Re: [asterisk-users] externip nat audio sip trunk issue problem

[Gordon Messmer]

On 02/01/2012 06:14 PM, Gabriel Ortiz Lour wrote:

when I configure externip/localnet correctly my SIP trunk simply
disappear! Checking the signalling with tcpdump shows me that Im sending
the packets to the correct SIP trunk IP but there is no response AT ALL
from it...


Use tcpdump -v (or wireshark) to look at the SIP packet contents as well 
as the IP headers.  If possible, do this on the external interface of 
your firewall to see what's getting sent out to your peer.  See if the 
values all appear to be sane.


Compare the SIP session contents of packets with externip configured and 
packets without that setting.


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Re: [asterisk-users] Is this doable?

[Gordon Messmer]

On 02/01/2012 04:48 PM, Josh wrote:

The purpose of registering this external account is so that both the
smart phones (tun0) and the internal net (eth1) users could use this
account to make external calls (starting with "0", i.e "_0[0-9]."
pattern in extensioins.conf). Obviously, I need these calls to be routed
properly via the external VOIP account. In addition to that, I would
also need to receive calls from that external account to a nominated
internal one (say on extension 20).

Is this achievable?


I can't see any reason it shouldn't be.


If so, I am not completely clear on whether I need to explicitly specify
my public IP address (via externip/externhost) or whether Asterick is
able to find it without this option?


As I understand it, that depends on your router.  If you have a Linux 
router with the ip_nat_sip module, it'll "fix" your SIP packets so that 
you don't need to use the externip setting.  However, you'll need to 
test to verify that.


Asterisk won't be able to figure out your external address on its own, 
so if your firewall isn't fixing packets, then you'd need to specify 
externip.



If not, then my plan is to use
external program to find it and then use a script in Asterick to set it
up as an environment variable. Would that work?


http://www.voip-info.org/wiki/view/Asterisk+variables
According to the information here, you should be able to use 
${ENV(externip)} to reference the value of an environment variable named 
"externip".



I am also not sure whether to specify
"nat=yes" or just have "nat=route" only - any ideas?


http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html
For a SIP trunk... no, I don't.  The above link may be useful as it 
describes NAT issues with SIP.  If you have to specify NAT options at 
all, start with "yes" and try "route" if that doesn't work.



Is there a comprehensive list of all the options available in sip.conf
and what they do, because I was unable to find such a list?


http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf
I wish I knew.  The link above seems fairly complete, but also terse.


One final question about binding: in order to be able to use both tun0
and eth1 interfaces so that Asterick serves the calls from both eth1 and
tun0, do I have to use "bind 0.0.0.0"? Is there an alternative, like
specifying "bind 10.1.1.1" for eth1 and then "bind 10.1.2.1" for the
tun0 interface - is this possible?


Start with binding to 0.0.0.0.

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Re: [asterisk-users] amd detect answering machine

[Aurimas Skirgaila]

Hi,

do noop(${AMDCAUSE}) after  exten => 1,1,AMD() , run some test calls and
find out why the call was detected as Answering Machine and adjust amd.conf
accordingly. if I recall correctly, you can also see the AMD flow in
Asterisk in verbose mode.

I'd suspect low silence_threshold . I usually set it 384, but it's
very dependent on carrier.



On Thu, Feb 2, 2012 at 5:51 PM, Etann  wrote:

> **
> Hi,
> I have IVR and when I press 1, asterisk calls my mobile phone.
> If my mobile phone is offline, asterisk transfers to asterisk voicemail.
> I'd like asterisk detects my mobile voicemail and if my mobile voicemail
> answers, asterisk transfers to asterisk voicemail.
> For that, I used AMD.
> So I have problems ! Asterisk detects answering machine everytime!
> How do I do please ?
>
>
> extensions.conf
> [ServeurPro]
> exten => s,1,Ringing()
> exten => s,2,Wait(2)
> exten => s,3,Answer()
> exten => s,4,Set(NbInvalide=0)
> exten => s,5,Set(NbEssai=0)
> exten => s,6,background(${ChmAudio}/ServeurProBienvenu)
> exten => s,7,WaitExten(2)
>
> exten => 1,1,AMD()
> exten => 1,2,GotoIf($["${AMDSTATUS}"="MACHINE"]?1,4)
> exten => 1,3,Dial(SIP/@ippi_outgoing2,40,r)
> exten => 1,4,Voicemail(801@FloriePro,us)
> exten => i,1,Set(NbInvalide=$[${NbInvalide}+1]})
> exten => i,2,Gotoif($["${NbInvalide}" < "3"]?:6)
> exten => i,3,Playback(${ChmAudio}/ErreurSaisie)
> exten => i,4,Playback(${ChmAudio}/RetourMenu)
> exten => i,5,Goto(s,6)
> exten => i,6,Playback(${ChmAudio}/ErreurSaisie)
> exten => i,7,Playback(${ChmAudio}/Aurevoir)
> exten => i,8,Hangup()
> exten => t,1,Set(NbEssai=$[${NbEssai}+1])
> exten => t,2,Gotoif($["${NbEssai}" < "3"]?:5)
> exten => t,3,Playback(${ChmAudio}/DemandeIncomprise)
> exten => t,4,Goto(s,6)
> exten => t,5,PlayBack(${ChmAudio}/Aurevoir)
> exten => t,6,Hangup()
> exten => h,1,noOp("Statut AMD : "${AMDSTATUS})
>
> amd.conf
> [general]
> initial_silence = 2500  ; Maximum silence duration before the greeting.
> ; If exceeded then MACHINE.
> greeting = 1500   ; Maximum length of a greeting. If exceeded then MACHINE.
> after_greeting_silence = 500 ; Silence after detecting a greeting.
> ; If exceeded then HUMAN
> total_analysis_time = 5000 ; Maximum time allowed for the algorithm to
> decide
> ; on a HUMAN or MACHINE
> min_word_length = 120  ; Minimum duration of Voice to considered as a word
> between_words_silence = 50 ; Minimum duration of silence after a word to
> consider
> ; the audio what follows as a new word
> maximum_number_of_words = 3 ; Maximum number of words in the greeting.
> ; If exceeded then MACHINE
> silence_threshold = 256
>
> Thank you for your reply and for help!
>  --
>
> AMICALEMENT
> Manu
>
> SITES WEBS
> Mon site web Officiel (Manu-dpk.net) 
> Ecoutez Radio DPK 
>
> CONTACT
> - E-mail : manuli...@manu-dpk.net
> - Messenger (WLM) : m...@manu-dpk.net
> - Skype : manu-dpk
> --
>
> PS : Pour le respect de l'environnnement, n'imprimez ce mail qu'en cas de
> nécessité.
>
>
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Mvh,
Aurimas Skirgaila
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Re: [asterisk-users] Web and Email Chat

[Goke M Aruna]

Use jabberd and qmail.



On 2/3/12, bilal ghayyad  wrote:
> Hi All;
>
> Any advise for a good collaboration solution (open source)? Chat + Email
> call center.
>
> Regards
> Bilal
>
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-- 
Sent from my mobile device

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Re: [asterisk-users] Asterisk version that support Database Configuration

[Sammy Govind]

Hey Savinovich ,
Strange, its kind of recursive search. I got back to this same thread
following the very first search result.

@Bilal,
All asterisk versions support realtime (DB configurations):  If you go to a
web site called google.com, and enter "asterisk realtime", and read the
first search result, you will get your answer.


Regards,
Sammy.

On Fri, Feb 3, 2012 at 2:16 AM, C. Savinovich
wrote:

>
>
> If you go to a web site called google.com, and enter "asterisk version
> that support the ability to have the configuration in the database", and
> read the first search result, you will get your answer.
>
>
> Christian Savinovich
>
>
>   Original Message 
> Subject: [asterisk-users] Asterisk version that support Database
> Configuration
> From: bilal ghayyad 
> Date: Thu, February 02, 2012 3:51 pm
> To: asterisk-users@lists.digium.com
>
> Hi All;
>
> Which asterisk version that support the ability to have the configuration
> in the database?
>
> Regards
> Bilal
>
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>
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Re: [asterisk-users] externip nat audio sip trunk issue problem

[C F]

On Thu, Feb 2, 2012 at 8:51 AM, Gabriel Ortiz Lour
 wrote:
> As soon as I activate the exterip/localnet config there is no response at
> all, as if that IP address desappeared.
> Any packets send to it simply get no response.
>
> I've considered being linux kernel routing issue, but since without the
> exterip/localnet config it works OK I don't think this is the case.
>
> I could put the tcpdump/sip debug info here, but it would be like a
> monologue, with only the packets being sent showing without any response.
> But I can put it here if it helps...

Change it to externip and localnat and enable sip debug in asterisk
cli, then do a sip reload and post the cli output.

>
> Didn't anyone had this problem with this config option?
>
>
> 2012/2/2 C F 
>>
>> On Wed, Feb 1, 2012 at 9:14 PM, Gabriel Ortiz Lour
>>  wrote:
>> > Hi all,
>> >
>> >   I've tried search this problem on the list... no luck...
>> >
>> >   The case is:
>> >
>> > without externip/localnet config on sip.conf [general] my SIP trunk
>> > works,
>> > but with no audio NAT problem (asterisk sends the private 192 address to
>> > the
>> > outside...)
>> >
>> > when I configure externip/localnet correctly my SIP trunk simply
>> > disappear!
>> > Checking the signalling with tcpdump shows me that Im sending the
>> > packets to
>> > the correct SIP trunk IP but there is no response AT ALL from it...
>>
>> Can you explain this?
>> What do you mean no response? Is it registering? Do you have a debug
>> output?
>>
>> >
>> > Anyone had this problem?
>> >
>> > Thanks,
>> > Gabriel
>> >
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>
>
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[asterisk-users] Web and Email Chat

[bilal ghayyad]

Hi All;

Any advise for a good collaboration solution (open source)? Chat + Email call 
center.

Regards
Bilal

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Re: [asterisk-users] Quick bash tip for finding free SIP extensions from your sip.conf

[Kyle Sexton]

On Thu, Feb 2, 2012 at 5:34 PM, Steve Edwards wrote:

> On Thu, 2 Feb 2012, Kyle Sexton wrote:
>
>  Created this function on one of my machines today, thought others might
>> find it useful:
>> freesip() {
>> comm -2 <(seq $2 $3) <(cat $1 | grep ^\\[ | sort | uniq | tr -d \[ | tr
>> -d \]) | grep ^[[:digit:]]
>> }
>>
>
> Shave 4 processes:
>
> freesip()
>{
>comm -2 -3\
><(seq $2 $3)\
><(grep ^\\[[[:digit:]] $1 | sort -u | tr -d \[])
>}
>
> --
> Thanks in advance,
> --**--**
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline  Fax: +1-760-731-3000
>
>
Yeah, I didn't really optimize this.  Thanks for the tip!

-- 
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Re: [asterisk-users] Quick bash tip for finding free SIP extensions from your sip.conf

[Steve Edwards]

On Thu, 2 Feb 2012, Kyle Sexton wrote:


Created this function on one of my machines today, thought others might find it 
useful:
freesip() {
comm -2 <(seq $2 $3) <(cat $1 | grep ^\\[ | sort | uniq | tr -d \[ | tr -d \]) 
| grep ^[[:digit:]]
}


Shave 4 processes:

freesip()
{
comm -2 -3\
<(seq $2 $3)\
<(grep ^\\[[[:digit:]] $1 | sort -u | tr -d \[])
}

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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[asterisk-users] Quick bash tip for finding free SIP extensions from your sip.conf

[Kyle Sexton]

Created this function on one of my machines today, thought others might
find it useful:

freesip() {
comm -2 <(seq $2 $3) <(cat $1 | grep ^\\[ | sort | uniq | tr -d \[ | tr -d
\]) | grep ^[[:digit:]]
}

On RedHat/CentOS based systems you can create the following file to have
the function available on login:

/etc/profile.d/freesip.sh
# Free SIP extensions
freesip() {
  comm -2 <(seq $2 $3) <(cat $1 | grep ^\\[ | sort | uniq | tr -d \[ | tr
-d \]) | grep ^[[:digit:]]
}

Then if you have a large sip.conf and want to find a free extension between
200 and 299 you can login and do the following:


[host]$ freesip /etc/asterisk/sip.conf 200 299
200
201
233
245
[host]$



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Re: [asterisk-users] Router that support Asterisk

[Tzafrir Cohen]

On Wed, Feb 01, 2012 at 06:47:49PM -0500, James Sharp wrote:
> On 02/01/2012 02:17 PM, bilal ghayyad wrote:
> >Hi All;
> >
> >I heard from some friends that there are a dsl router that has Linux OS
> >and it has asterisk on it, so the ip phone can register on this router,
> >also if the router has FXS or FXO ports then it can be used to place
> >calls through them.

Build your own. Search for e.g. OpenWRT. I currently run Asterisk (from
a Debian package) on a DreamPlug running a stock Debian/armel system.

As a rule of thumb: you need a "router" with an option for extra storage
(e.g. through USB).

> >
> >Is it really? Where I can these routers? Did anyone try it to tell us if
> >it is stable and working fine?
> >
> >Regards
> >Bilal
> 
> The Cisco DDR2200 that I just got from Centurylink for DSL appears
> to be just that.  I haven't tested the FXS ports on it yet, though.

Is there any such device with drivers for the FXS and / or FXO ports?

http://villagetelco.org/mesh-potato/ has FXS, but I'm not sure about
FXOs.

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Re: [asterisk-users] Proposed changes to Asterisk release and support cycles

[Tzafrir Cohen]

On Tue, Jan 31, 2012 at 11:39:21AM -0600, Kevin P. Fleming wrote:
> I've created a page on wiki.asterisk.org outlining some changes
> we're proposing to make to the Asterisk release and support cycles.
> As always, before implementing any changes of this type, we'd like
> to collect some community feedback on the proposal.
> 
> The page is here:
> https://wiki.asterisk.org/wiki/x/5ggiAQ
> 
> Feel free to comment here, or on the page itself if you find any
> errors or inconsistencies in the page's content.

For the record: the next LTS (11) happens at a bad timing for Debian:
the freeze is at about the same time of the planned Debian freeze. Which
means that the release will just barely not make it, and we're probably
stuck with either 1.8 or 10 . Given that we'll need to maintain it for
~4 years, 1.8 is preffered to 10.

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Re: [asterisk-users] [NAT] SSH vs. OpenVPN?

[Jeff LaCoursiere]

On Thu, 2 Feb 2012, Tzafrir Cohen wrote:


Oh, and for the record, you can tunnel practically on top of anything.
Just in case you're not familiar with it: IP over DNS (which means you
don't even need direct access, and can use proxied DNS queries).
http://code.kryo.se/iodine/
I figure you won't get quality audio with that, though.



Don't forget RFC 1149!

j

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Re: [asterisk-users] [NAT] SSH vs. OpenVPN?

[Tzafrir Cohen]

On Tue, Jan 31, 2012 at 12:54:41PM +, Arthur Stanfield wrote:
> Hi Gilles,
> 
> You can't tunnel UDP through SSH. 

For the record: you can. But it's not really a good idea. Two options:

1. ssh -D: "dynamic" port forwarding. Which basically means that it
creates a socks4/socks5 proxy. You can now use e.g. sockify and connect
UDP-based programs over that connection.

2. ssh -w: create a tun device and create a tunnel on top of that (root
access of some sort is required).

That said, the ssh connection is TCP. The basic reasoning in
http://sites.inka.de/sites/bigred/devel/tcp-tcp.html applies to the VoIP
UDP payload as well.


Oh, and for the record, you can tunnel practically on top of anything.
Just in case you're not familiar with it: IP over DNS (which means you
don't even need direct access, and can use proxied DNS queries).
http://code.kryo.se/iodine/
I figure you won't get quality audio with that, though.

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Re: [asterisk-users] Asterisk version that support Database Configuration

[C. Savinovich]

If you go to a web site called google.com, and enter "asterisk version that support the ability to have the configuration in the database", and read the first search result, you will get your answer.Christian Savinovich


 Original Message 
Subject: [asterisk-users] Asterisk version that support Database
Configuration
From: bilal ghayyad 
Date: Thu, February 02, 2012 3:51 pm
To: asterisk-users@lists.digium.com

Hi All;

Which asterisk version that support the ability to have the configuration in the database?

Regards
Bilal

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Re: [asterisk-users] asterisk dahdi problem.

[Oğuzhan Kayhan]

 

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of 
> Tzafrir Cohen
> Sent: Thursday, February 02, 2012 10:38 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] asterisk dahdi problem.
> 
> On Thu, Feb 02, 2012 at 10:45:21AM +0200, Oguzhan Kayhan wrote:
> > Hi all,
> > I was using dahdi 1.6.2.0.9 version for a long time.
> > We decided to upgrade to 1.6.2.22 a few days ago.
> > After that we started to have some problems with dahdi channels.
> > PS:DAHDI Version: 2.6.0 Echo Canceller: HWEC, MG2
> > 
> > 
> > We have 2 PRIs between Ericsson pbx and asterisk and a sip 
> trunk for 
> > outside calls.
> > 
> > At begining everything works fine but in a few hours, calls from 
> > asterisk to ericsson stops with app_dial.c:1780 
> dial_exec_full: Unable 
> > to create channel of type 'DAHDI'
> > (cause 0 - Unknown)
> > 
> > 
> >  My call rules are as DAHDI/g1/
> 
> Is there a number missing here?

Rule is as DAHDI/g1/  i didnt write the whole , it was just to give an
opinion.


> 
> > Even i restart dahdi or asterisk, i keep on getting the same error. 
> > But meanwhile calls from ericsson to asterisk works fine. WHen i 
> > replace
> > DAHDI/g1 as DAHDI/1, it starts to work again.
> > 
> > Is there a problem with g?? Is it a known problem?
> > What might be the solution??
> 
> Do you actually have dahdi channels marked with group=1 ?
> 
> Can you provide your chan_dahdi.conf with ever file 
> #include-d into it?


Yes i have group=1. The same config was running with 1.6.2.0.9 and now i
upgraded to 1.8.8.2 and it works fine. I only got the problem with 1.6.2.22

And as i say, it starts to work fine at the begining in a few hours i start
to get UNKNOWN error msgs and it wont dissapear till i restart asterisk.


> 
> -- 
>Tzafrir Cohen
> icq#16849755  jabber:tzafrir.co...@xorcom.com
> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
> 
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>http://lists.digium.com/mailman/listinfo/asterisk-users
> 


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[asterisk-users] Asterisk version that support Database Configuration

[bilal ghayyad]

Hi All;

Which asterisk version that support the ability to have the configuration in 
the database?

Regards
Bilal

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Re: [asterisk-users] asterisk dahdi problem.

[Tzafrir Cohen]

On Thu, Feb 02, 2012 at 10:45:21AM +0200, Oguzhan Kayhan wrote:
> Hi all,
> I was using dahdi 1.6.2.0.9 version for a long time.
> We decided to upgrade to 1.6.2.22 a few days ago.
> After that we started to have some problems with dahdi channels.
> PS:DAHDI Version: 2.6.0 Echo Canceller: HWEC, MG2
> 
> 
> We have 2 PRIs between Ericsson pbx and asterisk and a sip trunk for
> outside calls.
> 
> At begining everything works fine but in a few hours, calls from asterisk
> to ericsson stops with
> app_dial.c:1780 dial_exec_full: Unable to create channel of type 'DAHDI'
> (cause 0 - Unknown)
> 
> 
>  My call rules are as DAHDI/g1/

Is there a number missing here?

> Even i restart dahdi or asterisk, i keep on getting the same error. But
> meanwhile calls from ericsson to asterisk works fine. WHen i replace
> DAHDI/g1 as DAHDI/1, it starts to work again.
> 
> Is there a problem with g?? Is it a known problem?
> What might be the solution??

Do you actually have dahdi channels marked with group=1 ?

Can you provide your chan_dahdi.conf with ever file #include-d into it?

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] SIP Provider Russia, Ukraine, Poland

[Joseph]

On 02/01/12 13:29, Christian Gansberger wrote:

Hello List!

I'm searching for SIP-Providers in the following countries:
Russia
Ukraine
Poland

I need a geographical number for each country, maybe a prepaid
SIP-Account, trunking is not important.
Has anyone some experience with these countries?

yours
christian


I have a DID from:
http://www.actio.pl/
(though you need to know polish to setup an account)
Their system has been working for me flawlessly for several years.
I can receive the call through them (that is all I need) but I can not make a 
call out as it is SIP and I don't have any ports open on my firewall.

--
Joseph

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Re: [asterisk-users] T38 faxing - UDPTL creation failed

[Danny Nicholas]

Agreed - I think the "solution" is a patch to udptl.c to reset the counter
instead of dying with this message (just my opinion).

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ""
Sent: Thursday, February 02, 2012 1:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] T38 faxing - UDPTL creation failed

well...I did so to avoid that lock again...but this is not solution...

> Maybe you should change your values in udptl.conf?   By default the range
> is
> 4000 to 4099, but is effectively 4001 to 4099 because the protocol 
> doesn't use even numbers by default, so it runs out of entries in 500
tries.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ""
> Sent: Thursday, February 02, 2012 1:35 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] T38 faxing - UDPTL creation failed
>
> Good question.
> I can't simulate it right now so I will try it today or so...But I'm 
> not expecting, that it help, because I tried couple of ways before and 
> restart was the only way.
>
> But thanks anyway. Any other ideas?
>
>
>> What happens if you do a "SIP RELOAD" instead of restarting Asterisk?
>>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ""
>> Sent: Thursday, February 02, 2012 1:14 PM
>> To: asterisk-users@lists.digium.com
>> Subject: [asterisk-users] T38 faxing - UDPTL creation failed
>>
>> Hello guys.
>>
>> When I am trying to send fax through T38 to linksys SPA (properly 
>> configured etc. - I have tried it with other systems), I'm getting 
>> error and fax is not delivered.
>>
>> I'm getting this errors in asterisk.log:
>> WARNING[687] udptl.c: No UDPTL ports remaining ERROR[687] chan_sip.c:
>> UDPTL
>> creation failed WARNING[687] udptl.c: No UDPTL ports remaining
>>
>> then, couple lines down:
>> WARNING[3514] chan_sip.c: Unsupported SDP media type in offer: image
>> 16400 udptl t38 WARNING[3514] chan_sip.c: Failing due to no 
>> acceptable offer found
>>
>> sip_general_custom.conf contains t38pt_udptl=yes
>>
>> udptl.conf contains:
>> [general]
>> udptlstart=4000
>> udptlend=4999
>> T38FaxUdpEC = t38UDPRedundancy
>>
>> Asterisk version is 1.8.5.0
>>
>> When I restart asterisk, everything is working good. Then, after some 
>> time, fax stop working.
>> Do you have any idea what it could be?
>>
>> Thanks in advance.
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 
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>>http://www.asterisk.org/hello
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>>
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>
>
>
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Re: [asterisk-users] T38 faxing - UDPTL creation failed

[�蝞������]

well...I did so to avoid that lock again...but this is not solution...

> Maybe you should change your values in udptl.conf?   By default the range
> is
> 4000 to 4099, but is effectively 4001 to 4099 because the protocol doesn't
> use even numbers by default, so it runs out of entries in 500 tries.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ""
> Sent: Thursday, February 02, 2012 1:35 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] T38 faxing - UDPTL creation failed
>
> Good question.
> I can't simulate it right now so I will try it today or so...But I'm not
> expecting, that it help, because I tried couple of ways before and restart
> was the only way.
>
> But thanks anyway. Any other ideas?
>
>
>> What happens if you do a "SIP RELOAD" instead of restarting Asterisk?
>>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ""
>> Sent: Thursday, February 02, 2012 1:14 PM
>> To: asterisk-users@lists.digium.com
>> Subject: [asterisk-users] T38 faxing - UDPTL creation failed
>>
>> Hello guys.
>>
>> When I am trying to send fax through T38 to linksys SPA (properly
>> configured etc. - I have tried it with other systems), I'm getting
>> error and fax is not delivered.
>>
>> I'm getting this errors in asterisk.log:
>> WARNING[687] udptl.c: No UDPTL ports remaining ERROR[687] chan_sip.c:
>> UDPTL
>> creation failed WARNING[687] udptl.c: No UDPTL ports remaining
>>
>> then, couple lines down:
>> WARNING[3514] chan_sip.c: Unsupported SDP media type in offer: image
>> 16400 udptl t38 WARNING[3514] chan_sip.c: Failing due to no acceptable
>> offer found
>>
>> sip_general_custom.conf contains t38pt_udptl=yes
>>
>> udptl.conf contains:
>> [general]
>> udptlstart=4000
>> udptlend=4999
>> T38FaxUdpEC = t38UDPRedundancy
>>
>> Asterisk version is 1.8.5.0
>>
>> When I restart asterisk, everything is working good. Then, after some
>> time, fax stop working.
>> Do you have any idea what it could be?
>>
>> Thanks in advance.
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
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>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>http://www.asterisk.org/hello
>>
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>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New
> to
> Asterisk? Join us for a live introductory webinar every Thurs:
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Re: [asterisk-users] T38 faxing - UDPTL creation failed

[Danny Nicholas]

Maybe you should change your values in udptl.conf?   By default the range is
4000 to 4099, but is effectively 4001 to 4099 because the protocol doesn't
use even numbers by default, so it runs out of entries in 500 tries.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ""
Sent: Thursday, February 02, 2012 1:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] T38 faxing - UDPTL creation failed

Good question.
I can't simulate it right now so I will try it today or so...But I'm not
expecting, that it help, because I tried couple of ways before and restart
was the only way.

But thanks anyway. Any other ideas?


> What happens if you do a "SIP RELOAD" instead of restarting Asterisk?
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ""
> Sent: Thursday, February 02, 2012 1:14 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] T38 faxing - UDPTL creation failed
>
> Hello guys.
>
> When I am trying to send fax through T38 to linksys SPA (properly 
> configured etc. - I have tried it with other systems), I'm getting 
> error and fax is not delivered.
>
> I'm getting this errors in asterisk.log:
> WARNING[687] udptl.c: No UDPTL ports remaining ERROR[687] chan_sip.c:
> UDPTL
> creation failed WARNING[687] udptl.c: No UDPTL ports remaining
>
> then, couple lines down:
> WARNING[3514] chan_sip.c: Unsupported SDP media type in offer: image 
> 16400 udptl t38 WARNING[3514] chan_sip.c: Failing due to no acceptable 
> offer found
>
> sip_general_custom.conf contains t38pt_udptl=yes
>
> udptl.conf contains:
> [general]
> udptlstart=4000
> udptlend=4999
> T38FaxUdpEC = t38UDPRedundancy
>
> Asterisk version is 1.8.5.0
>
> When I restart asterisk, everything is working good. Then, after some 
> time, fax stop working.
> Do you have any idea what it could be?
>
> Thanks in advance.
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
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>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
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Re: [asterisk-users] T38 faxing - UDPTL creation failed

[�蝞������]

Good question.
I can't simulate it right now so I will try it today or so...But I'm not
expecting, that it help, because I tried couple of ways before and restart
was the only way.

But thanks anyway. Any other ideas?


> What happens if you do a "SIP RELOAD" instead of restarting Asterisk?
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ""
> Sent: Thursday, February 02, 2012 1:14 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] T38 faxing - UDPTL creation failed
>
> Hello guys.
>
> When I am trying to send fax through T38 to linksys SPA (properly
> configured
> etc. - I have tried it with other systems), I'm getting error and fax is
> not
> delivered.
>
> I'm getting this errors in asterisk.log:
> WARNING[687] udptl.c: No UDPTL ports remaining ERROR[687] chan_sip.c:
> UDPTL
> creation failed WARNING[687] udptl.c: No UDPTL ports remaining
>
> then, couple lines down:
> WARNING[3514] chan_sip.c: Unsupported SDP media type in offer: image 16400
> udptl t38 WARNING[3514] chan_sip.c: Failing due to no acceptable offer
> found
>
> sip_general_custom.conf contains t38pt_udptl=yes
>
> udptl.conf contains:
> [general]
> udptlstart=4000
> udptlend=4999
> T38FaxUdpEC = t38UDPRedundancy
>
> Asterisk version is 1.8.5.0
>
> When I restart asterisk, everything is working good. Then, after some
> time,
> fax stop working.
> Do you have any idea what it could be?
>
> Thanks in advance.
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New
> to
> Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
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> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] T38 faxing - UDPTL creation failed

[Danny Nicholas]

What happens if you do a "SIP RELOAD" instead of restarting Asterisk?

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ""
Sent: Thursday, February 02, 2012 1:14 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] T38 faxing - UDPTL creation failed

Hello guys.

When I am trying to send fax through T38 to linksys SPA (properly configured
etc. - I have tried it with other systems), I'm getting error and fax is not
delivered.

I'm getting this errors in asterisk.log:
WARNING[687] udptl.c: No UDPTL ports remaining ERROR[687] chan_sip.c: UDPTL
creation failed WARNING[687] udptl.c: No UDPTL ports remaining

then, couple lines down:
WARNING[3514] chan_sip.c: Unsupported SDP media type in offer: image 16400
udptl t38 WARNING[3514] chan_sip.c: Failing due to no acceptable offer found

sip_general_custom.conf contains t38pt_udptl=yes

udptl.conf contains:
[general]
udptlstart=4000
udptlend=4999
T38FaxUdpEC = t38UDPRedundancy

Asterisk version is 1.8.5.0

When I restart asterisk, everything is working good. Then, after some time,
fax stop working.
Do you have any idea what it could be?

Thanks in advance.


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[asterisk-users] T38 faxing - UDPTL creation failed

[�蝞������]

Hello guys.

When I am trying to send fax through T38 to linksys SPA (properly
configured etc. - I have tried it with other systems), I'm getting error
and fax is not delivered.

I'm getting this errors in asterisk.log:
WARNING[687] udptl.c: No UDPTL ports remaining
ERROR[687] chan_sip.c: UDPTL creation failed
WARNING[687] udptl.c: No UDPTL ports remaining

then, couple lines down:
WARNING[3514] chan_sip.c: Unsupported SDP media type in offer: image 16400
udptl t38
WARNING[3514] chan_sip.c: Failing due to no acceptable offer found

sip_general_custom.conf contains t38pt_udptl=yes

udptl.conf contains:
[general]
udptlstart=4000
udptlend=4999
T38FaxUdpEC = t38UDPRedundancy

Asterisk version is 1.8.5.0

When I restart asterisk, everything is working good. Then, after some
time, fax stop working.
Do you have any idea what it could be?

Thanks in advance.


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Re: [asterisk-users] read digits during recording / DTMF in conference?

[Kingsley Tart]

Hi,

I'm not sure what you mean. Can you elaborate?

Cheers,
Kingsley.

On Thu, 2012-02-02 at 18:13 +0530, virendra bhati wrote:
> You may used even capturing in the case... when call  is recoding in
> conference
> 
> On Wed, Feb 1, 2012 at 4:04 PM, Kingsley Tart
>  wrote:
> Hi,
> 
> I want to create a system for incoming calls where, under some
> circumstances, callers get routed straight to voicemail (or
> some other
> means of recording a message) but if they enter a valid
> extension number
> then the recorded message would be abandoned and they'd be
> diverted to
> the extension number they entered.
> 
> I realise this can be done with the voicemail app with
> operator=yes but
> the problem with this is that the caller has to press 0 while
> the
> announcement is being played. If they're too slow and
> recording has
> started, they've missed the opportunity.
> 
> So I played around with ConfBridge and a couple of call files,
> just to
> see if I could get it to work. It's a bit convoluted but the
> idea is
> that the caller gets silently put into a conference, then two
> call files
> make asterisk silently connect to other calls into the same
> conference,
> with one doing the recording and the other using Read() to
> collect
> digits.
> 
> If I just had the caller and one of the other calls in the
> conference
> (the one doing Read()) then this worked - Read() managed to
> read the
> DTMF digits and assign them to a variable.
> 
> However, when the 'recording' call is also in the conference,
> the 'read'
> call can no longer recognise the DTMF digits. To test, I made
> the 'read'
> call play a sound before calling Read() and I could hear this
> being
> played so the call was definitely there. However, regardless
> of the
> number of digits I pressed, Read() didn't notice any of them,
> even if I
> introduced a delay so that the other channels were quiet
> before the call
> to Read().
> 
> I realise this might seem a bit like a mad solution but can
> anyone else
> think of a way to get Asterisk to read (and react to) DTMF
> digits during
> a recording?
> 
> This is with Asterisk 1.8.7.
> 
> --
> Cheers,
> Kingsley.
> 
> 
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> 
> 
> -- 
> 
> Thanks and regards
> 
>  Virendra Bhati
> +91-8885268942
> Software Engineer
> E-mail-: virbh...@gmail.com
> Skype id:- virbhati2
> 
> 
> 
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Kingsley.


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Re: [asterisk-users] Proposed changes to Asterisk release and support cycles

[Olivier]

2012/1/31 John Knight 

> Personally, I don't think what Digium is doing is necessarily a perfect
> approach (hey, what is?  we're all human), but they've vastly improved the
> quality of Asterisk from a support perspective.
>
>
I also agree that IMHO, Asterisk quality has vastly improved.
Though nothing replace "field experience", as a consequence, I feel much
more confident when upgrading from one version to another.

So if the previous plan which focuses development teams on fewer releases,
still rules that would be OK for me as long as new Asterisk versions are
carefully published (regression testing, ...).
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Re: [asterisk-users] Getting one way audio even NAT is configured

[Ahmed Munir]

Hi Warren,

Device A is behind NAT with regards to asterisk server. As far as localnet
statement first I did configured localnet = 130.8.2.0/255.255.255.0 as per
local network, after that made a SIP call and the message I'm getting is
listed below;

[Feb  2 11:14:52] WARNING[23868]: chan_sip.c:3622 retrans_pkt:
Retransmission timeout reached on transmission
OGEzODA0MzI4MWE1NzdiZDlkNmQ3NjYyMzJjYzYyOTY. for seqno 1 (Critical
Response) -- See
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[Feb  2 11:14:52] WARNING[23868]: chan_sip.c:3651 retrans_pkt: Hanging up
call OGEzODA0MzI4MWE1NzdiZDlkNmQ3NjYyMzJjYzYyOTY. - no reply to our
critical packet (see
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).

So after setting to 130.0.0.0/130.0.0.0 I wasn't getting the above warning
message but facing one way audio.

Date: Wed, 1 Feb 2012 14:38:01 -0600
> From: Warren Selby 
> Subject: Re: [asterisk-users] Getting one way audio even NAT is
>configured
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Message-ID:
> >
> Content-Type: text/plain; charset="iso-8859-1"
>
> On Wed, Feb 1, 2012 at 1:16 PM, Ahmed Munir 
> wrote:
>
> > Hi all,
> >
> > I'm getting one way audio when calling over the SIP trunk i.e. end device
> > B (remote end of SIP trunk) can hear device A (softphone registered with
> > Asterisk) but device A can't hear device B. Even though I configured same
> > NAT configurations on other servers and they are working good. The NAT
> > configuration is listed below;
> >
> > localnet=130.0.0.0/130.0.0.0
> > externhost=12.131.12.13
> > externrefresh=10
> > fromdomain=test.localhost.com
> > nat=yes
> > qualify=yes
> > canreinvite=no
> >
> >
> > NAT on device end i.e. my softphone (extension) has already set to yes
> > with canreinvite=no  but still unable to resolve this issue. SIP traces
> are
> > listed below;
> >
> >
> 
>
>
> >
> > The Asterisk version I'm using is 1.8.5. Please assist me at earliest.
> >
>
> Which device (A or B) is behind NAT with regards to your asterisk server?
> Is that the actual localnet= statement you're using, because to my
> understanding that is not the proper format to use (should be
> localnet=x.x.x.x/y.y.y.y where x.x.x.x is your actual local network, and
> y.y.y.y is your subnet for your local network).
>
> --
> Thanks,
> --Warren Selby, dCAP
> http://www.SelbyTech.com 
>

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Ahmed Munir Chohan
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Re: [asterisk-users] FXS hangup issues

[Richard Mudgett]

> I currently have an Asterisk 1.8.8.1 system set up with SIP accounts
> as well as a Wildcard TDM400P REV I card with both FXS and FXO
> ports - FXO is connected to outside lines, FXS connected to inside
> analog phones. Everything about the setup works fine except one thing
> -
> after making calls to or from any of the analog phones, and the other
> side hangs up, the analog phone just gives a busy signal instead of
> hanging up. On the Asterisk console, it seems to think it's hung up
> the phone too:
> 
> == Spawn extension (from-office, 44, 50005) exited non-zero on
> 'DAHDI/5-1'
> -- Hanging up on 'DAHDI/5-1'
> -- Hungup 'DAHDI/5-1'
> 
> chan_dahdi.conf is mostly just the default with just the lines
> defined, nothing too fancy, and this doesn't happen for SIP clients
> or remote
> phones via the FXO ports.
> 
> Any ideas?

You have to hang up the phone too.

Richard

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[asterisk-users] amd detect answering machine

[Etann]

Hi,
I have IVR and when I press 1, asterisk calls my mobile phone.
If my mobile phone is offline, asterisk transfers to asterisk voicemail.
I'd like asterisk detects my mobile voicemail and if my mobile voicemail 
answers, asterisk transfers to asterisk voicemail.
For that, I used AMD.
So I have problems ! Asterisk detects answering machine everytime!
How do I do please ?


extensions.conf
[ServeurPro]
exten => s,1,Ringing() 
exten => s,2,Wait(2)
exten => s,3,Answer()  
exten => s,4,Set(NbInvalide=0)
exten => s,5,Set(NbEssai=0)
exten => s,6,background(${ChmAudio}/ServeurProBienvenu)
exten => s,7,WaitExten(2)

exten => 1,1,AMD()
exten => 1,2,GotoIf($["${AMDSTATUS}"="MACHINE"]?1,4)
exten => 1,3,Dial(SIP/@ippi_outgoing2,40,r)
exten => 1,4,Voicemail(801@FloriePro,us)
exten => i,1,Set(NbInvalide=$[${NbInvalide}+1]})
exten => i,2,Gotoif($["${NbInvalide}" < "3"]?:6)
exten => i,3,Playback(${ChmAudio}/ErreurSaisie) 
exten => i,4,Playback(${ChmAudio}/RetourMenu) 
exten => i,5,Goto(s,6)
exten => i,6,Playback(${ChmAudio}/ErreurSaisie)
exten => i,7,Playback(${ChmAudio}/Aurevoir)
exten => i,8,Hangup()
exten => t,1,Set(NbEssai=$[${NbEssai}+1])
exten => t,2,Gotoif($["${NbEssai}" < "3"]?:5)
exten => t,3,Playback(${ChmAudio}/DemandeIncomprise) 
exten => t,4,Goto(s,6)
exten => t,5,PlayBack(${ChmAudio}/Aurevoir)
exten => t,6,Hangup()
exten => h,1,noOp("Statut AMD : "${AMDSTATUS})


amd.conf
[general]
initial_silence = 2500  ; Maximum silence duration before the greeting.
; If exceeded then MACHINE.
greeting = 1500   ; Maximum length of a greeting. If exceeded then MACHINE.
after_greeting_silence = 500 ; Silence after detecting a greeting.
; If exceeded then HUMAN
total_analysis_time = 5000 ; Maximum time allowed for the algorithm to decide
; on a HUMAN or MACHINE
min_word_length = 120  ; Minimum duration of Voice to considered as a word
between_words_silence = 50 ; Minimum duration of silence after a word to 
consider
; the audio what follows as a new word
maximum_number_of_words = 3 ; Maximum number of words in the greeting.
; If exceeded then MACHINE
silence_threshold = 256

Thank you for your reply and for help!


 
AMICALEMENT
Manu

SITES WEBS
Mon site web Officiel (Manu-dpk.net)
Ecoutez Radio DPK

CONTACT
- E-mail : manuli...@manu-dpk.net
- Messenger (WLM) : m...@manu-dpk.net
- Skype : manu-dpk






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Re: [asterisk-users] Is this doable?

[Josh]

I think you might want to split your questions first.
I thought that instead of creating a dozen different threads (and 
clogging the ML in the process) it would be better to put everything 
into one place - just pick the issue (or issues) you could address and 
leave (i.e. delete) the rest out.


1. You can't have multiple externip, but it's not necessary to run two 
Asterisk instances, because you can set routes to different 
destinations via particular interfaces.
I have no problems with the routing - that is already done. I am not 
certain how Asterisk handles a stream running across multiple interfaces 
and how the packet NAT is done. I am also aware that SIP packets "embed" 
the IP address in it so not sure how this is handled either.



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Re: [asterisk-users] Is this doable?

[Aurimas Skirgaila]

I think you might want to split your questions first.


this might work from local ISP network, but in my experience it might
depend on provider.

1. You can't have multiple externip, but it's not necessary to run two
Asterisk instances, because you can set routes to different destinations
via particular interfaces.





On Thu, Feb 2, 2012 at 2:48 AM, Josh  wrote:

> I am trying to configure Asterick, having the following system setup on
> the Asterick server:
>
> * eth0 faces the external Internet interface, *but* it does not have IP
> address (it has a private one given to it by my ISP's DHCP server);
> * eth1 faces my internal network (say 10.1.1.0/24);
> * tun0 serves all mobile smartphones and connects to the internal
> network (it has a different ip range, say 10.1.2.0/24) - they are all
> connected via the Internet using OpenVPN;
>
> I would like to configure Asterick for internal calls between ourselves
> (eth1<->tun0) and I think I have no problem with configuring this part.
> I would also like to use one external VOIP provider to which Asterick
> registers on startup. I think I know how to do that and use the
> "register" option in sip.conf, though I am not sure for the rest of the
> NAT-related entries (see below).
>
> The purpose of registering this external account is so that both the
> smart phones (tun0) and the internal net (eth1) users could use this
> account to make external calls (starting with "0", i.e "_0[0-9]."
> pattern in extensioins.conf). Obviously, I need these calls to be routed
> properly via the external VOIP account. In addition to that, I would
> also need to receive calls from that external account to a nominated
> internal one (say on extension 20).
>
> Is this achievable?
>
> If so, I am not completely clear on whether I need to explicitly specify
> my public IP address (via externip/externhost) or whether Asterick is
> able to find it without this option? If not, then my plan is to use
> external program to find it and then use a script in Asterick to set it
> up as an environment variable. Would that work? That external IP address
> is going to change, but only in rare circumstances and in such cases I
> have to restart a lot of stuff (including Asterick) on that server (this
> is usually triggered by a monitoring program), so it won't be a problem
> once it is setup initially. I am also not sure whether to specify
> "nat=yes" or just have "nat=route" only - any ideas?
>
> Is there a comprehensive list of all the options available in sip.conf
> and what they do, because I was unable to find such a list?
>
> If the above is doable, I would also like to add the following 2 features:
>
> 1. Secondary external VOIP account, though I have no idea how to specify
> its port in "register" (it uses port 5065 instead of the standard 5060).
> That account would need to be used on a separate interface (eth2) with a
> different public IP address. Would it be possible to use
> externip/externhost inside that external account section to specify it?
> If this is not possible, then I am thinking of running a separate
> instance of Asterick with the second VOIP account/public IP address set
> up - would that work?
>
> 2. I would like to be able to configure the following work flow: for a
> specific set of (external) calling numbers (including where no Caller ID
> is available):
> a) these callers to be prompted to specify the "reason" for their call;
> b) their response to be temporarily "recorded"/stored (a short message
> of, say no more than 10 seconds long or when they press '#' for that
> recording to stop);
> c) Asterick then rings the nominated number for external VOIP calls
> (extension 20) and play that recorded message back;
> d) then asks for one of four possible outcomes:
> - accept this call (pressing, say 1) in which case the call is connected
> as normal;
> - reject it with a message that that number/person is "unavailable"
> (say, by pressing 0);
> - ask the caller to leave a message by transferring them to a voicemail
> (say by pressing 2); or
> - end the initial call completely with a message that the caller/number
> has been "blacklisted" (say, by pressing the 9 key);
>
> Could this be achieved?
>
> One final question about binding: in order to be able to use both tun0
> and eth1 interfaces so that Asterick serves the calls from both eth1 and
> tun0, do I have to use "bind 0.0.0.0"? Is there an alternative, like
> specifying "bind 10.1.1.1" for eth1 and then "bind 10.1.2.1" for the
> tun0 interface - is this possible?
>
> Many thanks in advance!
>
>
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Re: [asterisk-users] Is this doable?

[Josh]

Whats asterick?
  

I blame my spell checker! :-P

Do you have anything to offer in terms of help or advice on the 
issues/questions I posted?


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Re: [asterisk-users] externip nat audio sip trunk issue problem

[Gabriel Ortiz Lour]

As soon as I activate the exterip/localnet config there is no response at
all, as if that IP address desappeared.
Any packets send to it simply get no response.

I've considered being linux kernel routing issue, but since without the
exterip/localnet config it works OK I don't think this is the case.

I could put the tcpdump/sip debug info here, but it would be like a
monologue, with only the packets being sent showing without any response.
But I can put it here if it helps...

Didn't anyone had this problem with this config option?


2012/2/2 C F 

> On Wed, Feb 1, 2012 at 9:14 PM, Gabriel Ortiz Lour
>  wrote:
> > Hi all,
> >
> >   I've tried search this problem on the list... no luck...
> >
> >   The case is:
> >
> > without externip/localnet config on sip.conf [general] my SIP trunk
> works,
> > but with no audio NAT problem (asterisk sends the private 192 address to
> the
> > outside...)
> >
> > when I configure externip/localnet correctly my SIP trunk simply
> disappear!
> > Checking the signalling with tcpdump shows me that Im sending the
> packets to
> > the correct SIP trunk IP but there is no response AT ALL from it...
>
> Can you explain this?
> What do you mean no response? Is it registering? Do you have a debug
> output?
>
> >
> > Anyone had this problem?
> >
> > Thanks,
> > Gabriel
> >
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Re: [asterisk-users] read digits during recording / DTMF in conference?

[virendra bhati]

You may used even capturing in the case... when call  is recoding in
conference

On Wed, Feb 1, 2012 at 4:04 PM, Kingsley Tart wrote:

> Hi,
>
> I want to create a system for incoming calls where, under some
> circumstances, callers get routed straight to voicemail (or some other
> means of recording a message) but if they enter a valid extension number
> then the recorded message would be abandoned and they'd be diverted to
> the extension number they entered.
>
> I realise this can be done with the voicemail app with operator=yes but
> the problem with this is that the caller has to press 0 while the
> announcement is being played. If they're too slow and recording has
> started, they've missed the opportunity.
>
> So I played around with ConfBridge and a couple of call files, just to
> see if I could get it to work. It's a bit convoluted but the idea is
> that the caller gets silently put into a conference, then two call files
> make asterisk silently connect to other calls into the same conference,
> with one doing the recording and the other using Read() to collect
> digits.
>
> If I just had the caller and one of the other calls in the conference
> (the one doing Read()) then this worked - Read() managed to read the
> DTMF digits and assign them to a variable.
>
> However, when the 'recording' call is also in the conference, the 'read'
> call can no longer recognise the DTMF digits. To test, I made the 'read'
> call play a sound before calling Read() and I could hear this being
> played so the call was definitely there. However, regardless of the
> number of digits I pressed, Read() didn't notice any of them, even if I
> introduced a delay so that the other channels were quiet before the call
> to Read().
>
> I realise this might seem a bit like a mad solution but can anyone else
> think of a way to get Asterisk to read (and react to) DTMF digits during
> a recording?
>
> This is with Asterisk 1.8.7.
>
> --
> Cheers,
> Kingsley.
>
>
> --
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>



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Thanks and regards

 Virendra Bhati
+91-8885268942
Software Engineer
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
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Re: [asterisk-users] Router that support Asterisk

[Gilles]

On Wed, 01 Feb 2012 18:47:49 -0500, James Sharp 
wrote:
>The Cisco DDR2200 that I just got from Centurylink for DSL appears to be 
>just that.  I haven't tested the FXS ports on it yet, though.

"Cisco announces the end-of-sale and end-of-life dates for the Cisco
DDR2200, DDR2201, and WAG310G ADSL2+ Residential Gateways. "

www.cisco.com/en/US/prod/collateral/video/ps8611/ps9520/ps9524/end_of_life_notice_c51-694180.html


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[asterisk-users] MixMonitor and ChanSpy

[Jonas Kellens]

Hello,

ChanSpy can not be used on a  Channel that is being recorded with 
MixMonitor.


How can I verify if a channel which I want to spy on, is currently not 
being recorded ?!




Kind regards,
Jonas.
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[asterisk-users] asterisk dahdi problem.

[Oguzhan Kayhan]

Hi all,
I was using dahdi 1.6.2.0.9 version for a long time.
We decided to upgrade to 1.6.2.22 a few days ago.
After that we started to have some problems with dahdi channels.
PS:DAHDI Version: 2.6.0 Echo Canceller: HWEC, MG2


We have 2 PRIs between Ericsson pbx and asterisk and a sip trunk for
outside calls.

At begining everything works fine but in a few hours, calls from asterisk
to ericsson stops with
app_dial.c:1780 dial_exec_full: Unable to create channel of type 'DAHDI'
(cause 0 - Unknown)


 My call rules are as DAHDI/g1/
Even i restart dahdi or asterisk, i keep on getting the same error. But
meanwhile calls from ericsson to asterisk works fine. WHen i replace
DAHDI/g1 as DAHDI/1, it starts to work again.

Is there a problem with g?? Is it a known problem?
What might be the solution??

Thank you.



PS: While i was monitoring it started to give same error with dahdi/1
config too.

I got no alarm on dahdi channels or ericsson side.
dahdi show channels are

  4DID_span_1 default 
   In Service

core show calls
12 active calls


and


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Re: [asterisk-users] asterisk 10.0 realtime

[Andrew Nowrot]

On 1 February 2012 21:26, Andrzej Nowrot  wrote:
> Hi
>
> I have noticed new behaviour of asterisk 10.0 realtime.
> In 1.6 when I was using realtime:
>
> """
> [somecontext]
>
>  exten => someexten1..
>  exten => someexten2..
>  exten => someexten3..
>  exten => someexten4..
>
> switch => Realtime/${CONTEXT}@extensions
> """
>
> switch statement was executed after lines above (so there was a
> precedence of the lines declared in a extensions.conf over the ones in
> database).
>
> In asterisk 10.0 switch is executed before extens declared in the
> extensions.conf file.
>
> Is there a way to change that and have previous behaviour?
>
>
> Cheers

Never mind. I works OK (the same as in 1.6) I was my mistake.

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