wasn't? heh
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--
Com os melhores cumprimentos,

Marco Mouta
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--
No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.1.408 / Virus Database: 268.13.17/505 - Release Date: 
10/27/2006





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RE: [asterisk-users] light web user interface

[Curt Shaffer]

Basically I would like a page that would
allow a user to log in and modify their extension only. So for example, I log
in for extension 102 once in there I can turn on or off my call waiting. Add a
number to call forward to. Change the email address my voice mail gets sent to.
Add any numbers I may want to block via caller ID. Maybe view my voice mails
that are saved and be able to download them in wav format from there. Add find
me follow me extensions and numbers, etc I would also like it open enough
that I can add features to it. Im not the best at PHP but I can work my
way around in it. I thought maybe freePBX allowed this with its users but I cant
see where you can lock them down to only see information on a particular
extension.











From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Dovid B
Sent: Tuesday, October 31, 2006
3:44 AM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
light web user interface







What attributes are you talking about ? Depending on what
they are it may be real simple to set something up.







- Original Message - 





From: Curt Shaffer 





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





Sent: Monday, October
30, 2006 9:51 PM





Subject: [asterisk-users]
light web user interface









Does anyone know of a really lightweight web interface that
allows users to log in and modify attributes of their extension only?



Thanks



Curt







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Re: [asterisk-users] +Ura +md3200 nao encaminha ligacao

[Bernardo Vieira]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Did you ignore that is an english discussion list or did you just post
on the wrong list altogether?

Anyhoo, try relaxdtmf=yes in your zapata.conf.
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.1 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFFR1P+2QVs8jsa1mQRAnE9AKCNj8pK4EEFx8TWQFuLXXIH+TbRawCfXB/S
7sbNds3FrP8tnNQyb++YbJw=
=w56E
-END PGP SIGNATURE-
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[asterisk-users] SIP RTP flow

[Mike Williams]

Hey,

This is probably a rather stilly question...

If I pick up my SIP phone that's registered to my asterisk server and dial a 
number that asterisk recognises as destined for a SIP trunk (could be a 
static route, or an ENUM lookup) or another SIP device registered on said 
asterisk server (internal extension to extension call), what route does the 
actual audio take?

The control connection (port 5060) obviously goes via the asterisk server as 
it has to work out where to send the control to, but I could quite easily 
imagine the audio going directly handset to remote server or handset to 
asterisk to remote, and handset to handset or handset to asterisk to handset.

Thanks

-- 
Mike Williams
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[asterisk-users] Asterisk dial out (in SIP) to another asterisk context !

[Michel]

Hello list,


We have 2 asterisk servers (without firewall and NAT), 
and We want to do :


From the first server, we have a .call file which dial out to the 
second server. The second server automatically answers and Play a music 
during X seconds, then it hangs up.


Is it possible?


Thanks you!
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[asterisk-users] Asterisk and ARI (Aterisk Recording Interface) integration problem

[Zeeshan Zakaria]

Anybody knows why ARI gives this error message when I enter extension number and password.

Warning: file(/var/spool/asterisk/voicemail/default/222/INBOX/msg.txt): failed to open stream: Permission denied in /var/www/html/recordings/modules/voicemail.module on line 525

It doesn't show the voicemails, although it shows that there is 1 or 2 voicemails in the INBOX.
-- Zeeshan A Zakaria 
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Re: [asterisk-users] Cisco 7960 Skinny calling SIP phone

[Anthony LaMantia]

Which asterisk release are you running chan_skinny under?

- Original Message -
From: Will Roy [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, October 30, 2006 7:52:01 PM GMT-0600 US/Central
Subject: [asterisk-users] Cisco 7960 Skinny calling SIP phone


Before I got down the path of converting a Cisco 7960 I have over to SIP I 
wanted to try and set it up using Skinny. 

The phone registers ok with Asterisk. When I call a SIP softphone extension on 
my network the call is made and I can answering it. However no voice is heard 
over the call. 

When I debug Skinny on the console after the call has connected I see the 
following messag: 

Recieved Alarm Message: DSP Keepalive Timeout [0x3, 0x10, 0x0, 0x7] 

What additional information would be required to troubleshoot this? or should I 
stop wasting time and just convert the phone to SIP? :) 

regards 
Wil 

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Re: [asterisk-users] SIP RTP flow

[Moises Silva]

You can make RTP pass through Asterisk, or not. Look in voip-info.org
about Native Bridge and sip.conf canreinvite option. And may be
this page will be usefull too:

http://www.voip-info.org/wiki/index.php?page=Asterisk+SIP+not-proxy

Regards

On 10/31/06, Mike Williams [EMAIL PROTECTED] wrote:

Hey,

This is probably a rather stilly question...

If I pick up my SIP phone that's registered to my asterisk server and dial a
number that asterisk recognises as destined for a SIP trunk (could be a
static route, or an ENUM lookup) or another SIP device registered on said
asterisk server (internal extension to extension call), what route does the
actual audio take?

The control connection (port 5060) obviously goes via the asterisk server as
it has to work out where to send the control to, but I could quite easily
imagine the audio going directly handset to remote server or handset to
asterisk to remote, and handset to handset or handset to asterisk to handset.

Thanks

--
Mike Williams
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--
Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org;
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Re: [asterisk-users] Bridging Video Calls using Zap

[Steffen Weinreich]

Tristan schrieb:
 Hi,

 You have to modify the setup sent for the call outgoing...
OK, is there a way to this from the dialplan of have i to modify source
for this?

cheeiro
   Steve

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Re: [asterisk-users] Asterisk and ARI (Aterisk Recording Interface) integration problem

[Rodrigo Gonzalez]

User that web server is running has to have read permissions to file 
/var/spool/asterisk/voicemail/default/222/INBOX/msg.txt


Easier option is to run apache as asterisk user

Zeeshan Zakaria wrote:
Anybody knows why ARI gives this error message when I enter extension 
number and password.
 
*Warning*: 
file(/var/spool/asterisk/voicemail/default/222/INBOX/msg.txt): 
failed to open stream: Permission denied in 
*/var/www/html/recordings/modules/voicemail.module* on line *525*
It doesn't show the voicemails, although it shows that there is 1 or 2 
voicemails in the INBOX.


--
Zeeshan A Zakaria


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Re: [asterisk-users] Asterisk and ARI (Aterisk Recording Interface) integration problem

[Guillermo Salas M.]

On Tue, 2006-10-31 at 09:55 -0500, Zeeshan Zakaria wrote:
 Anybody knows why ARI gives this error message when I enter extension
 number and password.
  
 Warning:
 file(/var/spool/asterisk/voicemail/default/222/INBOX/msg.txt):
 failed to open stream: Permission denied
 in /var/www/html/recordings/modules/voicemail.module on line 525
 

Are you sure about the file permissions? The
file /var/spool/asterisk/voicemail/default/222/INBOX/msg
txt must be permissions for the apache user or group.

Try changing the ownership of the file.

Using Debian will be like (apache group is called www-data):

chown
asterisk:www-data /var/spool/asterisk/voicemail/default/222/INBOX/msg

Regards,


 It doesn't show the voicemails, although it shows that there is 1 or 2
 voicemails in the INBOX.
 
 -- 
 Zeeshan A Zakaria 
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-- 
Guillermo Salas M.
Telconet S.A.
Calle 15 y Avenida 24 Esq
Edificio Barre #2 Primer Piso
Telefono : +593 5 262 8071
Celular  : +593 9 985 5138
e-mail   : [EMAIL PROTECTED]
www  : http://www.manta.telconet.net
   http://www.telcocarrier.net

Linux User: 255902

Beat me, whip me, make me use Windows!

Please avoid sending me Word or PowerPoint attachments.
See http://www.fsf.org/philosophy/no-word-attachments.html

Please avoid the Top Posting, see
http://es.wikipedia.org/wiki/Top-posting

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[asterisk-users] Asterisk web interface is not parsing the PHP pages

[Alok Mohapatra]

Hi All,

 I
have installed Asterisk 1.2.10 on Fedora 5. I have installed Asterisk
Management Portal (AMP) for web interface. 

After installing properly when opening in the webpage it is
not parsing the index.php for the AMP. My Database is MySQL.and web server is
Apache 2.2.



Please let me know is this configuration problem or this is
the problem with Apache (Apache 2.2) .





Thanks and Regards

Alok Mohapatra










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Re: [asterisk-users] Asterisk and ARI (Aterisk Recording Interface) integration problem

[Tom Vile]

it told you:Permission DeniedCheck the permissions on that directory.On 10/31/06, Zeeshan Zakaria 
[EMAIL PROTECTED] wrote:Anybody knows why ARI gives this error message when I enter extension number and password.


Warning: file(/var/spool/asterisk/voicemail/default/222/INBOX/msg.txt): failed to open stream: Permission denied in /var/www/html/recordings/modules/voicemail.module on line 525


It doesn't show the voicemails, although it shows that there is 1 or 2 voicemails in the INBOX.
-- Zeeshan A Zakaria 

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http://lists.digium.com/mailman/listinfo/asterisk-users-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephony
www.baldwintechsolutions.comPhone: 518-631-2855 x205Fax: 518-631-2856
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[asterisk-users] Asterisk on virtual machine

[Adam Robins]

We have a 
centralized infrastructure where we deploy Asterisk servers in remote call 
centers for authentication and transcoding. SIP g729a calls are then sent 
over an MPLS VPN to a central Asterisk farm, from which calls 
aresent/received via PRI.

To avoid placing two 
servers in each call center, one for Asterisk and another for Windows AD 
services, we have been playing with VMWare. Can anyone provide their 
experiences in using Asterisk in a VMWare configuration? 
Good/bad/ugly?

Thanks,
Adam
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[asterisk-users] auto recording extensions

[Ed Nuñez]

I would like to know how to record all calls on a queue.  Anu good sugestions?

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Re: [asterisk-users] auto recording extensions

[Dovid B]

If you did make samples you should see an example in queues.conf. By default 
it's commented out.



- Original Message - 
From: Ed Nuñez [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: Tuesday, October 31, 2006 5:31 PM
Subject: [asterisk-users] auto recording extensions


I would like to know how to record all calls on a queue.  Anu good 
sugestions?


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[asterisk-users] Asterisk Call Statistics

[omar parihuana]

Hi Folks,

I would like to recover all information about the calls, incoming
calls, call time, call history, etc in a Web Format,  are  there some
open source aplication for Asterisk that be easier for use. Pls
anything suggestion will be very appreciate.

Thanks

Rgds.
--
Omar E.P.T
-
Certified Networking Professionals make better Connections!

http://omarept.blogspot.com/

 Usysnet Corp
Open Source Solutions
www.usysnet.com.pe
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Re: [asterisk-users] Asterisk web interface is not parsing the PHP pages

[Tom Vile]

Did you install PHP?On 10/31/06, Alok Mohapatra [EMAIL PROTECTED] wrote:













Hi All,

 I
have installed Asterisk 1.2.10 on Fedora 5. I have installed Asterisk
Management Portal (AMP) for web interface. 

After installing properly when opening in the webpage it is
not parsing the index.php for the AMP. My Database is MySQL.and web server is
Apache 2.2.



Please let me know is this configuration problem or this is
the problem with Apache (Apache 2.2) .





Thanks and Regards

Alok Mohapatra











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-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephonywww.baldwintechsolutions.com
Phone: 518-631-2855 x205Fax: 518-631-2856
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Re: [asterisk-users] Audiocodes MP-114 noise

[Jessee J Holmes]

Dear Jason,Please define better noisy? You talking echo issues? Is it on just your side or on the called party's side as well?This start happening immediately, or was the box working before and the problem just started?Also, a quick heads up, make sure before even beginning to troubleshoot an issue like this you do a factory reset to the unit and get the latest available firmware on it. Usually that fixes annoying issues like this.Thanks, Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Oct 30, 2006, at 10:36 PM, Jason Kim wrote:It's noisy while talking.Any idea?Thanks in advance.JasonCheap Talk? Check out Yahoo! Messenger's low PC-to-Phone call rates (http://voice.yahoo.com)___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
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Re: [asterisk-users] Asterisk web interface is not parsing the PHP pages

[Rodrigo Gonzalez]

If it is not parsing the index.php mean that you see the code in your 
browser, install php


Alok Mohapatra wrote:


Hi All,

  I have installed Asterisk 1.2.10 on Fedora 5. I have 
installed Asterisk Management Portal (AMP) for web interface.


After installing properly when opening in the webpage it is not 
parsing the index.php for the AMP. My Database is MySQL.and web server 
is Apache 2.2.


 

Please let me know is this configuration problem or this is the 
problem with Apache (Apache 2.2) .


 


Thanks and Regards

Alok Mohapatra

 




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[asterisk-users] Astricon followup

[Dean Collins]

For the benefit of those outside of the USA or those
unable to make it to Astricon; I wanted to send out this email.





For those of you who attended Astricon in Dallas last week what was
the one thing that you saw that made the trip worthwhile?

(if we post enough information or comments
it will be of benefit for those that didnt attend)







For me personally it was the volume of neat add-on
applications that the Asterisk community are developing; Over time Im
hoping that this leads to something like AppExchange from Salesforce.com were
people can choose from over 300+ applications or addons for SF.



I really want to see more speech recognition applications
but I think its great what Lumen-vox are doing.



Id also like to see someone post some more modified ftp
to text to speech http://www.voip-info.org/wiki/view/asterisk+at+home+festival+weather+configuration

It doesnt need to be weather, how about Oil futures
or wheat prices or score for the weekends games. Any text file accessible by
FTP can be implemented into this script. Id like to see more.



Im hoping that over time we can see even more to the
point that people buy Asterisk just for the applications and we can quote the
same price if not more than cisco because of these addon applications.









Cheers,

Dean

www.Mexuar.com 








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Re: [asterisk-users] Asterisk web interface is not parsing the PHP pages

[Time Bandit]

After installing properly when opening in the webpage it is not parsing the
index.php for the AMP. My Database is MySQL.and web server is Apache 2.2.


Please let me know is this configuration problem or this is the problem with
Apache (Apache 2.2) .

The problem is probably that you didn't install PHP

yum install php
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Re: [asterisk-users] Asterisk Call Statistics

[yusuf]

omar parihuana wrote:

Hi Folks,

I would like to recover all information about the calls, incoming
calls, call time, call history, etc in a Web Format,  are  there some
open source aplication for Asterisk that be easier for use. Pls
anything suggestion will be very appreciate.

Thanks

Rgds.


Hi,

If you have asterisk-addons, you can get all CDR's, which include all the above statistics, written 
to a MySQL or PGSQL database.  It would then be very easy to get this on to a web page.


--
thanks,
yusuf

--
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[asterisk-users] channel.c: Unable to request channel ZAP

[Asterisk]

Hi All,



I have one rather
annoying problem...my PBX can work great for weeks, when suddenly I start
receiving these messages when I try to make a zaptel call:



Oct 31 13:52:47
NOTICE[15636] app_dial.c: Unable to create channel of type 'ZAP' (cause 34 -
Circuit/channel congestion)

Oct 31 13:52:49
NOTICE[15648] channel.c: Unable to request channel ZAP/g1/247



I'm using Sangoma A104
card (with four E1 spans), and these problems are only occurring on the first
two spans (which are connected to a legacy PBX)  the second two spans, which
are connected to the Telco, work perfectly. Even more: when these messages
start to occur, I can hardly initiate any call via problematic two spans (1st and
2nd), where I can with no problem initiate a new call thru the unproblematic
two spans (3rd and 4th).



Restart of the Asterisk
is the only cure so far



Does anyone know what
could possibly be the cause, or how could I troubleshot this problem?



Regards.

Alex






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re: [asterisk-users] Live creation of trunk groups

[Andre Courchesne - Consultant]

Well it works. 


If I have group=0 that includes all my channels, I can create group=1 which is 
a subset and a simple reload makes this g1 available to dial on that subset.


Message: 12
Date: Mon, 30 Oct 2006 15:25:06 -0700
From: Alyed Tzompa [EMAIL PROTECTED]
Subject: re: [asterisk-users] Live creation of trunk groups
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=iso-8859-1


My advice is to first make some tests to see if a reload is 
enough for Asterisk to read any group definitions change in zapata.conf, 
otherwise no on-the-fly change will work

Alyed  




Return-Path: [EMAIL PROTECTED] Mon Oct 30 13:23:36 2006
Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by 
maila11.webcontrolcenter.com with SMTP;
Mon, 30 Oct 2006 13:23:36 -0700

Hi,

Is there a way to create trunk groups while asterisk is running.

For exemple let's say that zapata.conf defines g0 as channels 1-23

I would like (while asterisk is running) define g1 as 1-10 and g1 as 10-23

Any hints appreciated.

Andre Courchesne
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Re: [asterisk-users] IVR

[Tzafrir Cohen]

On Mon, Oct 30, 2006 at 06:54:40PM -0500, Vitalie Apostu wrote:
 Greetings,
 
 If somebody knows how to concatenate several .gsm files in one  or create a
 macro and use with background() please reply.

As simple as:

  cat file1.gsm file2.gsm  both.gsm

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Asterisk web interface is not parsing the PHP pages

[Alex Robar]

Alok,

Two things: 

1) You said you installed AMP. AMP has ceased development a while ago, but is survived by the FreePBX project. If you actually installed AMP and not FreePBX, I would suggest you get FreePBX running first. A lot of effort went into improving FreePBX from AMP.


2) You typically won't find much help for the GUIs from this list because the GUIs have their own mailing lists and forums. Try posting your question to FreePBX.org. You're more likely to get a response there.


Alex
On 10/31/06, Alok Mohapatra [EMAIL PROTECTED] wrote:



Hi All,
 I have installed Asterisk 1.2.10 on Fedora 5. I have installed Asterisk Management Portal (AMP) for web interface. 

After installing properly when opening in the webpage it is not parsing the index.php for the AMP. My Database is MySQL.and web server is Apache 
2.2.

Please let me know is this configuration problem or this is the problem with Apache (Apache 2.2) .


Thanks and Regards
Alok Mohapatra
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RE: [asterisk-users] Asterisk web interface is not parsing the PHPpages

[Jordan Kirby]

Possibly a silly question, but do you have php installed 
and configured in apache?


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Alok 
MohapatraSent: 31 October 2006 15:45To: 
asterisk-users@lists.digium.comSubject: [asterisk-users] Asterisk web 
interface is not parsing the PHPpages 


Hi All,
 
I have installed Asterisk 1.2.10 on Fedora 5. I have installed Asterisk 
Management Portal (AMP) for web interface. 
After installing properly when 
opening in the webpage it is not parsing the index.php for the AMP. My Database 
is MySQL.and web server is Apache 2.2.

Please let me know is this 
configuration problem or this is the problem with Apache (Apache 2.2) 
.


Thanks and 
Regards
Alok 
Mohapatra

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Re: [asterisk-users] Asterisk on virtual machine

[Andrew Latham]

What about the AD in the VM. or running Open LDAP on the Asterisk server.

On 10/31/06, Adam Robins [EMAIL PROTECTED] wrote:





We have a centralized infrastructure where we deploy Asterisk servers in
remote call centers for authentication and transcoding.  SIP g729a calls are
then sent over an MPLS VPN to a central Asterisk farm, from which calls are
sent/received via PRI.



To avoid placing two servers in each call center, one for Asterisk and
another for Windows AD services, we have been playing with VMWare.  Can
anyone provide their experiences in using Asterisk in a VMWare
configuration?  Good/bad/ugly?



Thanks,

Adam
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--
---
Andrew Latham - AKA: LATHAMA (lay-th-ham-eh)
[EMAIL PROTECTED] - [EMAIL PROTECTED]
If any of the above are down we have bigger problems than my email!
Hind sight is most always 20/20 or better.
---
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Re: [asterisk-users] Asterisk web interface is not parsing the PHP pages

[Giorgio Incantalupo]

Hi Alok,
it seems like libapache2-mod-php is missing in your linux box.
Have you tried to make a simple index.php file to test?


Giorgio Incantalupo


Alok Mohapatra wrote:


Hi All,

  I have installed Asterisk 1.2.10 on Fedora 5. I have 
installed Asterisk Management Portal (AMP) for web interface.


After installing properly when opening in the webpage it is not 
parsing the index.php for the AMP. My Database is MySQL.and web server 
is Apache 2.2.


 

Please let me know is this configuration problem or this is the 
problem with Apache (Apache 2.2) .


 


Thanks and Regards

Alok Mohapatra

 




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Re: [asterisk-users] Asterisk and ARI (Aterisk Recording Interface) integration problem

[Zeeshan Zakaria]

Everytime a voicemail is recorded, a .txt file is generated. It was working fine before and permissions were automatically set. On my home server it is working perfectly fine. This is another server, with the same settings, and all of a sudden today it has started to give this error. Voicemails etc recorded yesterday are all fine, no problem with permissions. I don't remember changing anything on the server today which could have started giving this error.
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[asterisk-users] Snom or Cisco Phones?

[Joao Pereira]

Hello
I need to buy IP Phones to work with Asterisk, and I'm in doubt between 
Snom and Cisco Phones.
Can you gurus, please, give me your impression of these 2 brands? I need 
to focus more in SIP and Asterisk compatibility and less in pricing 
(yes, I know the Cisco are more expensive).
Are there any features that Snom has, that Cisco doesnt? And are these 
features important?

Thanks

Joao Pereira

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[asterisk-users] Example Polycom function key config

[Jamie Heckford]

Hi,

Has anyone here reprogrammed their Polycom features keys using
sip/ipmid.cfg?

If so I would be really grateful if someone could send me an example as
I have tried various entries for hours now and don't seem to be getting
anywhere.

Any help appreciated. 


Kind regards

Jamie Heckford
Technical Consultant
  

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[asterisk-users] simultaneous ring - call groups or queues or something else?

[Stephen Bosch]

Hi, folks:

I need to be able to have a single DID ring multiple remote (IP and
PSTN) extensions, and then pass the call to whichever picks up first.
I'm sure this is old hat -- lots of providers offer it.

I see that Trixbox will do it, but it's not clear how it's doing it.
They use different terminology -- a ring group and hunt strategy

How can it be done with a straight Asterisk server?

Thanks for the help!

-Stephen-
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RE: [asterisk-users] Asterisk on virtual machine

[Ryan Amos]

Asterisk does not work very well in a VM
due to the timeslicing. Dropped calls, jittery audio and echo can all creep in.



Good news is that an AD controller runs
just fine in VMware. Just make sure the box has enough RAM to keep it happy,
and use a physical second disk for the Windows install. So Id suggest
running Asterisk in Linux as the native OS, and running VMware with Windows Server
as a guest OS. This setup should work just fine for you.











From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam Robins
Sent: Tuesday, October 31, 2006
9:53 AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [asterisk-users] Asterisk
on virtual machine





We have a centralized infrastructure where
we deploy Asterisk servers in remote call centers for authentication and
transcoding. SIP g729a calls are then sent over an MPLS VPN to a central
Asterisk farm, from which calls aresent/received via PRI.



To avoid placing two servers in each call
center, one for Asterisk and another for Windows AD services, we have been
playing with VMWare. Can anyone provide their experiences in using Asterisk in
a VMWare configuration? Good/bad/ugly?



Thanks,

Adam






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Re: [asterisk-users] Astricon followup

[Dal]

Hello All,

This is a great list post, I have blogged about it 
here: http://www.asteriskvoipnews.com/asterisk_news/astricon_2006_followup.html

It would be great if people could post there 
response on this post along with the list. I love reading answers to 
questions like this. Thanks,

-Dal

  - Original Message - 
  From: 
  Dean Collins 
  
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Tuesday, October 31, 2006 8:44 
  AM
  Subject: [asterisk-users] Astricon 
  followup
  
  
  For the benefit of those outside 
  of the USA or those unable to make it to 
  Astricon; I wanted to send out this email.
  
  
  For those of 
  you who attended Astricon in Dallas last week what was the one thing that 
  you saw that made the trip worthwhile?
  (if we post 
  enough information or comments it will be of benefit for those that didn’t 
  attend)
  
  
  
  For me personally it was the 
  volume of neat add-on applications that the Asterisk community are developing; 
  Over time I’m hoping that this leads to something like AppExchange from 
  Salesforce.com were people can choose from over 300+ applications or addons 
  for SF.
  
  I really want to see more speech 
  recognition applications but I think it’s great what Lumen-vox are 
  doing.
  
  I’d also like to see someone post 
  some more modified “ftp to text to speech” http://www.voip-info.org/wiki/view/asterisk+at+home+festival+weather+configuration
  It doesn’t need to be weather, how 
  about Oil futures or wheat prices or score for the weekends games. Any text 
  file accessible by FTP can be implemented into this script. I’d like to see 
  more.
  
  I’m hoping that over time we can 
  see even more to the point that people buy Asterisk just for the applications 
  and we can quote the same price if not more than cisco because of these addon 
  applications.
  
  
  
  
  Cheers,
  Dean
  www.Mexuar.com 
  
  
  

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[asterisk-users] overlap of zap trunk groups

[Damon Estep]

Can anyone suggest any reasons why a zap (PRI) b channel
should not be a member of multiple zap trunk group definitions?



For example;



Group 1 = Channels 1 to 23

Group 2 = channels 1 to 12

Group 3 = channels 13 to 23



The purpose is to restrict the number of channels a
particular extensions can use, but use the entire span for other extensions.






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[asterisk-users] FXO Cards vs. Channel bank with T1

[Dovid B]

Is there any advantage of getting a T1 card with a 
channel bank over 2-3 FXO cards ?
Thanks.

Dovid
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Re: [asterisk-users] Asterisk both behind a NAT and outside at the same time

[Brad Templeton]

On Tue, Oct 31, 2006 at 07:40:35PM +0800, Leo Ann Boon wrote:
   
 Have you tried setting the externalip and localnet parameters?
 
Localnet makes some sense, and is set (should be the default anyway, no?)

externalip, as I understand it, is for an Asterisk which is behind
a NAT.  This asterisk is not behind a NAT to anybody.  The
phones are behind a NAT to the outside world but not to the
Asterisk box, which has two ethernets on it, one for the internal
natwork and one for the real internet.

It uses bindaddr=0.0.0.0 and listens to both addresses.  


 Sorry for my previous post I misunderstood the problem.
 You should set canreinvite=no to all sip peers that connect from outside.


That's precisely what I don't want to do.  This would block native
bridging in the one case where it's most important.


The correct behaviour, as I see it is:

a) Native bridge when connecting two external channels -- everybody is on 
the real internet
b) Native bridge when connecting two internal channels -- everybody is on 
the 192.168.* network
c) Route RTP through Asterisk when connecting internal and external
d) When a channel is to a device behind a remote NAT, the usual rules apply
   (either use STUN or other smart NAT, or route RTP through Asterisk)

The super correct behaviour, which I don't expect but would be nice is

e) Clever native bridge between internal and external by being aware that 
the device
   talks to the outside world using a different address than it talks to 
you.
   (Possibly if the phones use STUN they will tell Asterisk their external 
IP, which
   is not the same as Asterisk's though it's on the same subnet)



I have used localnet=192.168.* and nat=yes on a local device and it still
attempts an incorrect native bridge between internal and external, with
one-way audio.

If I do canreinvite=no on the local devices then it works of course, but
now means the local phones will never native bridge amongst themselves.
In a larger network, that would be a problem, and it's a poor result in any
network.

This is the latest svn of 1.2, by the way.
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RE: [asterisk-users] Snom or Cisco Phones?

[Ejay Hire]

Cisco Cisco or Linksys Cisco?

Cisco Cisco, I'd prefer the Snom.  Linksys Cisco, it's a tossup.

I've worked with dozens of the Cisco 7960 phones, 25 of the Linksys, and 3
Snom.

My specific issues with the Cisco included poor echo cancellation, problems
with nat traversal, and no web interface.  I didn't like any of the default
ringers on the Snom phones, but the users really liked the LED call
appearance lights compared to the 7960 LCD.  I have no complaints about the
Linksys phones.

Ejay Hire

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joao Pereira
Sent: Tuesday, October 31, 2006 11:21 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Snom or Cisco Phones?

Hello
I need to buy IP Phones to work with Asterisk, and I'm in doubt between Snom
and Cisco Phones.
Can you gurus, please, give me your impression of these 2 brands? I need to
focus more in SIP and Asterisk compatibility and less in pricing (yes, I
know the Cisco are more expensive).
Are there any features that Snom has, that Cisco doesnt? And are these
features important?
Thanks

Joao Pereira

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Re: [asterisk-users] Server Recommendations

[Carlos Rojas]

Hello,

I'm working with supermicro servers, for the irq problems with Dell, any people have problems

Regards
On 10/30/06, Paul Hales [EMAIL PROTECTED] wrote:
How many analog lines are you looking at? Hundreds?PaulHOn Mon, 2006-10-30 at 17:22 -0600, Joe Dennick wrote:
 We have a number of clients who will be needing a server to host Asterisk on.Many of these clients use analog (FXO) lines that will need to be connected to Asterisk via Sangoma cards.Can anyone
 recommend an industry-standard server (like IBM, Dell, HP, etc.) that has enough open PCI slots to handle up to six of the Sangoma cards?We would like to be able to tell the customer to just go purchase this
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Re: [asterisk-users] simultaneous ring - call groups or queues or something else?

[Brian Rogan]

You can just seperate multiple phones with  in the Dial command,
as the voip-info wiki page shows:

http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial

On Tue, Oct 31, 2006 at 10:28:32AM -0700, Stephen Bosch wrote:
 Hi, folks:
 
 I need to be able to have a single DID ring multiple remote (IP and
 PSTN) extensions, and then pass the call to whichever picks up first.
 I'm sure this is old hat -- lots of providers offer it.
 
 I see that Trixbox will do it, but it's not clear how it's doing it.
 They use different terminology -- a ring group and hunt strategy
 
 How can it be done with a straight Asterisk server?
 
 Thanks for the help!
 
 -Stephen-
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[asterisk-users] Strange Characters in CLI on TTY9

[Forrest Beck]

When I look at TTY9 (using init.d and safe_asterisk to start the
asterisk process), I am getting some strange characters.  When a
application is run the and the CLI shows the application executing the
languange almost looks russian...??

Anyone seen this before?
http://picasaweb.google.com/jonforrest.beck/AsteriskCLI
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[asterisk-users] DTMF Tones

[Jason Walker]

I have tried beta2, beta3 and now back to 1.2.12.1 and I have correct 
DTMF tones 25% of the time.  I have to call several times to enter an 
extension.  I have a router and a packet shaper and some other stuff. 
Anyone have any other ideas why this might happen.  I do not have any 
Zap channels but I am running CentOS4. I also do not have any cards 
installed. Thanks


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Re: [asterisk-users] Registration problem

[sergio . dippolito]

firewall? i dont think so because sometimes the phone can register ok  
and sudendly the appears unregistered


Leonardo Silva [EMAIL PROTECTED] ha escrito:


2006/10/31, Jon Farmer [EMAIL PROTECTED]:




Sergio R. D'Ippolito wrote:

Hi all, i have an * version: Asterisk SVN-branch-1.2-r45691, I need to
register a linksys 922 phone thru internet and when I make sip debug
command i see this debug information:



*/SIP/2.0 401 Unauthorized/*

/Via: SIP/2.0/UDP x.x.x.x:1025;branch=z9hG4bK-43bf8123;received=x.x.x.x/

/From: SPA922 sip:[EMAIL PROTECTED];tag=685bbad1fae3325do0/

/To: SPA922 sip:[EMAIL PROTECTED];tag=as4da6f6ce/

/Call-ID: [EMAIL PROTECTED]/

/CSeq: 5503 REGISTER/

/User-Agent: incore-PBX/

/Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY/

/WWW-Authenticate: Digest algorithm=MD5, realm=asterisk,

nonce=372b2479/

Asterisk is asking the phone to resend the registration with
WWW-Authenticate using MD5 hash. Make sure the phone supports this and
retry. Or you could turn this option off in the sip.conf.

Regards

Jon

--
Jon Farmer
Telford, Shropshire, UK
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Maybe a Firewall ?

--
Leonardo Silva
fone: 16 8143-1146




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Re: [asterisk-users] Snom or Cisco Phones?

[Joe Dennick]

Comparing Snom to Cisco phones is sort of like comparing Mercedes to Kia 
cars


Joao Pereira wrote:


Hello
I need to buy IP Phones to work with Asterisk, and I'm in doubt 
between Snom and Cisco Phones.
Can you gurus, please, give me your impression of these 2 brands? I 
need to focus more in SIP and Asterisk compatibility and less in 
pricing (yes, I know the Cisco are more expensive).
Are there any features that Snom has, that Cisco doesnt? And are these 
features important?

Thanks

Joao Pereira

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Re: [asterisk-users] FXO Cards vs. Channel bank with T1

[John Novack]

YES!

Many machines do NOT work well with multiple analog cards. Especially 
the Digium ones.
Channel banks with FXO circuits are harder to come by on the used 
market, though

Many all FXS channel banks can be had used, though.

If you want multiple FXO's and do not want to go the T1 route, look 
towards the Sangoma A200


John Novack


Dovid B wrote:
Is there any advantage of getting a T1 card with a channel bank over 
2-3 FXO cards ?

Thanks.
 
Dovid



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Re: [asterisk-users] FXO Cards vs. Channel bank with T1

[Eric \ManxPower\ Wieling]

Dovid B wrote:

Is there any advantage of getting a T1 card with a channel bank over 2-3 FXO 
cards ?
Thanks.


In my experience a T-1 port w/channel bank just works better.  The more 
cards you use, the more interrupts are generated.


My standard configuration for analog FXS ports is a T-1 card (Digium or 
Sangoma) and an Adtran TA750 Channel Bank.  The Adtrans can be found 
very cheap on eBay.  FXO ports tend to be much expensive, but you can 
find them on eBay as well.


Why not just get a PRI or channelized voice T-1?
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Re: [asterisk-users] overlap of zap trunk groups

[Eric \ManxPower\ Wieling]

Damon Estep wrote:

Can anyone suggest any reasons why a zap (PRI) b channel should not be a
member of multiple zap trunk group definitions?

 


For example;

 


Group 1 = Channels 1 to 23

Group 2 = channels 1 to 12

Group 3 = channels 13 to 23

 


The purpose is to restrict the number of channels a particular
extensions can use, but use the entire span for other extensions.


Part of a production /etc/asterisk/zaptel.conf:

group=1
channel = 1-6
group=1,2
channel = 7-12
group=0
channel = 13-16

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Re: [asterisk-users] FXO Cards vs. Channel bank with T1

[Jay R. Ashworth]

On Tue, Oct 31, 2006 at 08:20:57PM +0200, Dovid B wrote:
Is there any advantage of getting a T1 card with a channel bank
over 2-3 FXO cards ?

If you need enough ports to make a T-1 card cost-efficient, then you
might oughtta be looking at an Ethernet to FXO media gateway instead --
assuming you need analog interfaces.  FXO side, why not just go T-1 or
PRI?

Cheers,
-- jra
-- 
Jay R. Ashworth[EMAIL PROTECTED]
Designer  Baylink RFC 2100
Ashworth  AssociatesThe Things I Think'87 e24
St Petersburg FL USA  http://baylink.pitas.com +1 727 647 1274

That's women for you; you divorce them, and 10 years later,
  they stop having sex with you.  -- Jennifer Crusie; _Fast_Women_
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AW: [asterisk-users] Snom or Cisco Phones?

[Christian Stredicke]

I think one of the differences is: We do pay attention to Asterisk and this 
mailing list ;-)

CS 

-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Joao Pereira
Gesendet: Dienstag, 31. Oktober 2006 13:47
An: asterisk-users@lists.digium.com
Betreff: [asterisk-users] Snom or Cisco Phones?

Hello
I need to buy IP Phones to work with Asterisk, and I'm in doubt between Snom 
and Cisco Phones.
Can you gurus, please, give me your impression of these 2 brands? I need to 
focus more in SIP and Asterisk compatibility and less in pricing (yes, I know 
the Cisco are more expensive).
Are there any features that Snom has, that Cisco doesnt? And are these features 
important?
Thanks

Joao Pereira

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Re: [asterisk-users] channel.c: Avoided initial deadlock

[Tim Panton]

On 26 Oct 2006, at 13:25, asterisk wrote:


Hi all,

Can tell me somebody what meen : channel.c: Avoided initial deadlock

Our customer makes calls with our softphone (with IAX2).
Sometimes the softphon freezes. The call is ACTIVE but the user  
cant hang it up.

At this time in the log file (asterisk/messages) appear the next line:
channel.c: Avoided initial deadlock.

we use: SVN-branch-1.2-r46176M
with VoIP channel (ADSL)

Can you help me? What is the problem?


If you can send us either the output of
iax2 debug or an ethereal trace of the packets in
a conversation that fails I'll take a look.

At a guess your softphone has a bug, and asterisk is just issuing a  
warning,

but I don't have enough evidence yet.


Tim Panton

www.mexuar.com



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Re: [asterisk-users] Grandstream ATA 286 tdm400 and Asterisk 1.2-13

[Tzafrir Cohen]

On Mon, Oct 30, 2006 at 11:17:52AM -0500, Erick Perez wrote:
 Hi people,
 
 I would like to read your suggestions as to where the issue might be.
 ATA286=Grandstream Budgetone ATA 286 in SIP mode. One Lan Port and one FXS 
 port.
 TDM04B= 4 FXO signal fxls
 There is a 8FXO-to-SIP unit in this scenario that works perfectly so i
 will not make mention of it.
 
 PSTNVOIPprovider---Internet---ATA286--tdm04b---Asterisk1.2.-13

What exactly is the point is such settings? Why not connect directly to
the provider over SIP? Or to the ATA over SIP?

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Live creation of trunk groups

[Tzafrir Cohen]

On Mon, Oct 30, 2006 at 03:08:39PM -0500, Andre Courchesne - Consultant wrote:
 Hi,
 
  Is there a way to create trunk groups while asterisk is running.
 
  For exemple let's say that zapata.conf defines g0 as channels 1-23
 
  I would like (while asterisk is running) define g1 as 1-10 and g1 as 10-23
 
  Any hints appreciated.

Edit zapata.conf and from the asterisk cli run 'reload' or 'reload
chan_zap.so' . This will apply most changes from apata.conf. Basically
anything that doesn't change the very nature of the channel. Tat is:
you will not be able to create and destory channels that way, or even
change their signalling. But you'll be able to change probably all other
parameters.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] simultaneous ring - call groups or queues or something else?

[Stephen Bosch]

Brian Rogan wrote:
 You can just seperate multiple phones with  in the Dial command,
 as the voip-info wiki page shows:
 
 http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial

Thanks! It's not always clear where to look first for these things.

I'm repeatedly blown away by the ease of configuration and flexibility
of Asterisk.

-Stephen-

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Re: [asterisk-users] Cheapest way to determine channels in a group from outside asterisk?

[Tim Panton]

On 26 Oct 2006, at 12:12, Nick Adams wrote:

I need to determine the number of active calls in a group from  
outside of Asterisk. Currently I poll the manager API and parse the  
channel status list but this is becoming too expensive on CPU.


What are my options? What is considered standard practice ?  
Update a DB field? Poll the manager api? Use an asterisk -rv 'some  
command' call?


That depends on your configuration. If you already use SNMP in your  
organisation, you

might want to use that.

If you are/have a java coder, there is some support for the asterisk  
MIB in the free-ware

from snmp.westhawk.co.uk

(Disclaimer - I wrote large chunks of it so I'm biased :-) )


Tim Panton

www.mexuar.com



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Re: [asterisk-users] IPv6

[Tim Panton]

On 26 Oct 2006, at 15:33, David Bandel wrote:


Folks,

Anyone know if Asterisk supports IPv6?  If not, is support planned?


There was a talk at astricon on this. (I think the slides will be  
available

from astricon.net).

The short answer is no, not yet, but folks are working on it.

Tim.

Tim Panton

www.mexuar.com



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[asterisk-users] FXO Card's vs. T1

[Dovid B]

Is there any advantage of getting a T1 card with a 
channel bank over 2-3 FXO cards ?
Thanks.

Dovid
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Re: [asterisk-users] Asterisk architecure

[G(P)L]

jez . a écrit :

Dear all,

I've recently installed Asterisk and am trying to understand where 
exactly Asterisk 'fits' in my VOIP architecture. Can/does Asterisk work 
as a proxy. I am specifically interested in SIP. Could anyone perhaps 
point me out to a diagram with SIP users and Asterisk to better 
understand how I should set up my network?


Thank you



Hi,

You can find some interesting diagram here :
http://www.tech-invite.com/Ti-sip-dialog.html

Other diagrams more architecture ortiented :
http://lehmann.free.fr/divers/SIP%20tutorial.pdf
slides 32 and after.
The document is not mine :)

If you want something more specific to Asterisk's architecture, I 
recommand you this book : 
http://www.eyrolles.com/Informatique/Livre/9780596009625/livre-asterisk.php


Bye
Guillaume Lehmann

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RE: [asterisk-users] overlap of zap trunk groups

[Damon Estep]

 
 Damon Estep wrote:
  Can anyone suggest any reasons why a zap (PRI) b channel should not
be a
  member of multiple zap trunk group definitions?
 
 
 
  For example;
 
 
 
  Group 1 = Channels 1 to 23
 
  Group 2 = channels 1 to 12
 
  Group 3 = channels 13 to 23
 
 
 
  The purpose is to restrict the number of channels a particular
  extensions can use, but use the entire span for other extensions.
 
 Part of a production /etc/asterisk/zaptel.conf:
 
 group=1
 channel = 1-6
 group=1,2
 channel = 7-12
 group=0
 channel = 13-16
 

So the correct solution is to define the channel only once, but the
group= parameter can contain many groups delimited by a comma, correct?
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[asterisk-users] Re: DTMF Tones

[Nick Adams]

Jason Walker wrote:
I have tried beta2, beta3 and now back to 1.2.12.1 and I have correct 
DTMF tones 25% of the time.  I have to call several times to enter an 
extension.  I have a router and a packet shaper and some other stuff. 
Anyone have any other ideas why this might happen.  I do not have any 
Zap channels but I am running CentOS4. I also do not have any cards 
installed. Thanks


What phones and codec are you using?

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Re: [asterisk-users] Strange Characters in CLI on TTY9

[Tzafrir Cohen]

On Tue, Oct 31, 2006 at 02:03:33PM -0500, Forrest Beck wrote:
 When I look at TTY9 (using init.d and safe_asterisk to start the
 asterisk process), I am getting some strange characters.  When a
 application is run the and the CLI shows the application executing the
 languange almost looks russian...??
 
 Anyone seen this before?
 http://picasaweb.google.com/jonforrest.beck/AsteriskCLI

Bogus terminal settings show color as cyrillic. vim with syntax
hilighting will probably give you a similar result.

Consult your distro's gurus. Some relevant keyfors: consolechars ,
setfont 

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Snom or Cisco Phones?

[Mike Dent]

On 10/31/06, Joao Pereira [EMAIL PROTECTED] wrote:

Hello
I need to buy IP Phones to work with Asterisk, and I'm in doubt between
Snom and Cisco Phones.
Can you gurus, please, give me your impression of these 2 brands? I need
to focus more in SIP and Asterisk compatibility and less in pricing
(yes, I know the Cisco are more expensive).
Are there any features that Snom has, that Cisco doesnt? And are these
features important?
Thanks

Joao Pereira



I have a Cisco 7960 here in the home office, I recenty purchased a
Snom 300 for the lounge. I wrote a very quick mini-review on my blog:-

http://www.g6phf.co.uk/site/2006/10/05/snom-300-voip-phone-mini-review/

Christian @ Snom, whilst I have your 'ear' here :) Please can you add
a backlight to future revisions of the Snom 300, it would be most
welcome!!

thanks
Mike
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Re: [asterisk-users] Server Recommendations

[Conrad Wood]

On Tue, 2006-10-31 at 13:52 -0500, Carlos Rojas wrote:
 Hello,
  
 I'm working with supermicro servers, for the irq problems with Dell,
 any people have problems
  

I second the supermicro servers - particularly the opteron range based on 
Serverworks HS1000  chipset.
Excellent stuff. Well designed, no irq problems and no timing problems.

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Re: AW: [asterisk-users] Snom or Cisco Phones?

[Aaron Daniel]

That and Cisco won't give you the time of day if you don't use their
stuff ;)

We have about 1600 of the Cisco's on campus, and unless you run them on
the call manager, you're not gonna have nearly as many features as any
other phone that's designed with SIP in mind.  That said, if you need a
phone with dialtone, a pretty screen, and limited xml services, then I
will say that the cisco's are extremely easy to provision once you
figure out the upgrade paths.

(Oh, and we're running 7940's and 7960's... if you're looking at the
7912's, etc, good luck, they're a _complete_ pain to work with)

Aaron

On Tue, 2006-10-31 at 20:41 +0100, Christian Stredicke wrote:
 I think one of the differences is: We do pay attention to Asterisk and this 
 mailing list ;-)
 
 CS 
 
 -Ursprüngliche Nachricht-
 Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Joao Pereira
 Gesendet: Dienstag, 31. Oktober 2006 13:47
 An: asterisk-users@lists.digium.com
 Betreff: [asterisk-users] Snom or Cisco Phones?
 
 Hello
 I need to buy IP Phones to work with Asterisk, and I'm in doubt between Snom 
 and Cisco Phones.
 Can you gurus, please, give me your impression of these 2 brands? I need to 
 focus more in SIP and Asterisk compatibility and less in pricing (yes, I know 
 the Cisco are more expensive).
 Are there any features that Snom has, that Cisco doesnt? And are these 
 features important?
 Thanks
 
 Joao Pereira
 
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-- 
Aaron Daniel
Senior Voice Analyst
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198

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Re: [asterisk-users] Snom or Cisco Phones?

[Conrad Wood]

On Tue, 2006-10-31 at 13:29 -0600, Joe Dennick wrote:
 Comparing Snom to Cisco phones is sort of like comparing Mercedes to Kia 
 cars

Not really. Both are very good phones. 

* My Clients prefer cisco because it looks more business-like. - The new
snom phones do look better though and the side car rules.
* The Cisco phone 'feels' very good in your hand, and the voicequality
is superb. (I'd say slightly better than that of the snom 360)

* Technically, I find the snom phone more advanced and I can do more
cool stuff with it - Cisco doesn't seem to like giving features away in
SIP.
* Snom phones, for example, have freely programmable buttons that can
park/retrieve/transfer calls, show line status etc. I can't get that to
work with Cisco phones at all.
* Putting custom ringtones (and choosing which ones to use) is a
no-brainer with snoms and real trouble with ciscos.
* On ciscos, I find the upgrade path from sccp to sip a totally
unnecessary annoyance.



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[asterisk-users] compilation problem with asterisk-addons

[Erick Perez]

Hi,

Trying to compile asterisk-addons 1.2.5 on Centos 4.4 produces this:

Note: MySQL libraries are installed and the structure is as follows:
/usr/src/astsources/asterisk-1.2.13
/usr/src/astsources/asterisk-addons-1.2.5

in /usr/src/astsources/asterisk-addons-1.2.5 I do:
make clean
make

and the output is:

./mkdep -fPIC -I../asterisk -D_GNU_SOURCE  -I/usr/include/mysql   `ls *.c`
app_addon_sql_mysql.c:25:27: asterisk/file.h: No such file or directory
app_addon_sql_mysql.c:26:29: asterisk/logger.h: No such file or directory
app_addon_sql_mysql.c:27:30: asterisk/channel.h: No such file or directory
app_addon_sql_mysql.c:28:26: asterisk/pbx.h: No such file or directory
app_addon_sql_mysql.c:29:29: asterisk/module.h: No such file or directory
app_addon_sql_mysql.c:30:34: asterisk/linkedlists.h: No such file or directory
app_addon_sql_mysql.c:31:31: asterisk/chanvars.h: No such file or directory
app_addon_sql_mysql.c:32:27: asterisk/lock.h: No such file or directory
app_saycountpl.c:11:27: asterisk/file.h: No such file or directory
app_saycountpl.c:12:29: asterisk/logger.h: No such file or directory
app_saycountpl.c:13:30: asterisk/channel.h: No such file or directory
app_saycountpl.c:14:26: asterisk/pbx.h: No such file or directory
app_saycountpl.c:15:29: asterisk/module.h: No such file or directory
app_saycountpl.c:16:27: asterisk/lock.h: No such file or directory
cdr_addon_mysql.c:23:29: asterisk/config.h: No such file or directory
cdr_addon_mysql.c:24:30: asterisk/options.h: No such file or directory
cdr_addon_mysql.c:25:30: asterisk/channel.h: No such file or directory
cdr_addon_mysql.c:26:26: asterisk/cdr.h: No such file or directory
cdr_addon_mysql.c:27:29: asterisk/module.h: No such file or directory
cdr_addon_mysql.c:28:29: asterisk/logger.h: No such file or directory
cdr_addon_mysql.c:29:26: asterisk/cli.h: No such file or directory
res_config_mysql.c:41:30: asterisk/channel.h: No such file or directory
res_config_mysql.c:42:29: asterisk/logger.h: No such file or directory
res_config_mysql.c:43:29: asterisk/config.h: No such file or directory
res_config_mysql.c:44:29: asterisk/module.h: No such file or directory
res_config_mysql.c:45:27: asterisk/lock.h: No such file or directory
res_config_mysql.c:46:30: asterisk/options.h: No such file or directory
res_config_mysql.c:47:26: asterisk/cli.h: No such file or directory
res_config_mysql.c:48:28: asterisk/utils.h: No such file or directory
make -C format_mp3 all
make[1]: Entering directory
`/usr/src/astsources/asterisk-addons-1.2.5/format_mp3'
gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations   -D_REENTRANT -D_GNU_SOURCE  -O6-c -o
common.o common.c
common.c:1:29: asterisk/logger.h: No such file or directory
common.c: In function `decode_header':
common.c:93: warning: implicit declaration of function `ast_log'
common.c:93: error: `LOG_WARNING' undeclared (first use in this function)
common.c:93: error: (Each undeclared identifier is reported only once
common.c:93: error: for each function it appears in.)
make[1]: *** [common.o] Error 1
make[1]: Leaving directory
`/usr/src/astsources/asterisk-addons-1.2.5/format_mp3'
make: *** [format_mp3/format_mp3.so] Error 2


Thanks for your help.


--

Erick Perez
Panama Sistemas
Integradores de Telefonia IP y Soluciones Para Centros de Datos
Panama, Republica de Panama
Cel Panama. +(507) 6694-4780

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Re: [asterisk-users] Asterisk both behind a NAT and outside at the same time

[C F]

On 10/31/06, Brad Templeton [EMAIL PROTECTED] wrote:

On Tue, Oct 31, 2006 at 07:40:35PM +0800, Leo Ann Boon wrote:
 
 Have you tried setting the externalip and localnet parameters?

Localnet makes some sense, and is set (should be the default anyway, no?)

externalip, as I understand it, is for an Asterisk which is behind
a NAT.  This asterisk is not behind a NAT to anybody.  The
phones are behind a NAT to the outside world but not to the
Asterisk box, which has two ethernets on it, one for the internal
natwork and one for the real internet.

It uses bindaddr=0.0.0.0 and listens to both addresses.


 Sorry for my previous post I misunderstood the problem.
 You should set canreinvite=no to all sip peers that connect from outside.


That's precisely what I don't want to do.  This would block native
bridging in the one case where it's most important.


The correct behaviour, as I see it is:

a) Native bridge when connecting two external channels -- everybody is on 
the real internet


It might not work if one of them is NATed. Therefore the correct way
to do this is to use canreinvite=no


b) Native bridge when connecting two internal channels -- everybody is on 
the 192.168.* network


canreinvite=yes will take care of this.


c) Route RTP through Asterisk when connecting internal and external


Again by adding canreinvite=no to externals you have this.


d) When a channel is to a device behind a remote NAT, the usual rules apply
   (either use STUN or other smart NAT, or route RTP through Asterisk)


How will asterisk know? The correct *setting* (not behavior) is
canreinvite=no for the external devices.



The super correct behaviour, which I don't expect but would be nice is

e) Clever native bridge between internal and external by being aware that 
the device
   talks to the outside world using a different address than it talks to 
you.
   (Possibly if the phones use STUN they will tell Asterisk their external 
IP, which
   is not the same as Asterisk's though it's on the same subnet)



I have used localnet=192.168.* and nat=yes on a local device and it still
attempts an incorrect native bridge between internal and external, with
one-way audio.

If I do canreinvite=no on the local devices then it works of course, but
now means the local phones will never native bridge amongst themselves.
In a larger network, that would be a problem, and it's a poor result in any
network.



Why are you so against having the RTP go thru asterisk?
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Re: [asterisk-users] Re: IAX2 show peers - description

[Marian Rychtecky]

Hi friends,
 thank you for comments...

Marian

Tomislav Parčina napsal(a):

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...

Hi,

I think the (T) is for Trunk.

Regards
Fred


Hi Fred!

I believe that T is for trunk. Thank you.


--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)270248
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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--
Marian Rychtecky
[EMAIL PROTECTED]

Tel. +420 724 397 441
ICQ 76582857
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Re: [asterisk-users] Grandstream ATA 286 tdm400 and Asterisk 1.2-13

[Erick Perez]

I forgot to mention that the Carrier that owns the ATA box was not
willing to let me connect directly over IP, I was only allowed to use
the FXS port. He already ack that he has a problem with
disconnections.


On 10/31/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:

On Mon, Oct 30, 2006 at 11:17:52AM -0500, Erick Perez wrote:
 Hi people,

 I would like to read your suggestions as to where the issue might be.
 ATA286=Grandstream Budgetone ATA 286 in SIP mode. One Lan Port and one FXS
 port.
 TDM04B= 4 FXO signal fxls
 There is a 8FXO-to-SIP unit in this scenario that works perfectly so i
 will not make mention of it.

 PSTNVOIPprovider---Internet---ATA286--tdm04b---Asterisk1.2.-13

What exactly is the point is such settings? Why not connect directly to
the provider over SIP? Or to the ATA over SIP?

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

Erick Perez
Panama Sistemas
Integradores de Telefonia IP y Soluciones Para Centros de Datos
Panama, Republica de Panama
Cel Panama. +(507) 6694-4780

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Re: [asterisk-users] FXO Cards vs. Channel bank with T1

[Dovid B]

Looking at the number's now it seems that a T1 will be more.
Anyone here sell PRI's ?

- Original Message - 
From: Jay R. Ashworth [EMAIL PROTECTED]

To: asterisk-users@lists.digium.com
Sent: Tuesday, October 31, 2006 9:38 PM
Subject: Re: [asterisk-users] FXO Cards vs. Channel bank with T1



On Tue, Oct 31, 2006 at 08:20:57PM +0200, Dovid B wrote:

   Is there any advantage of getting a T1 card with a channel bank
   over 2-3 FXO cards ?


If you need enough ports to make a T-1 card cost-efficient, then you
might oughtta be looking at an Ethernet to FXO media gateway instead --
assuming you need analog interfaces.  FXO side, why not just go T-1 or
PRI?

Cheers,
-- jra
--
Jay R. Ashworth 
[EMAIL PROTECTED]
Designer  Baylink RFC 
2100
Ashworth  AssociatesThe Things I Think'87 
e24
St Petersburg FL USA  http://baylink.pitas.com +1 727 647 
1274


That's women for you; you divorce them, and 10 years later,
  they stop having sex with you.  -- Jennifer Crusie; _Fast_Women_
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Re: [asterisk-users] overlap of zap trunk groups

[Eric \ManxPower\ Wieling]

Damon Estep wrote:

Damon Estep wrote:

Can anyone suggest any reasons why a zap (PRI) b channel should not

be a

member of multiple zap trunk group definitions?



For example;



Group 1 = Channels 1 to 23

Group 2 = channels 1 to 12

Group 3 = channels 13 to 23



The purpose is to restrict the number of channels a particular
extensions can use, but use the entire span for other extensions.

Part of a production /etc/asterisk/zaptel.conf:

group=1
channel = 1-6
group=1,2
channel = 7-12
group=0
channel = 13-16



So the correct solution is to define the channel only once, but the
group= parameter can contain many groups delimited by a comma, correct?


Correct.
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[asterisk-users] Compatability

[Joel Hill]

Hi All,

I have a new client who has an existing  Asterisk  PABX and is looking 
for us to install a TE110P for him, However he has a Dell SC420 and I 
have never used one before.
I have had no problems with any other Dell servers which we use almost 
exclusively.


Has anyone had any good/bad experiences with the SC420 in relation with 
Digium cards?


Thanks for your help.

Joel
Asterisk IT
www.asteriskit.com.au
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Re: [asterisk-users] Asterisk Call Statistics

[Andrew Joakimsen]

Check out voip-info.org, there are quite a few GUIS some even generate nice graphs!On 10/31/06, omar parihuana 
[EMAIL PROTECTED] wrote:Hi Folks,I would like to recover all information about the calls, incoming
calls, call time, call history, etc in a Web Format,arethere someopen source aplication for Asterisk that be easier for use. Plsanything suggestion will be very appreciate.ThanksRgds.--
Omar E.P.T-Certified Networking Professionals make better Connections!http://omarept.blogspot.com/Usysnet CorpOpen Source Solutions
www.usysnet.com.pe___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list
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Re: [asterisk-users] DTMF Tones

[Andrew Joakimsen]

Where are these DTMF tones going? From where? Be specific, post the relevant config file sections I can't read minds and I'd be surprised if 0.1% of the people who read this can either
On 10/31/06, Jason Walker [EMAIL PROTECTED] wrote:
I have tried beta2, beta3 and now back to 1.2.12.1 and I have correctDTMF tones 25% of the time.I have to call several times to enter anextension.I have a router and a packet shaper and some other stuff.
Anyone have any other ideas why this might happen.I do not have anyZap channels but I am running CentOS4. I also do not have any cardsinstalled. Thanks___
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RE: [asterisk-users] Server Recommendations

[shadowym]

I'm working with Supermicro as well. 

-Original Message-
From: Carlos Rojas [mailto:[EMAIL PROTECTED] 
Sent: Tuesday, October 31, 2006 10:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Server Recommendations

Hello,
 
I'm working with supermicro servers, for the irq problems with Dell, any
people have problems
 
Regards

 
On 10/30/06, Paul Hales [EMAIL PROTECTED] wrote: 


How many analog lines are you looking at? Hundreds?

PaulH

On Mon, 2006-10-30 at 17:22 -0600, Joe Dennick wrote: 
 We have a number of clients who will be needing a server to host
 Asterisk on.  Many of these clients use analog (FXO) lines that
will
 need to be connected to Asterisk via Sangoma cards.  Can anyone 
 recommend an industry-standard server (like IBM, Dell, HP, etc.)
that
 has enough open PCI slots to handle up to six of the Sangoma
cards?  We
 would like to be able to tell the customer to just go purchase
this 
 model server from this manufacturer and it will work.
Suggestions?

 Thank you!

 Joe Dennick
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Re: [asterisk-users] Re: [OT] wi-fi ip phone scenario

[Andrew Joakimsen]

I've done extensive testing, WDS is just as reliable as wired,however at first we had issues with some AP that would not respond and needed to be rebooted. But if its possible to wire the AP you should since WDS will eat alot of bandwidth and also decrease the range since most the AP will have to be within range of eachother, way more than overlapping coverage
Alberto:I would suggest you try to keep all the AP on the same channel. With that large of a space I wouldnt expect too much interferance from the outside.On 10/28/06, 
Martin Joseph [EMAIL PROTECTED] wrote:
On 2006-10-27 11:55:14 -0700, Andrew Joakimsen [EMAIL PROTECTED] said: Are you using WDS? While it won't totally fix every issue, I've found in my
 trials that turning off WDS and making sure all the AP were connected to the same wired network was way more reliable, no more random unregistartion and issue with registering (still seems to unregister at times, but
 re-registartion won't require a reboot).I think it's cleary true that wiring WIFI infrastructure is easier andmore reliable then WDS.On the other hand,I have been running my little network with WDS for
over three weeks now, and it has been completely reliable.The tricks where to configure things properly and to have the basescloser together then one would think would be needed.Once this was setup. It works, and it keeps working.We had a couple
of stress tests also, one black out and one unplugged router(carpenter).Came up cleanly and continued working fine.No mis-registrations andno problems.Marty___
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Re: [asterisk-users] Asterisk and Panasonic KX Model

[C F]

I did this today with a Panasonic KX-TD1232 and a Digium TDM2401E
Card. I hope to put it on the wiki soon, if you need help just tell me
with what.



On 10/30/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:

If Someone did that, How I connect extensions.conf with this type of Hybrid
system to work with asterisk inside this schema:

PSTN---PANASONIC KX -- Asterisk
|
|-send internal call


Thanks.

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