[asterisk-users] rtp connection remained when call busy using agi for call control

[CHEN XUEQIN]

Hi there:

I compiled asterisk 1.6.2.6 and install it on my Ubuntu
OS. I run a perl fastagi for call control. Two SIP ua register
  asterisk, here as UA_1, UA_2. The problem can be reproduced
in following step.

1) SIP UA_1 and UA_2 successfully register to asterisk

2) UA_1 call UA_2, send invite to asterisk

3) asterisk execute a dialpan, as following

  exten => _X.,1,Agi(agi://127.0.0.1/dial_handler)
  exten => _x.,1,Hangup

  dial_handler is a perl FastAGI server, it just query
  some variable, call SetAMAFlags application, finally
  call Dial application for invite UA_2

 3) UA_2 ring, but reject the call by send 486 busy

 4) AGI call hangup then exit

 5) call end, use netstat find two udp port used by asterisk
like these

netstat -anup
udp0  0 0.0.0.0:14634   0.0.0.0:*   
4219/asterisk
udp0  0 0.0.0.0:14635   0.0.0.0:*   
4219/asterisk

It will remain forever, unless restart asterisk.

For information,

 1) UA_1 call UA_2 directly by dial application, rtp connection will be
 destroyed normal.
 2) UA_1 call UA_2 also by AGI, if UA_2 answer, then hangup, rtp
  connection will also be destroyed normal.

Why the rtp connection remained when UA_2 reject the call? If I use
asterisk for many calls, the rtp connection resource will be occupied
to much. Anything wrong in my AGI program or dialplan ?

Thanks in advance.


Regards,

Chen






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Re: [asterisk-users] Asterisk running on a Xen Centos Server challenge!!!

[Warren Selby]

On Thu, Mar 18, 2010 at 6:26 PM, Tzafrir Cohen wrote:

> > After you install the kernel source, you'll need to rerun ./configure.
>
> Nope. The dahdi-linux makefile has no ./configure .
>
> >
> > You may want to run "make clean" and / or "make distclean" before
> rerunning
> > ./configure.
>
> Nope.
>
>
Oops.  :)  My mistake.

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[asterisk-users] how to configure caller id

[cool dude]

hello i had configured a Dial plan in which i am using application time base i.e

1 - if  call comes to PSTN line    from 8pm  evening till morning 8am the call 
should  automatically forward to guard exten i.e exten 211, and if guard dosent 
receive call in 30 secs message should be saved in voicemail.

2 - if call comes in working hours than it should be received by ext 112 n from 
there using transfer (tT)application  call is tranfered to desired extensions.

now i want when i call from my mobile to pstn line my mobile no should be 
displayed in softphone


###


vi extensions.conf



[from-zaptel]
exten => s,1,Wait(2)
exten => s,n,GotoIfTime(20:00-7:59|mon-sun|*|*?closed,s,1)
exten => s,n,Dial(SIP/112,5,tT)
exten => s,n,Goto(mainmenu,s,1)


[my-phones]
exten => 112,1,Dial(SIP/112,5,T)
exten => 113,1,Dial(SIP/113)
exten => 114,1,Dial(SIP/114)

[mainmenu]
exten => s,1,Answer
exten => s,n,Noop(CALLERID(name))
exten => s,n,SetMusicOnHold(default)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Background(enter-ext-of-person)
exten => s,n,WaitExten(5)
exten => _11[2-4],1,Goto(my-phones,${EXTEN},1)
exten => i,1,Playback(pbx-invalid)
exten => t,1,Playback(vm-goodbye)


[closed]
exten => s,1,Dial(SIP/211,30)
exten => s,n,VoiceMail(211,u)
exten => 2999,1,VoiceMailMain(${CALLERID(num)},s) ; by 2999 voicemail can be 
heard.

#



;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit
;autogenrated on 2010-03-18
;Dahdi Channels Configurations
;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak

[trunkgroups]

[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no


;Sangoma AU100 [slot:0 bus: span:1]  
context=from-zaptel
group=0
echocancel=yes
signalling = fxs_ks
channel => 1

context=from-zaptel
group=0
echocancel=yes
signalling = fxs_ks
channel => 2



any help n support will be highly appreciated.
thx



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Re: [asterisk-users] Live Audio Streaming- From Aux interface-Online resource

[ABBAS SHAKEEL]

Thanks I will look into it.

On Fri, Mar 19, 2010 at 2:26 AM, Philipp von Klitzing <
klitz...@pool.informatik.rwth-aachen.de> wrote:

> > I would like to know if any one have experience with live audio
> > streaming like 1. Streaming from an online resource
>
> Look at app_ices and icecast.
> http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Ices
>
> Philipp
>
>
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Re: [asterisk-users] confbridge not working?

[Magnus Benngård]

Hi! 

Did a quick test, worked as a clock: 

exten => 0317998959,1,Set(CHANNEL(language)=se)
exten => 0317998959,n,Answer()
exten => 0317998959,n,ConfBridge(1001,s)  0317998959,n,Hangup() 

On Thu, 18 Mar 2010 20:20:35 -0700, Kelvin Chan  wrote:  

Hi guys,

I'm trying to move away from meetme to loose the dependency on dahdi. 
ConfBridge seems to be a good fit but I can't get it going. The document 
sounds like an easy to use app. Am I missing any bridge_ modules?

Asterisk 1.6.2.0~rc2-0ubuntu1.2

 -- Executing [...@outbound:1] Answer("SIP/109-b877a8c8", "") in new 
stack
 -- Executing [...@outbound:2] ConfBridge("SIP/109-b877a8c8", 
"conf") in new stack
[Mar 19 03:16:33] ERROR[2294]: app_confbridge.c:434 
join_conference_bridge: Conference bridge '521' could not be created.

dial plan:

exten => _52X,1,Answer()
exten => _52X,n,ConfBridge(${EXTEN})

Thanks,

Kelvin

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Re: [asterisk-users] confbridge not working?

[Alex Balashov]

What does the source code tell you about the circumstances in which  
that particular error string is produced?

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On Mar 18, 2010, at 11:20 PM, Kelvin Chan   
wrote:

> Hi guys,
>
> I'm trying to move away from meetme to loose the dependency on dahdi.
> ConfBridge seems to be a good fit but I can't get it going. The  
> document
> sounds like an easy to use app. Am I missing any bridge_ modules?
>
>
>
> Asterisk 1.6.2.0~rc2-0ubuntu1.2
>
> -- Executing [...@outbound:1] Answer("SIP/109-b877a8c8", "") in  
> new
> stack
> -- Executing [...@outbound:2] ConfBridge("SIP/109-b877a8c8",
> "conf") in new stack
> [Mar 19 03:16:33] ERROR[2294]: app_confbridge.c:434
> join_conference_bridge: Conference bridge '521' could not be created.
>
>
>
> dial plan:
>
> exten => _52X,1,Answer()
> exten => _52X,n,ConfBridge(${EXTEN})
>
> Thanks,
>
> Kelvin
>
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Re: [asterisk-users] How to detect a PSTN telephone is busy or not?

[Zhang Shukun]

Thanks!  but if  i use Queue to call out not Dial.

how should i know the status like busy or free?

for now . i know asterisk have QUEUESTATUS variable，

QUEUESTATUS   The status of the call as a text string, one of TIMEOUT
| FULL | JOINEMPTY | LEAVEEMPTY | JOINUNAVAIL | LEAVEUNAVAIL

but the variable have no busy or free status?  how to  know the
numbers in the queue is busy or not at present?

Need your help. thanks!

2010/3/18 ABBAS SHAKEEL :
> Hello,
> Please have a look to DIALSTATUS variable.
> here http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS
> I hope it helps
>
>
> On Thu, Mar 18, 2010 at 1:31 PM, Zhang Shukun  wrote:
>>
>> hi,all
>>
>> one problem confuse me these days. i want to sequence dial three PSTN
>> number(a,b,c)
>>
>> first, if i dial number a, if a is busy , i will dial number b. if b
>> is busy, i will dial number c.
>>
>> Dial(SIP/a...@ip,30)
>> Dial(SIP/b...@ip,30)
>> Dial(SIP/c...@ip,30)
>>
>> i want to know before i dial number a, how to know if a is busy now?
>>
>> if a is busy now. i will not dial a, instead, i will dial number b
>> directly.
>>
>> to summary is : in asterisk, how to detect a pstn telephone number is
>> busy or not before dialing it?
>>
>> Thanks!
>>
>> --
>> Best regards,
>> Sucan
>>
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> --
> Best Regards
> Shakeel Abbas
>
>
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-- 
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Sucan

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[asterisk-users] confbridge not working?

[Kelvin Chan]

Hi guys,

I'm trying to move away from meetme to loose the dependency on dahdi. 
ConfBridge seems to be a good fit but I can't get it going. The document 
sounds like an easy to use app. Am I missing any bridge_ modules?



Asterisk 1.6.2.0~rc2-0ubuntu1.2

 -- Executing [...@outbound:1] Answer("SIP/109-b877a8c8", "") in new 
stack
 -- Executing [...@outbound:2] ConfBridge("SIP/109-b877a8c8", 
"conf") in new stack
[Mar 19 03:16:33] ERROR[2294]: app_confbridge.c:434 
join_conference_bridge: Conference bridge '521' could not be created.



dial plan:

exten => _52X,1,Answer()
exten => _52X,n,ConfBridge(${EXTEN})

Thanks,

Kelvin

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Re: [asterisk-users] Better SIP security please! Was: (no subject)

[Zeeshan Zakaria]

Philipp, remembering sip user agent is a wondeful idea, and if you goggle
it, somebody had made a patch for it, so that one could identify sip devices
by their sip user agent names. Surprisingly the decision makers didn't like
to put it in the production branch of asterisk at that time, however it is
still avialble online somewhere as a patch for older releases of asterisk. I
came across it when hackers where attacking my server on constant basis. I
however ended up writing a security code within the dialplan to catch the
sip user agent fields and ip addresses and compare them with info in the
actual user database, which worked good for me. Here the only problem could
be with change of sip user agent info, e.g. x-lite puts version number in
sip user agent field, which changes as you upgrade it to newer versions. A
relatively more complicated code probably will however recognize it. And a
hacker can always send a fake sip user agent field if he is really desparate
to hack your server, which can also be caught using fail2ban.

On 2010-03-18 10:45 PM, "Philipp von Klitzing" <
klitz...@pool.informatik.rwth-aachen.de> wrote:

Hey hey!

> > My first step will be to strengthen the passwords in use, and for the
> > hardphones to restrict by IP address, but that still leaves the
> > softphone quite widely open.
>
> Asterisk doesn't differentiate between a hard phone and a soft phone.

Although: One could think about enhancing Asterisk security by allowing
only a (number of) specific SIP user agent header (vendor, model) for a
SIP account - next to a strong password, of course. Or implement
something more dynamic like: Read and lock the current (or first) user
agent string, and then ping the admin if that changes and request an un-
lock/re-auth.

> > Does Asterisk 1.6 have anything in it that can automatically block out
> > an attacking IP, say if it receives several 20 or so failed attempts
> > from that IP in x minutes?

It would still be important to have a sip.conf paramter in 1.4 that is
similar to "delayreject" in iax.conf! One of my system has been scanned
3 times in the past days, and it takes just a little over a minute for a
10.000 account registration scan.

Philipp


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Re: [asterisk-users] Better SIP security please! Was: (no subject)

[Philipp von Klitzing]

Hey hey!

> > My first step will be to strengthen the passwords in use, and for the
> > hardphones to restrict by IP address, but that still leaves the
> > softphone quite widely open.
> 
> Asterisk doesn't differentiate between a hard phone and a soft phone.

Although: One could think about enhancing Asterisk security by allowing 
only a (number of) specific SIP user agent header (vendor, model) for a 
SIP account - next to a strong password, of course. Or implement 
something more dynamic like: Read and lock the current (or first) user 
agent string, and then ping the admin if that changes and request an un-
lock/re-auth.

> > Does Asterisk 1.6 have anything in it that can automatically block out
> > an attacking IP, say if it receives several 20 or so failed attempts
> > from that IP in x minutes?

It would still be important to have a sip.conf paramter in 1.4 that is 
similar to "delayreject" in iax.conf! One of my system has been scanned 
3 times in the past days, and it takes just a little over a minute for a 
10.000 account registration scan.

Philipp


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Re: [asterisk-users] Define an array of sip number in sip.conf

[Zeeshan Zakaria]

You'll have to type them all in manually. Or do what I did several times,
write a script in php which will generate the sip.conf with that many
extensions. Even better look into using realtime architecture, where you can
quickly generate as many extensions as you like.

On 2010-03-18 10:09 PM, "huu giang"  wrote:

Hi List,

How can I define an array of sip number in sip.conf ?
I want to define an array of sip number from 1000 to 2000, so I can make a
performance test on Asterisk using sipp.

Thanks in Advance,
Giangnh


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[asterisk-users] Define an array of sip number in sip.conf

[huu giang]

Hi List,

How can I define an array of sip number in sip.conf ?
I want to define an array of sip number from 1000 to 2000, so I can make a 
performance test on Asterisk using sipp.

Thanks in Advance,
Giangnh



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Re: [asterisk-users] (no subject)

[Zeeshan Zakaria]

Fail2ban is a must. I was a victim of such attacks, and have implemented
some other measures too, but fail2ban is a must have with the link posted by
Matt which describes how to set it up for asterisk. Make sure you put your
own ip address in ignore list otherwise it can block you too.

On 2010-03-18 8:45 PM, "Matt Riddell"  wrote:

On 19/03/10 1:19 PM, Adrian Marsh wrote:
> Hello,
>
> I’m looking for some advice on securing Asteri...
Have a look at fail2ban:

http://www.voip-info.org/wiki/view/Fail2Ban+%28with+iptables%29+And+Asterisk

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Re: [asterisk-users] (no subject)

[Steve Edwards]

On Fri, 19 Mar 2010, Adrian Marsh wrote:


I’m looking for some advice on securing Asterisk.

My first step will be to strengthen the passwords in use, and for the 
hardphones to restrict by IP address, but that still leaves the 
softphone quite widely open.


Asterisk doesn't differentiate between a hard phone and a soft phone. You 
can restrict by IP address for soft phones as well.


Does Asterisk 1.6 have anything in it that can automatically block out 
an attacking IP, say if it receives several 20 or so failed attempts 
from that IP in x minutes?


I'm a 1.2 Luddite, so I can't speak for 1.6.

I think any "brute force" or DOS security policy needs to be implemented 
external to Asterisk. I don't think there are any AMI events you could 
listen to. I think you are limited to what you can scrounge out of a log 
file.


How about setting up a couple of "honey-pot" SIP accounts with obvious 
passwords and in the context fire off a user event? Then you could listen 
for the event via AMI.



Any other suggestions?


Repost with a meaningful subject -- a blank subject labels you as a newbie 
who is probably not worth the time of members with relevant experience.


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Re: [asterisk-users] (no subject)

[Matt Riddell]

On 19/03/10 1:19 PM, Adrian Marsh wrote:
> Hello,
>
> I’m looking for some advice on securing Asterisk.

Have a look at fail2ban:

http://www.voip-info.org/wiki/view/Fail2Ban+%28with+iptables%29+And+Asterisk

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Managing Director
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[asterisk-users] (no subject)

[Adrian Marsh]

Hello,

 

I'm looking for some advice on securing Asterisk.

Recently my servers been under several brute-force SIP attacks.

 

I have several remote sites, as well as many roaming users, who may have
PC softclients and/or SIP based hardphones.

 

My first step will be to strengthen the passwords in use, and for the
hardphones to restrict by IP address, but that still leaves the
softphone quite widely open.

 

Does Asterisk 1.6 have anything in it that can automatically block out
an attacking IP, say if it receives several 20 or so failed attempts
from that IP in x minutes?

 

I haven't looked at Secure SIP in quite a while, is that now integrated
into 1.6 ?

 

One thing that's confusing me in my config,  is that I thought that if I
set NAT=no in sip.conf, then I wouldn't be able to connect to that SIP
account unless I was on the local LAN, specified by locallan=   However
in some testing, I'm finding that I can still connect from an external
SIP client.

 

Also, I tried setting one SIP account from host=dynamic to
host=, and when that client tried to register, then Asterisk
complained that the account wasn't supposed to be trying to register.

 

My next step is also to upgrade my Asterisk itself up to the latest
stable 1.6

 

Any other suggestions?

 

Thanks,

 

Adrian

 

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Re: [asterisk-users] Asterisk running on a Xen Centos Server challenge!!!

[Tzafrir Cohen]

On Thu, Mar 18, 2010 at 05:03:12PM -0500, Warren Selby wrote:
> On Thu, Mar 18, 2010 at 6:56 PM, Daniel Leite de Abreu 
> wrote:
> 
> > Hi David!
> >
> >
> > Thanks very much for helping me out will all !
> >
> >
> > Ok i try your tip and @ the moment i still have the same problem but now i
> > have the kernel and the kernel devel the same but wend i try to run make i
> > still get the same erro, do you guys have any idea how to fix it?
> >
> > -bash-3.2# rpm -qa kernel*
> > kernel-xen-devel-2.6.18-164.6.1.el5
> > kernel-xen-2.6.18-164.6.1.el5
> > -bash-3.2#
> >
> 
> After you install the kernel source, you'll need to rerun ./configure.

Nope. The dahdi-linux makefile has no ./configure .

> 
> You may want to run "make clean" and / or "make distclean" before rerunning
> ./configure.

Nope.

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Re: [asterisk-users] Asterisk running on a Xen Centos Server challenge!!!

[Warren Selby]

On Thu, Mar 18, 2010 at 6:56 PM, Daniel Leite de Abreu wrote:

> Hi David!
>
>
> Thanks very much for helping me out will all !
>
>
> Ok i try your tip and @ the moment i still have the same problem but now i
> have the kernel and the kernel devel the same but wend i try to run make i
> still get the same erro, do you guys have any idea how to fix it?
>
> -bash-3.2# rpm -qa kernel*
> kernel-xen-devel-2.6.18-164.6.1.el5
> kernel-xen-2.6.18-164.6.1.el5
> -bash-3.2#
>

After you install the kernel source, you'll need to rerun ./configure.

You may want to run "make clean" and / or "make distclean" before rerunning
./configure.

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Re: [asterisk-users] Free Daily Asterisk News iPhone and iPod Touch app

[Zeeshan Zakaria]

On 2010-03-18 5:31 PM, "Matt Riddell"  wrote:

Hi all,

I've released another free app for the iPhone and iPod touch - this one
lets you read the Daily Asterisk News.

Hope you enjoy it :D

http://www.venturevoip.com/news.php?rssid=2371

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Re: [asterisk-users] Free Daily Asterisk News iPhone and iPod Touch app

[Zeeshan Zakaria]

Thanks Matt. This should be useful. I'll give it a try on my Motorola
Droid/Milestone.

On 2010-03-18 5:31 PM, "Matt Riddell"  wrote:

Hi all,

I've released another free app for the iPhone and iPod touch - this one
lets you read the Daily Asterisk News.

Hope you enjoy it :D

http://www.venturevoip.com/news.php?rssid=2371

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Re: [asterisk-users] Asterisk running on a Xen Centos Server challenge!!!

[Daniel Leite de Abreu]

Hi David!


Thanks very much for helping me out will all !


Ok i try your tip and @ the moment i still have the same problem but now i have 
the kernel and the kernel devel the same but wend i try to run make i still get 
the same erro, do you guys have any idea how to fix it?

-bash-3.2# rpm -qa kernel*
kernel-xen-devel-2.6.18-164.6.1.el5
kernel-xen-2.6.18-164.6.1.el5
-bash-3.2# 

Please!


Thanks very much.


Daniel Abreu.
On 17 Mar 2010, at 12:29 PM, David Backeberg wrote:

> On Wed, Mar 17, 2010 at 10:16 AM, Daniel Leite de Abreu
>  wrote:
>> -bash-3.2# cd dahdi-linux-complete-2.2.1+2.2.1/
>> -bash-3.2# make all
>> make -C linux all
>> make[1]: Entering directory 
>> `/usr/src/asterisk/dahdi-linux-complete-2.2.1+2.2.1/linux'
>> make -C drivers/dahdi/firmware firmware-loaders
>> make[2]: Entering directory 
>> `/usr/src/asterisk/dahdi-linux-complete-2.2.1+2.2.1/linux/drivers/dahdi/firmware'
>> make[2]: Leaving directory 
>> `/usr/src/asterisk/dahdi-linux-complete-2.2.1+2.2.1/linux/drivers/dahdi/firmware'
>> You do not appear to have the sources for the 2.6.18-164.6.1.el5xen kernel 
>> installed.
>> make[1]: *** [modules] Error 1
>> make[1]: Leaving directory 
>> `/usr/src/asterisk/dahdi-linux-complete-2.2.1+2.2.1/linux'
>> make: *** [all] Error 2
>> 
>> 
>> This error tells me that i don't have the sources for the kernel 
>> 2.6.18-164.6.1.el5xen , so how can i find it?
> 
> http://wiki.centos.org/HowTos/I_need_the_Kernel_Source
> 
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Re: [asterisk-users] queue MOH

[Matt Riddell]

On 15/03/10 11:23 AM, Thomas Perron wrote:
> I want callers to enter a queue and then hear music on hold.
> does anyone have notes on how to integrate queuing to a dial plan that uses 
> moh?

You can just set the music on hold class for the Queue in queues.conf - 
you actually have to provide an option (r IIRC) to provide ringing 
instead of music on hold.

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Managing Director
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Re: [asterisk-users] Live Audio Streaming- From Aux interface-Online resource

[Philipp von Klitzing]

> I would like to know if any one have experience with live audio
> streaming like 1. Streaming from an online resource

Look at app_ices and icecast.
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Ices

Philipp


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[asterisk-users] Free Daily Asterisk News iPhone and iPod Touch app

[Matt Riddell]

Hi all,

I've released another free app for the iPhone and iPod touch - this one 
lets you read the Daily Asterisk News.

Hope you enjoy it :D

http://www.venturevoip.com/news.php?rssid=2371

-- 
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Matt Riddell
Managing Director
___

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http://www.venturevoip.com/exchange.php (Full ITSP Solution)
http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)

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Re: [asterisk-users] SPA3102 5.1.7 Firmware (codec bug in 5.1.10 ?)

[Joseph]

On 03/18/10 16:22, Sebastian Milioto wrote:
>Somebody has 5.1.7 firmware for SPA3102?
>I'm having issues with inbound/outbound calls using asterisk through SPA3102
>with firmware 5.1.10. I've read it has a codec bug, since it doesn't care
>about what you set up in Preferred Codec.
>
>Any help will be appreciated.
>
>Sebastian

You will find it here:
http://prov.802.cz/fw/

Ever since the Linksys took over from Sipura and now by Cisco, thoese devices 
are of very poor quality.
Two of SPA3102 died on me within two years, in addition lots of echo impossible 
to eliminate.

I've switched/replaced my Linksys/Sipura units with Audiocodes MP-114 but they 
are not perfect either.
Though, I can say they don't have/generate any echo problems and fixes go 
through without any problem (which I can not say the same about Linksys/Sipura 
units.) 

-- 
Joseph

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[asterisk-users] SPA3102 5.1.7 Firmware (codec bug in 5.1.10 ?)

[Sebastian Milioto]

Somebody has 5.1.7 firmware for SPA3102?
I'm having issues with inbound/outbound calls using asterisk through SPA3102
with firmware 5.1.10. I've read it has a codec bug, since it doesn't care
about what you set up in Preferred Codec.

Any help will be appreciated.

Sebastian
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Re: [asterisk-users] Asterisk DIES with no trace. PLEASE

[Zeeshan Zakaria]

E1 channels are also zap channels. Zap show channels doesn't differentiate
between them.

On 2010-03-18 2:05 PM, "Danny Dias"  wrote:

I'm not having problems with hanging up the calls, my problems is that i
asterisk dies, i'm using a pri and always zap show channel X will always
show Hookstate (FXS only): "Onhook" beacuse it only applies to FXS and i'm
using digital e1 trunk

Or am i wrong?


> Message: 1
> Date: Thu, 18 Mar 2010 11:20:38 -0400
>
>
> > From: Zeeshan Zakaria 
> > Subject: Re: [asterisk-users] Asterisk DIES with no ...
>
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> ><5ad99e891003180820l56882922gd84102d158918...@mail.gmail.com>
>
>
> > Content-Type: text/plain; charset="iso-8859-1"
> >
>
> > Do you properly hang up the calls. Does 'zap show channel  number>'
> > shows that the chann...
>
> > On 2010-03-18 10:06 AM, "Danny Dias"  wrote:
> >
> > Thanks Zeeshan,
> >
> > SAng...
>

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Re: [asterisk-users] Asterisk DIES with no trace. PLEASE

[Danny Dias]

I'm not having problems with hanging up the calls, my problems is that i
asterisk dies, i'm using a pri and always zap show channel X will always
show Hookstate (FXS only): "Onhook" beacuse it only applies to FXS and i'm
using digital e1 trunk

Or am i wrong?


> Message: 1
> Date: Thu, 18 Mar 2010 11:20:38 -0400
> From: Zeeshan Zakaria 
> Subject: Re: [asterisk-users] Asterisk DIES with no trace. PLEASE
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Message-ID:
><5ad99e891003180820l56882922gd84102d158918...@mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Do you properly hang up the calls. Does 'zap show channel '
> shows that the channel is 'on hook' after its hang up?
>
> On 2010-03-18 10:06 AM, "Danny Dias"  wrote:
>
> Thanks Zeeshan,
>
> SAngoma told me that the asterisk problem is unrelated to wanpipe drivers,
> they told me to reinstall asterisk again
>
> But, i still having doubts about the problem :(
>
> Thanks in advance
>
>
>
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Re: [asterisk-users] Polycom DHCP Option 66 FTP provisioning

[Mike Diehl]

On Thursday 18 March 2010 11:24:18 am Karl Fife wrote:
> - Original Message -
> From: "Lee, John (Sydney)" 
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Sent: Wednesday, March 17, 2010 9:50 PM
> Subject: Re: [asterisk-users] Polycom DHCP Option 66 FTP provisioning
>
> >> I'll see if E4Strategies can open a support ticket at Polycom.
> >> They're really good about stuff like that.  I'll let you know either
> >
> > way.
> >
> > What is E4Strategies?
> > Polycom support is hopeless in Oz.  They just shove you to some
> > distributer who only knows to replace your hardware.
>
> e4strategies is the where I bought all of my Polycom hardware.
>
> It would seem that e4 moves enough Polycom hardware so P pays attention to
> e4 when they open a support ticket, [which is why] I (and others) buy
> Polycom from e4, which is why P pays attention to e4 , which is why others
> buy from e4...  Loop: goto [which is why] :-)
>
> Preliminarily it appears that option 66 will only work with TFTP, but I'll
> follow up with whether or not that's  'official' or de-facto.
>
> -Karl

I know for a fact that you can provision a Polycom via ftp.  I've included 
much of my dhcpd.conf file below.  Pick out what you need.  Let me know if 
you have questions or further difficulty.

==
ddns-update-style ad-hoc;

option subnet-mask 255.255.255.0;
option netbios-name-servers 10.0.1.1;
option domain-name-servers 208.67.222.222;
option subnet-mask 255.255.255.0;
option boot-server code 66 = string;
option time-servers pool.ntp.org;

subnet 10.0.1.0 netmask 255.255.255.0 {
option broadcast-address 10.0.1.255;
option routers 10.0.1.1;
option tftp-server-name "10.0.1.1";
range 10.0.1.50 10.0.1.60;
allow unknown-clients;
authoritative;
one-lease-per-client off;
}

# Polycom phones
group {
option boot-server "ftp://polycom:pas...@10.0.1.1";;
option tftp-server-name "ftp://polycom:pas...@10.0.1.1";;

option time-offset -25200;

host 0004f2278ff8 {
hardware ethernet 00:04:F2:27:8F:F8;
}

host 0004f22afafd{
hardware ethernet 00:04:F2:2A:5A:FD;
}
}


-- 

Take care and have fun,
Mike Diehl.

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Re: [asterisk-users] Polycom DHCP Option 66 FTP provisioning

[Karl Fife]

- Original Message - 
From: "Lee, John (Sydney)" 
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Wednesday, March 17, 2010 9:50 PM
Subject: Re: [asterisk-users] Polycom DHCP Option 66 FTP provisioning


>
>> I'll see if E4Strategies can open a support ticket at Polycom.
>> They're really good about stuff like that.  I'll let you know either
> way.
>>
> What is E4Strategies?
> Polycom support is hopeless in Oz.  They just shove you to some
> distributer who only knows to replace your hardware.
>

e4strategies is the where I bought all of my Polycom hardware.

It would seem that e4 moves enough Polycom hardware so P pays attention to 
e4 when they open a support ticket, [which is why] I (and others) buy 
Polycom from e4, which is why P pays attention to e4 , which is why others 
buy from e4...  Loop: goto [which is why] :-)

Preliminarily it appears that option 66 will only work with TFTP, but I'll 
follow up with whether or not that's  'official' or de-facto.

-Karl


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[asterisk-users] Problem with forwarding: Now forwarding SIP/ XX to Local/

[Alex Rendour]

Hello,

here my achitecture:
client1--Asterisk1ser1---centile
client2--

client1 do a call to centile.
centile do a forward to client2 (Diversion) and then use the same CALL-ID!
when asterisk1 receive the call with the same CALL-ID, it screen "Now
forwarding SIP/ -02f6 to 'Local/m...@kamailio ' (thanks to
SIP/YYY-02f7)"

I don't want that asterisk receive the call in local because I can't
read headers in local...

anyone have a solution to accept a call with the same CALL-ID in SIP
channel ?


thank you for all,
regards,


-- 
Alexandre Rendour

Acropolis Telecom 
Direct: +33 (0) 181813201
Support: +33 (0) 811 851 851
rend...@acropolistelecom.net

Adresse : 161-163 avenue Gallieni
Paris - Porte de Bagnolet
93170 Bagnolet

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Re: [asterisk-users] Setting up RTP to flow between endpoints directlybypassingAsterisk

[Jeff Brower]

Klaus-

> Looks like the term "native bridging" is a bit overloaded.
>
> Some text from channel.h:
>  -# When the call is answered, Asterisk bridges the media streams
> so the caller on the first channel can speak with the callee
> on the second, outbound channel
>  -# In some cases where we have the same technology on both
> channels and compatible codecs, a native bridge is used.
> In a native bridge, the channel driver handles forwarding
> of incoming audio to the outbound stream internally, without
> sending audio frames through the PBX.
>  -# In SIP, theres an "external native bridge" where Asterisk
> redirects the endpoint, so audio flows directly between the
> caller's phone and the callee's phone. Signalling stays in
> Asterisk in order to be able to provide a proper CDR record
> for the call.
>
> See also
> http://lists.digium.com/pipermail/asterisk-dev/2010-March/043052.html

Yes seems so.  Many layers of subtlety :-)

-Jeff

> Am 17.03.2010 23:34, schrieb Jeff Brower:
>> Klaus-
>>
>>> Am 16.03.2010 01:42, schrieb Jeff Brower:
 Vikram-

> http://www.voip-info.org/wiki/view/Asterisk+Letting+SIP+clients+connect+directly
>
> The link above indicates that it is possible to setup RTP streams to
> directly flow between endpoints and completely bypass Asterisk. I would
> like to know if this configuration would work when,
>
> a) both endpoints are behind NAT, and/or
> b) both endpoints don't support same codecs
>
> with media flowing through a SIP+rtpproxy server that can do
> transcoding ?

 This would be 'native bridging' mode as I've seen it described a few 
 places on the web, correct?  If Asterisk is
 "out
 of the RTP loop", then what can it still do?  Only billing?  It would not 
 detect DTMF, no RTP record or
 announcement
 playout, etc.
>>>
>>> No, this this is not native bridging.
>>>
>>> Asterisk supports 3 methods of media handling:
>>> 1. bridging: media (audio, video) is received on one channel, handled
>>> over to Asterisk's core, forwarded to the bridged channel, and sent out
>>> again.
>>>
>>> 2. native-bridging: if both bridged channels use the same technology
>>> then media can be bridged directly in the channel driver, no need to
>>> feed the media into Asterisk's core. For example SIP-to-SIP calls or
>>> DAHDI-to-DAHDI calls.
>>
>> This is not what I understood initially from the Digium / voip-info.org web 
>> pages.
>> For example in the SIP-to-SIP case, are you saying that still the 
>> motherboard NIC
>> would be used and the Linux kernel would "touch" every packet, but Asterisk 
>> software
>> would not?  My understanding was that RTP would flow direct between the NICs 
>> on the
>> devices.
>>
>>> 3. bypass: here, the media flow bypasses Asterisk directly. AFAIK this
>>> works only with SIP as channel technology. This comes in 2 flavors:
>>>
>>>3a) During call setup the media will be forwarded via Asterisk. Once
>>> the call is set-up, Asterisk will send reINVITEs to both clients using
>>> the clients original SDP contact information. For this you must set
>>> canreinvite=yes (1.4) or directmedia=yes (1.6) in sip.conf. Of course
>>> Asterisk will initiate the direct media only if the media is not needed
>>> in Asterisk, e.g. if you monitor a call, the media will always be routed
>>> via Asterisk.
>>
>> Ok this is what I was expecting.  I thought that canreinvite=yes was 
>> equivalent to
>> native bridging, but evidently there is a distinction here that I need to 
>> study.
>>
>>>3b) Media will bypass Asterisk from the beginning. Therefore you have
>>> to set directrtpsetup=yes. This is still experimental and causes weird
>>> reINVITEs (e.g. after call setup to lock down on a certain codec or
>>> after call termination to redirect media to Asterisk before hanging up).
>>>
>>> Both bypass modes Note only work if either there are no NATs at all, or
>>> the clients are behind the same NAT and do not use STUN.
>>
>> Ok.  Thanks for this info.  I was not aware of directrtpsetup.
>>
>> -Jeff
>


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Re: [asterisk-users] Setting up RTP to flow between endpoints directlybypassingAsterisk

[Klaus Darilion]

Hi Jeff!

Looks like the term "native bridging" is a bit overloaded.

Some text from channel.h:
 -# When the call is answered, Asterisk bridges the media streams
so the caller on the first channel can speak with the callee
on the second, outbound channel
 -# In some cases where we have the same technology on both
channels and compatible codecs, a native bridge is used.
In a native bridge, the channel driver handles forwarding
of incoming audio to the outbound stream internally, without
sending audio frames through the PBX.
 -# In SIP, theres an "external native bridge" where Asterisk
redirects the endpoint, so audio flows directly between the
caller's phone and the callee's phone. Signalling stays in
Asterisk in order to be able to provide a proper CDR record
for the call.


See also
http://lists.digium.com/pipermail/asterisk-dev/2010-March/043052.html


klaus

Am 17.03.2010 23:34, schrieb Jeff Brower:
> Klaus-
>
>> Am 16.03.2010 01:42, schrieb Jeff Brower:
>>> Vikram-
>>>
 http://www.voip-info.org/wiki/view/Asterisk+Letting+SIP+clients+connect+directly

 The link above indicates that it is possible to setup RTP streams to
 directly flow between endpoints and completely bypass Asterisk. I would
 like to know if this configuration would work when,

 a) both endpoints are behind NAT, and/or
 b) both endpoints don't support same codecs

 with media flowing through a SIP+rtpproxy server that can do
 transcoding ?
>>>
>>> This would be 'native bridging' mode as I've seen it described a few places 
>>> on the web, correct?  If Asterisk is "out
>>> of the RTP loop", then what can it still do?  Only billing?  It would not 
>>> detect DTMF, no RTP record or announcement
>>> playout, etc.
>>
>> No, this this is not native bridging.
>>
>> Asterisk supports 3 methods of media handling:
>> 1. bridging: media (audio, video) is received on one channel, handled
>> over to Asterisk's core, forwarded to the bridged channel, and sent out
>> again.
>>
>> 2. native-bridging: if both bridged channels use the same technology
>> then media can be bridged directly in the channel driver, no need to
>> feed the media into Asterisk's core. For example SIP-to-SIP calls or
>> DAHDI-to-DAHDI calls.
>
> This is not what I understood initially from the Digium / voip-info.org web 
> pages.
> For example in the SIP-to-SIP case, are you saying that still the motherboard 
> NIC
> would be used and the Linux kernel would "touch" every packet, but Asterisk 
> software
> would not?  My understanding was that RTP would flow direct between the NICs 
> on the
> devices.
>
>> 3. bypass: here, the media flow bypasses Asterisk directly. AFAIK this
>> works only with SIP as channel technology. This comes in 2 flavors:
>>
>>3a) During call setup the media will be forwarded via Asterisk. Once
>> the call is set-up, Asterisk will send reINVITEs to both clients using
>> the clients original SDP contact information. For this you must set
>> canreinvite=yes (1.4) or directmedia=yes (1.6) in sip.conf. Of course
>> Asterisk will initiate the direct media only if the media is not needed
>> in Asterisk, e.g. if you monitor a call, the media will always be routed
>> via Asterisk.
>
> Ok this is what I was expecting.  I thought that canreinvite=yes was 
> equivalent to
> native bridging, but evidently there is a distinction here that I need to 
> study.
>
>>3b) Media will bypass Asterisk from the beginning. Therefore you have
>> to set directrtpsetup=yes. This is still experimental and causes weird
>> reINVITEs (e.g. after call setup to lock down on a certain codec or
>> after call termination to redirect media to Asterisk before hanging up).
>>
>> Both bypass modes Note only work if either there are no NATs at all, or
>> the clients are behind the same NAT and do not use STUN.
>
> Ok.  Thanks for this info.  I was not aware of directrtpsetup.
>
> -Jeff

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Re: [asterisk-users] Asterisk DIES with no trace. PLEASE

[Zeeshan Zakaria]

Was there any hardware upgrade in December after which you recompiled
libpri?

On 2010-03-18 10:06 AM, "Danny Dias"  wrote:

Thanks Zeeshan,

SAngoma told me that the asterisk problem is unrelated to wanpipe drivers,
they told me to reinstall asterisk again

But, i still having doubts about the problem :(

Thanks in advance


>
> Message: 10
> Date: Thu, 18 Mar 2010 00:21:06 -0400
> From: Zeeshan Zakaria 
> Subject: Re: [asterisk-users] Asterisk DIES with no trace. PLEASE
>HELP!
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Message-ID:
><5ad99e891003172121s52a3b386k5210ce03e7196...@mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Your configs seem good. It is hard to guess why you are having this
> problem.
> You'll need to get help from Sangoma as it is their hardware and they'll be
> able to check if it is a driver related issue.
>
>
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Re: [asterisk-users] Asterisk DIES with no trace. PLEASE

[Zeeshan Zakaria]

Do you properly hang up the calls. Does 'zap show channel '
shows that the channel is 'on hook' after its hang up?

On 2010-03-18 10:06 AM, "Danny Dias"  wrote:

Thanks Zeeshan,

SAngoma told me that the asterisk problem is unrelated to wanpipe drivers,
they told me to reinstall asterisk again

But, i still having doubts about the problem :(

Thanks in advance


>
> Message: 10
> Date: Thu, 18 Mar 2010 00:21:06 -0400
> From: Zeeshan Zakaria 
> Subject: Re: [asterisk-users] Asterisk DIES with no trace. PLEASE
>HELP!
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Message-ID:
><5ad99e891003172121s52a3b386k5210ce03e7196...@mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Your configs seem good. It is hard to guess why you are having this
> problem.
> You'll need to get help from Sangoma as it is their hardware and they'll be
> able to check if it is a driver related issue.
>
>
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[asterisk-users] SIP Router Project

[Randy R]

Hello,

This Friday on VUC, the SIP Router Project,  Kamailio 3.0 will be
discussed with a couple experts. Your questions are welcome, as
always.

See the site: http://vuc.me for ways to phone in. For the best sound,
use g722 and call 200...@login.zipdx.com at 12 Noon Eastern.
See http://vuc.me/next for exact time in your area.

Reminder too that the VoIP Users Conference will be exactly three
years running on Friday March 26th. We'll be doing a special 24 hour
Voipathon. Those of you in the Southern Hemisphere and Asia who are
usually not awake at the time of VUC are cordially invited to join in
the discussion, which will be much wider than VoIP geekdom:
http://voipathon.org for more info on that.

We've had some great discussions on VUC and hope to continue to do so.
Please consider joining us as we are a true community, not just a
podcast. We meet weekly and the talk is live. IRC channel : #vuc

/r

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Re: [asterisk-users] Voicemail Remote Access

[Dan Journo]

Its ok, I discovered the issue.

The DTMP signals weren't being received.

All sorted now.

Thanks
Dan

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: 18 March 2010 14:10
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voicemail Remote Access

Dan Journo wrote:
>
> Hi,
>
> Any ideas?
>
>

I'd be helpful to see the console output.

Doug


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Re: [asterisk-users] Voicemail Remote Access

[Doug Lytle]

Dan Journo wrote:
>
> Hi,
>
> Any ideas?
>
>

I'd be helpful to see the console output.

Doug


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Re: [asterisk-users] Asterisk DIES with no trace. PLEASE

[Danny Dias]

Thanks Zeeshan,

SAngoma told me that the asterisk problem is unrelated to wanpipe drivers,
they told me to reinstall asterisk again

But, i still having doubts about the problem :(

Thanks in advance


>
> Message: 10
> Date: Thu, 18 Mar 2010 00:21:06 -0400
> From: Zeeshan Zakaria 
> Subject: Re: [asterisk-users] Asterisk DIES with no trace. PLEASE
>HELP!
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Message-ID:
><5ad99e891003172121s52a3b386k5210ce03e7196...@mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Your configs seem good. It is hard to guess why you are having this
> problem.
> You'll need to get help from Sangoma as it is their hardware and they'll be
> able to check if it is a driver related issue.
>
>
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Re: [asterisk-users] spandsp with asterisk 1.4.x

[Vinícius Fontes]

- "Joao Gomes Pereira"  escreveu:

> Em 17-03-2010 20:51, Vinícius Fontes escreveu:
> > - "Joao Gomes Pereira"  escreveu:
> >
> >
> >> Hello
> >> Im trying to receive FAXes with my Asterisk with "rxfax" command.
> >>
> >> To do that, Im trying to load the "app_fax.so" module but
> asterisk
> >> says:
> >>
> >> [Mar 17 20:06:04] WARNING[11907]: loader.c:359
> load_dynamic_module:
> >> Error loading module 'app_fax.so': libspandsp.so.2: cannot open
> shared
> >>
> >> object file: No such file or directory
> >> [Mar 17 20:06:04] WARNING[11907]: loader.c:653 load_resource:
> Module
> >> 'app_fax.so' could not be loaded.
> >>
> >> But I do have libspandsp.so.2
> >> # find / -name "libspandsp.so.2"
> >> /usr/local/lib/libspandsp.so.2
> >>
> >>
> >> And yes, "/usr/local/lib" is in my "ld.so.conf":
> >>
> >> cat /etc/ld.so.conf
> >> include ld.so.conf.d/*.conf
> >> /etc/ld.so.conf.d/*.conf
> >> /usr/local/lib
> >> /usr/include
> >> /usr/local/include
> >>
> >> What could be missing?
> >>
> >> Thanks
> >> Regards
> >> Joao Pereira
> >>  
> > Sorry for kinda hijacking your topic, but where did you get the 1.4
> app_fax.so backport from? I'm really interested on that.
> >
> Yes, It was difficult to find...
> I dont have the page, but here is the wget:
>   wget 
> http://downloads.sourceforge.net/project/agx-ast-addons/precompiled-linux-spandsp-app-fax.tar.bz2

Thanks a lot! Too bad it requires Zaptel instead of DAHDI. :(

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[asterisk-users] Software for my laptop to send Fax via H.323 ?

[Jason Aarons (US)]

I'm trying to test a Diaglogic BrookTrout SR140 card. It uses H.323.

Trying to find a way I could use my laptop to send a fax over H323 to the 
BrookTrout card for testing.  Any thoughts?  Normally I'd setup a FXS interface 
on a Cisco router and setup a h323 dial peer to the BrookTrout, but I didn't 
the router with me!



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Re: [asterisk-users] spandsp with asterisk 1.4.x

[Joao Gomes Pereira]

Em 17-03-2010 20:28, Doug Lytle escreveu:
> Joao Gomes Pereira wrote:
>
>> What could be missing?
>>
>>
>>  
> Running ldconfig as root
>
>
>
>
Thanks, thats it!!!

Now the module is loaded.
I just hope the FAX code works:

[macro-faxreceive]
  exten => s,1,Set(FAXFILE=/var/spool/asterisk/fax/${CALLEDFAX}/${UNIQUEID})
exten => s,2,NoOP()
exten => s,3,NoOP()
  exten => s,4,rxfax(${FAXFILE}.tif)
  exten => s,103,Set(extmail...@startel.pt)
  exten => s,104,Goto(4)
  exten => s,105,Set(EXTNAME=Unknown)
  exten => s,106,Goto(4)
  exten => s,107,Set(EXTCOMPANY=Company)
  exten => s,108,Goto(4)


Thanks again
Regards
Joao Pereira

-- 
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Joao Gomes Pereira
www.startel.pt
+351 304500650
sip: gomespere...@startel.pt


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Re: [asterisk-users] sip send image

[Danny Nicholas]

Do a link to the image as URL on the dial command?

 

-  exten =>
s,1,Dial(SIP/12345,20,KkTT,http://www.yahoo.com/image.jpg)

 

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bhrugu mehta
Sent: Wednesday, March 17, 2010 8:29 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] sip send image

 

Thnks for ur reply,

SendImage() doesn't work with asterisk sip channel.
any other solution?

Regards,
-- 
Bhrugu Mehta
Sr. S/W Engineer (D&D)
VOIP,Telephony Team
India

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Re: [asterisk-users] Door Phone Assistance

[Danny Nicholas]

This is a longshot, but the FXS indication tells me you're using DAHDI.  Put
an answer at the start of the custom context and see if that solves your
problem.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Grignon
Sent: Wednesday, March 17, 2010 4:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Door Phone Assistance

 

I have two Viking E10 Door phones and a Rhino FXS channel bank...

 

I have the channel set to immediate=yes and defined a custom context...

 

When I press the button on the door phone, the inside phone rings and I can
hear the person talk through the door phone... The problem is I cant hear
anything through the speaker of the door phone...

 

I know the speaker works because I do hear the initial ringing but that's
it...

 

Could this be a voltage issue? I tried two different Viking Units...

 

Thanks for any assistance.

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Re: [asterisk-users] Call Filtering

[Danny Nicholas]

This is just my approach, but I would run call 1 into an AGI that produced
the second call through AMI, then proceeded based on the return.

-  exten => 123,1,answer

-  exten => 123,2,AGI(callproc.agi)

-  exten => 123,3,Gotoif($["${PROC}" = "VM"]?voicemail) 

-  exten => 123,4,Queue

-  exten => 123,5(voicemail),Voicemailmain

-  exten => 123,6,hangup

 

callproc.agi calls your staff member - if they press 1, variable proc is set
to VM.

 

 

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Wednesday, March 17, 2010 10:36 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Call Filtering

 

Hi,

 

I would like to develop a dialplan that allows the callee to reject the call
like this:-

 

1) Call comes in and receives a greeting and get put into a queue.

2) A second call is placed to the member of staff (SIP phone or mobile
phone)

3) The member of staff answers the call and is presented with a few options.

4) If the member of staff presses 1, the incoming call is connected to the
member of staff.

5) If the member of staff hangs up or presses 2, the incoming call is sent
to a voicemail box.

 

The problem being, I can't see to place the second call without bridging the
first call. 

 

Can anyone point me in the right direction?

 

Many thanks

Dan

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Re: [asterisk-users] spandsp with asterisk 1.4.x

[Joao Gomes Pereira]

Em 17-03-2010 20:51, Vinícius Fontes escreveu:
> - "Joao Gomes Pereira"  escreveu:
>
>
>> Hello
>> Im trying to receive FAXes with my Asterisk with "rxfax" command.
>>
>> To do that, Im trying to load the "app_fax.so" module but asterisk
>> says:
>>
>> [Mar 17 20:06:04] WARNING[11907]: loader.c:359 load_dynamic_module:
>> Error loading module 'app_fax.so': libspandsp.so.2: cannot open shared
>>
>> object file: No such file or directory
>> [Mar 17 20:06:04] WARNING[11907]: loader.c:653 load_resource: Module
>> 'app_fax.so' could not be loaded.
>>
>> But I do have libspandsp.so.2
>> # find / -name "libspandsp.so.2"
>> /usr/local/lib/libspandsp.so.2
>>
>>
>> And yes, "/usr/local/lib" is in my "ld.so.conf":
>>
>> cat /etc/ld.so.conf
>> include ld.so.conf.d/*.conf
>> /etc/ld.so.conf.d/*.conf
>> /usr/local/lib
>> /usr/include
>> /usr/local/include
>>
>> What could be missing?
>>
>> Thanks
>> Regards
>> Joao Pereira
>>  
> Sorry for kinda hijacking your topic, but where did you get the 1.4 
> app_fax.so backport from? I'm really interested on that.
>
Yes, It was difficult to find...
I dont have the page, but here is the wget:
  wget 
http://downloads.sourceforge.net/project/agx-ast-addons/precompiled-linux-spandsp-app-fax.tar.bz2

Regards
Joao Pereira

-- 
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Joao Gomes Pereira
www.startel.pt
+351 304500650
sip: gomespere...@startel.pt


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Re: [asterisk-users] SIP codec negotiation / manipulation

[Kevin Sandy]

On 3/17/2010 6:25 PM, Jeff Brower wrote:
> Steve-
> 
> On Wed, Mar 17, 2010 at 6:02 PM, Jeff Brower 
> mailto:jbro...@signalogic.com>> wrote:
> Steve-
> 
>> 2010/3/17 Vinícius Fontes 
>> mailto:vinic...@canall.com.br>>
>>
>>> - "Kevin Sandy" mailto:kevin.sa...@snohio.net>> 
>>> escreveu:
>>>
 We're having an odd issue with codec negotiation from one of our SIP
 providers. Here's the basic situation.

 We receive an invite from them advertising support for G711, G729, and
 G723. In our response, we send back that we support G711 and G729. In
 about half the cases, this results in no problems, with audio being
 encoded with G711. The other half of the time, they send us a second
 invite requesting G729. However, they proceed to send us a G711
 encoded audio stream...

 They have somewhat acknowledged the problem, but their advice is for
 us to only accept a single codec in our 200 OK. We don't want to
 disable either; we have customers using G729, so we'd like to avoid
 transcoding when possible, but we also do some T38 faxing, which I
 believe requires G711 to start off.

 My first thought was to selectively force the codec on inbound calls -
 if it is for a voice number, use 729, otherwise 711. However, I can't
 find any way of doing this within Asterisk. (We do have an OpenSIPS
 server sitting between us and the provider, and I could use OpenSIPS
 features to do this; however, right now the OpenSIPS server is fairly
 dumb - it's only proxying traffic between us and the provider and
 knows nothing about our specific DIDs.)

 A couple more details in case anyone has seen a similar issue. The
 provider is Broadvox, and this issue only seems to manifest on calls
 coming to them via Skype. They claim to not have any direct link with
 Skype, but it seems odd that the problem would be specific to Skype
 callers if the call is coming to Broadvox as a standard PSTN call.

 Is there any way to do this? Am I totally missing something and making
 a stupid mistake, or making the issue more complicated than it needs
 to be?

>>>
>>> If your only concern about using G711 is regarding T38, go ahead and enable
>>> G729 only. T38 doesn't need G711 at all.
>>>
>>>
>> If your customers don't mind G729 then what is said above is fine.
>>
>> There will be a T.38 reinvite so it won't be G729 anymore.  Canreinvite does
>> not need to be set to yes for this to work in your sip.conf either.  It can
>> be confusing but they are different types of reinvites.
> 
> I don't see how this can work if Broadvox then sends G711 anyway.  I 
> understand that to be the OP's root problem.
> 
> -Jeff
> 
> It doesn't matter what the codec is initially, if the provider supports T.38 
> and you do too, a reinvite is sent changing whatever codec over to T.38.
> 
> I meant for the Broadvox voice output, but maybe your suggestion works Ok and 
> solves his problem.
> 
> -Jeff
> 
> 



Well, I at least have more things to look into. A couple notes...

1. The provider's thought is that the reason audio is encoded
incorrectly is that our 200 OK with multiple codecs is confusing their
equipment. They believe that if we respond with only a single codec
(which can be any of their supported codecs, and can be different per
call), their equipment will handle it correctly and use the single
negotiated codec.

2. I had thought there was a limit in Asterisk that it would only detect
fax tones and send the re-invites for T38 if the call started as a more
or less uncompressed G711 call. I may be confusing that with a
limitation of some of the desktop soft-phone / fax clients I've used.


It seems that at the moment, the simplest solution is going to be to
setup our outbound SIP proxy as a peer in Asterisk (we're currently just
hitting it by using the proxy's IP in the Dial command). We can then
enable only a single codec for the outbound proxy while still allowing
customer phones to use either.

Unless... and I doubt it... there is some command or variable I can set
before calling Answer that will modify the list of codecs we send back.
That would be my ideal solution, as we could then look at the DID and
decide which codec we would like for this particular call based on which
codec that customer is using.



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[asterisk-users] Voicemail Remote Access

[Dan Journo]

Hi,

I'm trying to set up remote voicemail pickup. I've created the following 
dialplan, but when I press *, I am not sent to voicemailmain. The unavailable 
message just continues to play as normal.

exten => 234555,1,Set(MAILBOXID=1)
exten => 234555,n,Set(MAILBOXCONTEXT=company3)
exten => 234555,n,Voicemail(${mailbox...@${mailboxcontext},u)
exten => a,1,VoicemailMain(${mailbox...@${mailboxcontext})

Any ideas?

Thanks
Dan

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Re: [asterisk-users] Door Phone Assistance

[Robert Grignon]

Yes it does. 

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F
Sent: Wednesday, March 17, 2010 8:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Door Phone Assistance

Does a regular phone work on that port of the channel bank?

On Wed, Mar 17, 2010 at 5:00 PM, Robert Grignon  wrote:
> I have two Viking E10 Door phones and a Rhino FXS channel bank...
>
> I have the channel set to immediate=yes and defined a custom context...
>
> When I press the button on the door phone, the inside phone rings and 
> I can hear the person talk through the door phone... The problem is I 
> cant hear anything through the speaker of the door phone...
>
> I know the speaker works because I do hear the initial ringing but 
> that's it...
>
> Could this be a voltage issue? I tried two different Viking Units...
>
> Thanks for any assistance.
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Re: [asterisk-users] Call Filtering

[Dan Journo]

Thanks. However, I discovered a guide on doing this at the following url:-

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

Example 2 shows to use a macro to present a menu to the member of staff before 
the call is bridged.

Many thanks
Dan

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Sandy
Sent: 17 March 2010 17:48
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call Filtering

We've done this with a bit of trickery - I believe we use System to copy 
a blank audio file into place before calling Dial so that Asterisk 
thinks the caller has already recorded their name.

On 3/17/2010 1:42 PM, Dan Journo wrote:
> Thats similar to how I want it to work, however I dont want the caller to 
> have to give their name (even the first time they call)
> Is there any way of using the p option of the dial command, but totally 
> remove the caller name recording feature?
>
> Thanks
> Dan
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Sandy
> Sent: 17 March 2010 15:47
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Call Filtering
>
> Sounds like you want some type of call screening. Check out the p
> option to the Dial command.
>
>
>> Hi,
>>
>> I would like to develop a dialplan that allows the callee to reject
>> the call like this:-
>>
>> 1) Call comes in and receives a greeting and get put into a queue.
>> 2) A second call is placed to the member of staff (SIP phone or
>> mobile phone)
>> 3) The member of staff answers the call and is presented with a few
>> options.
>> 4) If the member of staff presses 1, the incoming call is connected
>> to the member of staff.
>> 5) If the member of staff hangs up or presses 2, the incoming call
>> is sent to a voicemail box.
>>
>> The problem being, I can't see to place the second call without
>> bridging the first call.
>>
>> Can anyone point me in the right direction?
>>
>> Many thanks
>> Dan
>

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[asterisk-users] Live Audio Streaming- From Aux interface-Online resource

[ABBAS SHAKEEL]

Hello all,

I would like to know if any one have experience with live audio streaming
like

1. Streaming from an online resource
2. Streaming from sound card AUX interface..

What i want to accomplish is that on receiving a callers call i play back a
live audio stream or stream from sound card AUX interface(It depend on
caller choice).

Thanks

-- 
Best Regards
Shakeel Abbas
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Re: [asterisk-users] How to detect a PSTN telephone is busy or not?

[ABBAS SHAKEEL]

Hello,

Please have a look to DIALSTATUS variable. here
http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS

I hope it
helps



On Thu, Mar 18, 2010 at 1:31 PM, Zhang Shukun  wrote:

> hi,all
>
> one problem confuse me these days. i want to sequence dial three PSTN
> number(a,b,c)
>
> first, if i dial number a, if a is busy , i will dial number b. if b
> is busy, i will dial number c.
>
> Dial(SIP/a...@ip,30)
> Dial(SIP/b...@ip,30)
> Dial(SIP/c...@ip,30)
>
> i want to know before i dial number a, how to know if a is busy now?
>
> if a is busy now. i will not dial a, instead, i will dial number b
> directly.
>
> to summary is : in asterisk, how to detect a pstn telephone number is
> busy or not before dialing it?
>
> Thanks!
>
> --
> Best regards,
> Sucan
>
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Shakeel Abbas
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Re: [asterisk-users] press release: Attrafax t.30 and t.38 alternative now released as gpl2 + commercial license.

[Klaus Darilion]

Am 17.03.2010 19:31, schrieb Matt Watson:
>
>
> On Tue, Mar 9, 2010 at 6:31 PM, Klaus Darilion
> mailto:klaus.mailingli...@pernau.at>> wrote:
>
>
>
> Attached is an untested (I did not had the time yet) port to
> Asterisk 1.4.29.1 (DAHDI). Maybe the modules need some adaptions too.
>
> Maybe someone wants to give it a try.
>
> regards
> klaus
>
>
> Just as an FYI, your 1.4.29.1 patch applies successfully against 1.4.30
> as well.  I've got a patched 1.4.30 system compiled and ready to install
> later tonight during off-hours and will begin having people test
> tomorrow.  People here are going to be quite thrilled about having T.38
> transparent gatewaying again.

Fine. I had not time yet to test the 1.4.29.1 patch.

FYI: For 1.6.2.6 there is also a patch available [1], based on spandsp.

regards
klaus

[1] https://issues.asterisk.org/view.php?id=13405

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Re: [asterisk-users] asterisk fax handeling

[Klaus Darilion]

Am 18.03.2010 05:11, schrieb Olivier:
>
>
> 2010/3/17 Klaus Darilion  >
>
>
>
> Am 17.03.2010 10:40, schrieb Peter den Hartog:
>  > Hello,
>  >
>  > I was wondering if the following was possible:
>  > When somebody sends a fax to my direct number 0101234567105 (my
>  > extension will be 105) is it possible that Asterisk, or an addon sees
>  > this as a fax, and e-mail the fax to me?
>  > So everybody with a private extension will be able to receive
> faxes in
>  > his e-mailbox on his direct number.
>
> Yes, that should work (at least with 1.6.2):
>
> 1. Enable fax detection on the inbound channel, e.g. sip.conf:
>
> ; FAX detection will cause the SIP channel to jump to the 'fax'
> extension (if it exists)
> ; when a CNG tone is detected on an incoming call.
> ;
> ; faxdetect = yes  ; Default false
>
>
> chan_dahdi.conf:
> ; For fax detection, uncomment one of the following lines.  The default
> is *OFF*
> ;
> ;faxdetect=both
> ;faxdetect=incoming
> ;faxdetect=outgoing
> ;faxdetect=no
>
>
> Further, put a "fax" extension in the context where you handle the fax,
> e.g. forward to Hylafax are receive directly with ReceiveFAX().
>
> IIRc the originaly dialed number will be written in the variable
> FAXEXTEN. Thus, you can send the fax to the respective email address
> based on FAXEXTEN.
>
> regards
> klaus
>
>
>
> As IMHO, fax detection needs callee to answer the incoming call, how
> should I proceed to get a pre-recorded audio file played ("You're bout
> to receive an incoming fax call") to callee while the incoming fax call
> is converted into a fax file ?
>
> My understanding of Asterisk dialplan is :
> - a fax call comes in from channel A,
> - appropriate extension is dialed through channel B,
> - user answers and channels A and B are bridged,
> - Asterisk detects the call is a fax call and then :
> --- 1. stops channel B
> --- 2. jumps into dialplan fax priority
>
> Is it possible to play a message on channel B before stopping ?
>
> Regards
>
>
> makes me think that as soon as a fax is detected, the receiving channel
> is stopped

Actually I don't know, I have never used it yet.

regards
klaus


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[asterisk-users] How to detect a PSTN telephone is busy or not?

[Zhang Shukun]

hi,all

one problem confuse me these days. i want to sequence dial three PSTN
number(a,b,c)

first, if i dial number a, if a is busy , i will dial number b. if b
is busy, i will dial number c.

Dial(SIP/a...@ip,30)
Dial(SIP/b...@ip,30)
Dial(SIP/c...@ip,30)

i want to know before i dial number a, how to know if a is busy now?

if a is busy now. i will not dial a, instead, i will dial number b directly.

to summary is : in asterisk, how to detect a pstn telephone number is
busy or not before dialing it?

Thanks!

-- 
Best regards,
Sucan

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