Re: [asterisk-users] one way audio for xlite clients behind NAT

[Flavio Miranda]

If you are using linux firewall, try this, it was very usefull to me:


iptables -t nat -A PREROUTING -i ethx -p tcp --dport 5060 -j DNAT --to 
ip_phoneiptables -t nat -A PREROUTING -i ethx -p udp --dport 5060 -j DNAT --to 
iip_phoneiptables -A FORWARD -p TCP --dport 5060 -j ACCEPTiptables -A FORWARD 
-p UDP --dport 5060 -j ACCEPT



Att,

 

Flavio Roberto Miranda

MSN:flaviormira...@hotmail.com
Skype: flaviormiranda



> Date: Thu, 16 Sep 2010 18:45:38 -0400
> From: paul.belan...@polybeacon.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] one way audio for xlite clients behind NAT
> 
> On Thu, Sep 16, 2010 at 6:04 PM, Thomas Johnson  wrote:
> > The server is not behind NAT only the client above is
> >
> Sounds like a phone (not asterisk) issue then, make sure you have
> setup your NAT and port forwarding properly on the client side.
> 
> -- 
> Paul Belanger | dCAP
> Polybeacon | Consultant
> Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
> blog.polybeacon.com
> 
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Re: [asterisk-users] one way audio for xlite clients behind NAT

[Paul Belanger]

On Thu, Sep 16, 2010 at 6:04 PM, Thomas Johnson  wrote:
> The server is not behind NAT only the client above is
>
Sounds like a phone (not asterisk) issue then, make sure you have
setup your NAT and port forwarding properly on the client side.

-- 
Paul Belanger | dCAP
Polybeacon | Consultant
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blog.polybeacon.com

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Re: [asterisk-users] one way audio for xlite clients behind NAT

[Thomas Johnson]

I already have that covered

[tomfmason]
type=friend
secret=secret
callerid="Thomas Johnson" 
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=gsm
allow=ulaw
allow=alaw
qualify=yes
context=sip

The server is not behind NAT only the client above is

On Thu, Sep 16, 2010 at 4:59 PM, Paul Belanger  wrote:

> On Thu, Sep 16, 2010 at 5:50 PM, Thomas Johnson 
> wrote:
> > Also, if I disable the firewall in my router I lose incoming audio and
> > outgoing audio works.
> >
> http://www.aocomputing.net/?p=3
>
> --
> Paul Belanger | dCAP
> Polybeacon | Consultant
> Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
> blog.polybeacon.com
>
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Re: [asterisk-users] one way audio for xlite clients behind NAT

[Paul Belanger]

On Thu, Sep 16, 2010 at 5:50 PM, Thomas Johnson  wrote:
> Also, if I disable the firewall in my router I lose incoming audio and
> outgoing audio works.
>
http://www.aocomputing.net/?p=3

-- 
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Re: [asterisk-users] one way audio for xlite clients behind NAT

[Thomas Johnson]

I have tried doing that with just ulaw and alaw, respectively, and nothing
changed

Also, if I disable the firewall in my router I lose incoming audio and
outgoing audio works.



On Thu, Sep 16, 2010 at 2:50 PM, Sebastian  wrote:

>
>
> On 09/16/2010 07:59 PM, Thomas Johnson wrote:
> > the client that is behind nat is
> > [tomfmason]
> > type=friend
> > secret=secret
> > callerid="Thomas Johnson" 
> > host=dynamic
> > nat=yes
> > canreinvite=no
> > disallow=all
> > allow=gsm
> > allow=ulaw
> > allow=alaw
> > qualify=yes
> > context=sip
> >
> > do I have to enable nat on all of them?
>
> I don't think so. It's just that you didn't specify which client is which.
>
> My next guess is that it is a codec problem. I've had similar problems -
> and upon checking Asterisk logs - I discovered that the client and
> Asterisk weren't agreeing correctly on codecs. Can you double-check your
> X-lite configuration - and maybe try to ulaw or alaw as the only codec
> at both ends?
>
> Sebastian
>
> > On Thu, Sep 16, 2010 at 1:36 PM, Sebastian  > > wrote:
> >
> >
> >
> > On 09/16/2010 06:58 PM, Thomas Johnson wrote:
> >  > I am having a one way audio issue with xlite clients behind NAT.
> They
> >  > can connect to the server and make calls but no audio is heard on
> the
> >  > other end.
> >  >
> >  > my sip conf
> >  >
> >  > [general]
> >  > context=default
> >  > bindport=5060
> >  > bindaddr=0.0.0.0
> >  > srvlookup=yes
> >  > canreinvite=no
> >  >
> >  > [tomfmason]
> >  > type=friend
> >  > secret=secret
> >  > callerid="Thomas Johnson" 
> >  > host=dynamic
> >  > nat=yes
> >  > canreinvite=no
> >  > disallow=all
> >  > allow=gsm
> >  > allow=ulaw
> >  > allow=alaw
> >  > qualify=yes
> >  > context=sip
> >  >
> >  > [1001];Work
> >  > type=peer
> >  > dtmfmode=rfc2833
> >  > context=sip
> >  > insecure=very
> >  > host=sip.domain.com  <
> http://sip.domain.com>
> >  > nat=no
> >  >
> >  > [1000];IPKall
> >  > type=peer
> >  > dtmfmode=rfc2833
> >  > context=sip
> >  > insecure=very
> >  > host=voiper.ipkall.com 
> > 
> >  > nat=no
> >
> > You seem to be using nat=no
> >
> > shouldn't that be nat=yes?
> >
> >  >
> >  >
> >  >
> >  > I pasted the log here -> http://pastie.org/1163238
> >  >
> >  >
> >  > I have tried connecting both of the clients to another sip
> > service(sip2sip.info  )
> > and did not have the same problems.
> >  >
> >  >
> >  > Any suggestions would be great.
> >  >
> >  > Thanks,
> >  >
> >  > Tom
> >  >
> >
> > --
> > _
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com--
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> >
> >
>
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Re: [asterisk-users] AGI Delimiter in 1.6

[Barry Miller]

On Thu, Sep 16, 2010 at 07:44:23PM +0100, Jon Farmer wrote:
> Hi
> 
> I am currently using 1.2.x and 1.4.x behind OpenSER. One of the things
> I do on INVITES is to re-authenticate the user from OpenSER. Then when
> the INVITE gets passed to Asterisk I capture the AUTH to a variable in
> the dialplan and pass to an AGI script. I am now trying to set the
> same thing up in 1.6 However because the argument delimter in 1.6 has
> changed from pipe to comma this breaks as the AUTH line is also comma
> delimited. Thus the AGI sees the AUTH as extra arguments instead of a
> single argument. As the AUTH may contain varying number of arguments I
> need a new way for a my AGI to access this data.
> 
> Does anyone have any ideas how I might go about this?

For an interim fix, setting res_agi=1.4 in the [compat] section of
asterisk.conf should work.  See UPGRADE-1.6.txt .

-- 
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Re: [asterisk-users] AGI Delimiter in 1.6

[James A. Shigley]

So why can't you send the Auth line into the variable and then have your script 
do the parsing to break out the segments you want. 

Or if need be two scripts. The first can accept the authline as a full string 
from a variable and break it down to its parts and save those as channel 
variables. Then your second agi script which is basically the one that worked 
in 1.2/1.4 can use the channel variables.

James Shigley


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jon Farmer
Sent: Thursday, September 16, 2010 2:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] AGI Delimiter in 1.6

> On 16 September 2010 19:50, Danny Nicholas  wrote:

> If you make the string into a dialplan Variable, you can do pretty much
> anything with it.  Let's say your dialplan is like this
>
> - exten => 1234,1,blah
> - exten => 1234,n,AGI(myagi.xx,"1234")
>
> Change line 2 to
> - exten => 1234,n,AGI(myagi.xx,${VARNAME})
>
> Then you just "do your magic" on ${VARNAME}


Yes, but the problem is I am trying to pass the whole AUTH line which
is key=value pairs seperated by commas. e.g. username=myusername,
domain=mydomain

This breaks when passing to an AGI in 1.6.



-- 
Jon Farmer
Tel 07795 118140

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Re: [asterisk-users] one way audio for xlite clients behind NAT

[Sebastian]

On 09/16/2010 07:59 PM, Thomas Johnson wrote:
> the client that is behind nat is
> [tomfmason]
> type=friend
> secret=secret
> callerid="Thomas Johnson" 
> host=dynamic
> nat=yes
> canreinvite=no
> disallow=all
> allow=gsm
> allow=ulaw
> allow=alaw
> qualify=yes
> context=sip
>
> do I have to enable nat on all of them?

I don't think so. It's just that you didn't specify which client is which.

My next guess is that it is a codec problem. I've had similar problems - 
and upon checking Asterisk logs - I discovered that the client and 
Asterisk weren't agreeing correctly on codecs. Can you double-check your 
X-lite configuration - and maybe try to ulaw or alaw as the only codec 
at both ends?

Sebastian

> On Thu, Sep 16, 2010 at 1:36 PM, Sebastian  > wrote:
>
>
>
> On 09/16/2010 06:58 PM, Thomas Johnson wrote:
>  > I am having a one way audio issue with xlite clients behind NAT. They
>  > can connect to the server and make calls but no audio is heard on the
>  > other end.
>  >
>  > my sip conf
>  >
>  > [general]
>  > context=default
>  > bindport=5060
>  > bindaddr=0.0.0.0
>  > srvlookup=yes
>  > canreinvite=no
>  >
>  > [tomfmason]
>  > type=friend
>  > secret=secret
>  > callerid="Thomas Johnson" 
>  > host=dynamic
>  > nat=yes
>  > canreinvite=no
>  > disallow=all
>  > allow=gsm
>  > allow=ulaw
>  > allow=alaw
>  > qualify=yes
>  > context=sip
>  >
>  > [1001];Work
>  > type=peer
>  > dtmfmode=rfc2833
>  > context=sip
>  > insecure=very
>  > host=sip.domain.com  
>  > nat=no
>  >
>  > [1000];IPKall
>  > type=peer
>  > dtmfmode=rfc2833
>  > context=sip
>  > insecure=very
>  > host=voiper.ipkall.com 
> 
>  > nat=no
>
> You seem to be using nat=no
>
> shouldn't that be nat=yes?
>
>  >
>  >
>  >
>  > I pasted the log here -> http://pastie.org/1163238
>  >
>  >
>  > I have tried connecting both of the clients to another sip
> service(sip2sip.info  )
> and did not have the same problems.
>  >
>  >
>  > Any suggestions would be great.
>  >
>  > Thanks,
>  >
>  > Tom
>  >
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>
>

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Re: [asterisk-users] AGI Delimiter in 1.6

[Jon Farmer]

> On 16 September 2010 19:50, Danny Nicholas  wrote:

> If you make the string into a dialplan Variable, you can do pretty much
> anything with it.  Let's say your dialplan is like this
>
> - exten => 1234,1,blah
> - exten => 1234,n,AGI(myagi.xx,"1234")
>
> Change line 2 to
> - exten => 1234,n,AGI(myagi.xx,${VARNAME})
>
> Then you just "do your magic" on ${VARNAME}


Yes, but the problem is I am trying to pass the whole AUTH line which
is key=value pairs seperated by commas. e.g. username=myusername,
domain=mydomain

This breaks when passing to an AGI in 1.6.



-- 
Jon Farmer
Tel 07795 118140

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Re: [asterisk-users] AGI Delimiter in 1.6

[Danny Nicholas]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jon Farmer
Sent: Thursday, September 16, 2010 2:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] AGI Delimiter in 1.6

On 16 September 2010 19:50, Danny Nicholas  wrote:

> Two suggestions;
> #1.  "escape" the , as \,
> #2.  quote the string so 1,2,3 is "1,2,3"


I have thought about both of those ideas.

Is it possible to escape the string in the dialplan?

Applying quotes didn't seem to work, however I was pretty tired when I
tried so it might just need a fresh set of eyes.

Regards

Jon

If you make the string into a dialplan Variable, you can do pretty much
anything with it.  Let's say your dialplan is like this

- exten => 1234,1,blah
- exten => 1234,n,AGI(myagi.xx,"1234")

Change line 2 to 
- exten => 1234,n,AGI(myagi.xx,${VARNAME})

Then you just "do your magic" on ${VARNAME}



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Re: [asterisk-users] AGI Delimiter in 1.6

[Jon Farmer]

On 16 September 2010 19:50, Danny Nicholas  wrote:

> Two suggestions;
> #1.  "escape" the , as \,
> #2.  quote the string so 1,2,3 is "1,2,3"


I have thought about both of those ideas.

Is it possible to escape the string in the dialplan?

Applying quotes didn't seem to work, however I was pretty tired when I
tried so it might just need a fresh set of eyes.

Regards

Jon


-- 
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Tel 07795 118140

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Re: [asterisk-users] Purpose of qualify=yes

[Sebastian]

Hi,

On 09/16/2010 05:28 PM, Gareth Blades wrote:
> One of the main benefits of qualify=yes is to detect network problems
> with peers.
> We send a lot of calls via a service provider using SIP but we have
> qualify-yes set so that if it becomes unreachable the dial fails
> immediatly without having to wait for a timeout which enables us to
> seamlessly failover to an ISDN connection.

In have some experience with IAX qualify. Might not be of direct benefit 
though to this particular thread. I have the following IAX peer as the 
outgoing IAX trunk at one of my sites:

[outgoing_trunk]
type=peer
host=iax-out.our_iax_host.net
username=my_username
auth=plaintext
secret=my_password
qualify=no

To start with, I used qualify=yes. Which worked well for about 6 months. 
But after that, the peer would become unavailable every few days - and 
nobody could do outgoing calls. I would login remotely, and check that I 
can ping ok the IAX provider. However, Asterisk kept on showing the peer 
as unreachable even though it was clearly online. It looks to me like, 
from time to time, the IAX provider would become unavailable for a 
period of time (maybe maintenance at night) and Asterisk would just give 
up after some retries, and mark it as unreachable. The problem is that 
Asterisk wouldn't change its mind unless I did a full Asterisk restart.

After I turned off the qualify, no more problems. It looks like Asterisk 
is searching now for the IAX provider every time a call is attempted - 
so it doesn't matter if they are unavailable for a period of time.

At least this is the conclusions I have reached. So it seems that 
sometimes qualify is not a good idea.

Sebastian


>
> Chris Owen wrote:
>> We have a tenant who has been having issues with a congested connection and 
>> in trouble shooting it we've noticed that there seems to be a lot of SIP 
>> traffic even when none of the phones are doing anything.
>>
>> We've determined that this traffic is mostly INFO packets generated by 
>> setting qualify=2000.   I understand that 2000 ms is the default value for 
>> the qualification parameter but what I'm unclear on is exactly what the 
>> purpose of having asterisk qualify the phones is.
>>
>> I know that in a NAT situation, qualifications can help keep UDP sessions 
>> open in the firewall but in our case most phones are not behind NAT.
>>
>> I realize qualifying phones is also how asterisk keeps track of who is 
>> available for things like BLF but surely it doesn't need to do that every 2 
>> seconds to keep the BLFs reasonably current.
>>
>> So I guess my question is what is the real purpose of the qualify setting in 
>> a non-NAT situation and can one safely set the qualification as something 
>> higher.   I'd think something like 15 seconds would be more than enough for 
>> BLFs and the like.
>>
>> Chris
>>
>>
>>
>
>

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Re: [asterisk-users] one way audio for xlite clients behind NAT

[Thomas Johnson]

the client that is behind nat is
[tomfmason]
type=friend
secret=secret
callerid="Thomas Johnson" 
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=gsm
allow=ulaw
allow=alaw
qualify=yes
context=sip

do I have to enable nat on all of them?
On Thu, Sep 16, 2010 at 1:36 PM, Sebastian  wrote:

>
>
> On 09/16/2010 06:58 PM, Thomas Johnson wrote:
> > I am having a one way audio issue with xlite clients behind NAT. They
> > can connect to the server and make calls but no audio is heard on the
> > other end.
> >
> > my sip conf
> >
> > [general]
> > context=default
> > bindport=5060
> > bindaddr=0.0.0.0
> > srvlookup=yes
> > canreinvite=no
> >
> > [tomfmason]
> > type=friend
> > secret=secret
> > callerid="Thomas Johnson"  
> > host=dynamic
> > nat=yes
> > canreinvite=no
> > disallow=all
> > allow=gsm
> > allow=ulaw
> > allow=alaw
> > qualify=yes
> > context=sip
> >
> > [1001];Work
> > type=peer
> > dtmfmode=rfc2833
> > context=sip
> > insecure=very
> > host=sip.domain.com  
> > nat=no
> >
> > [1000];IPKall
> > type=peer
> > dtmfmode=rfc2833
> > context=sip
> > insecure=very
> > host=voiper.ipkall.com  
> > nat=no
>
> You seem to be using nat=no
>
> shouldn't that be nat=yes?
>
> >
> >
> >
> > I pasted the log here ->  http://pastie.org/1163238
> >
> >
> > I have tried connecting both of the clients to another sip service(
> sip2sip.info  ) and did not have the same problems.
> >
> >
> > Any suggestions would be great.
> >
> > Thanks,
> >
> > Tom
> >
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] AGI Delimiter in 1.6

[Danny Nicholas]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jon Farmer
Sent: Thursday, September 16, 2010 1:44 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] AGI Delimiter in 1.6

Hi

I am currently using 1.2.x and 1.4.x behind OpenSER. One of the things
I do on INVITES is to re-authenticate the user from OpenSER. Then when
the INVITE gets passed to Asterisk I capture the AUTH to a variable in
the dialplan and pass to an AGI script. I am now trying to set the
same thing up in 1.6 However because the argument delimter in 1.6 has
changed from pipe to comma this breaks as the AUTH line is also comma
delimited. Thus the AGI sees the AUTH as extra arguments instead of a
single argument. As the AUTH may contain varying number of arguments I
need a new way for a my AGI to access this data.

Does anyone have any ideas how I might go about this?

Regards

Jon

Two suggestions; 
#1.  "escape" the , as \,
#2.  quote the string so 1,2,3 is "1,2,3"


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[asterisk-users] AGI Delimiter in 1.6

[Jon Farmer]

Hi

I am currently using 1.2.x and 1.4.x behind OpenSER. One of the things
I do on INVITES is to re-authenticate the user from OpenSER. Then when
the INVITE gets passed to Asterisk I capture the AUTH to a variable in
the dialplan and pass to an AGI script. I am now trying to set the
same thing up in 1.6 However because the argument delimter in 1.6 has
changed from pipe to comma this breaks as the AUTH line is also comma
delimited. Thus the AGI sees the AUTH as extra arguments instead of a
single argument. As the AUTH may contain varying number of arguments I
need a new way for a my AGI to access this data.

Does anyone have any ideas how I might go about this?

Regards

Jon


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Re: [asterisk-users] one way audio for xlite clients behind NAT

[Sebastian]

On 09/16/2010 06:58 PM, Thomas Johnson wrote:
> I am having a one way audio issue with xlite clients behind NAT. They
> can connect to the server and make calls but no audio is heard on the
> other end.
>
> my sip conf
>
> [general]
> context=default
> bindport=5060
> bindaddr=0.0.0.0
> srvlookup=yes
> canreinvite=no
>
> [tomfmason]
> type=friend
> secret=secret
> callerid="Thomas Johnson"  
> host=dynamic
> nat=yes
> canreinvite=no
> disallow=all
> allow=gsm
> allow=ulaw
> allow=alaw
> qualify=yes
> context=sip
>
> [1001];Work
> type=peer
> dtmfmode=rfc2833
> context=sip
> insecure=very
> host=sip.domain.com  
> nat=no
>
> [1000];IPKall
> type=peer
> dtmfmode=rfc2833
> context=sip
> insecure=very
> host=voiper.ipkall.com  
> nat=no

You seem to be using nat=no

shouldn't that be nat=yes?

>
>
>
> I pasted the log here ->  http://pastie.org/1163238
>
>
> I have tried connecting both of the clients to another sip 
> service(sip2sip.info  ) and did not have the same 
> problems.
>
>
> Any suggestions would be great.
>
> Thanks,
>
> Tom
>

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Re: [asterisk-users] Bug with Realtime?

[Leif Madsen]

On 10-09-16 09:43 AM, Dan Journo wrote:
>> That's not a bug. Only when the phone registers or performs some sort of 
>> action
>> (such as placing a call, etc...) does Asterisk query the database. If your
>> phones have a short re-registration time this becomes less of a problem.
>
> How do you explain that as soon as I issue a "reload" command, the realtime 
> phones stop receiving calls?
> To test your theory, I rebooted the phone so that it had a fresh 
> registration, I made and receives calls successfully, then issued a 'reload', 
> then trying to dial in again, and the phone didnt ring.
>
> After a few seconds, the CLI says:-
> [2010-09-16 14:39:29] NOTICE[24611]: chan_sip.c:17200 sip_poke_noanswer: Peer 
> 'kesher_201' is now UNREACHABLE!  Last qualify: 58
>
> How can this not be a bug? The phone works fine for hours, and then as soon 
> as I issue a reload command, its UNREACHABLE.
> ps. The phone can still make calls after the "reload". It just stops 
> receiving calls after a "reload".

That sounds like a qualify issue in that the phone does not respond to a NOTIFY 
message.

Check the SIP debug and see what is going on. Alternatively you could turn off 
the qualify option with qualify=no.

Leif.

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[asterisk-users] one way audio for xlite clients behind NAT

[Thomas Johnson]

I am having a one way audio issue with xlite clients behind NAT. They can
connect to the server and make calls but no audio is heard on the other
end.

my sip conf

[general]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
canreinvite=no[tomfmason]
type=friend
secret=secret
callerid="Thomas Johnson" 
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=gsm
allow=ulaw
allow=alaw
qualify=yes
context=sip[1001];Work
type=peer
dtmfmode=rfc2833
context=sip
insecure=very
host=sip.domain.com
nat=no[1000];IPKall
type=peer
dtmfmode=rfc2833
context=sip
insecure=very
host=voiper.ipkall.com
nat=no


I pasted the log here -> http://pastie.org/1163238


I have tried connecting both of the clients to another sip
service(sip2sip.info) and did not have the same problems.


Any suggestions would be great.

Thanks,

Tom
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Re: [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11

[Paul Belanger]

On Thu, Sep 16, 2010 at 12:46 PM, Paul Belanger
 wrote:
> On Thu, Sep 16, 2010 at 12:11 PM, Jonas Kellens
>  wrote:
>> Is it normal that backtrace.txt is only 30K ??
>>
> Normal or not, simply post the results of backtrace.txt
>
Please do not send me direct email, post them to the list for others
to help.  Your backtrace is optimized ().  You
need to reinstall asterisk with DONT_OPTIMIZE enabled, described in
doc/backtrace.txt.

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Re: [asterisk-users] Purpose of qualify=yes

[Steve Totaro]

On Thu, Sep 16, 2010 at 12:03 PM, Chris Owen  wrote:
> On Sep 16, 2010, at 10:44 AM, Steve Totaro wrote:
>
>> The other purpose is for DCHP and the IP address of a particular phone
>> may change.  If you hard code the phone and the corresponding entry in
>> sip.conf, you don't need to register or use qualify.
>>
>> If the phone is reachable then it will reply and the call will go
>> normally.  If it doesn't reply, then on with the dialplan.
>
> Now I'm not sure that makes sense to me.  If the IP address of the phone 
> changes and the phone doesn't reregister then yes calls can't get to it but 
> neither can the qualify packets.  I'm not sure how sending a qualify helps 
> here.
>
> Chris
>
> --
> -
> Chris Owen         - Garden City (620) 275-1900 -  Lottery (noun):
> President          - Wichita     (316) 858-3000 -    A stupidity tax
> Hubris Communications Inc      www.hubris.net
> -

I think if you re-read my post, it will make more sense.

Look for, "If you hard code the phone and the corresponding entry in
sip.conf, you don't need to register or use qualify."

If it is static, how is it going to change?

Thanks,
Steve T

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Re: [asterisk-users] Help!! Call waiting issue

[Paul Belanger]

On Thu, Sep 16, 2010 at 1:03 PM, carem gyssell nieto
 wrote:
>    It's an asterisk Bug? I have asterisk 1.4.22.
>
Please direct your attention to the following:
- http://www.catb.org/esr/faqs/smart-questions.html
- 
http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt

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Re: [asterisk-users] Bug with Realtime?

[Tilghman Lesher]

On Thursday 16 September 2010 11:23:37 Dan Journo wrote:
> Is there any development work being done on the realtime addon? Theres been
> no updates since April.

Realtime is integrated into the core; it is not an addon.  Perhaps you're
referring to the mysql realtime driver?  The driver modules tend not to need
updates that often; but only when the API changes.

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Re: [asterisk-users] Realtime semi-colon

[Tilghman Lesher]

On Thursday 16 September 2010 07:50:33 Steve Howes wrote:
> On 16 Sep 2010, at 12:56, Andrew Thomas wrote:
> > Does anyone know how to send * a semi-colon from a realtime database.  I
> > know that * uses the semi-colon as a 'seperator' - but I need to be able
> > to use one in a command.  I know I can use \; in the non-realtime
> > configs, but this doesn't work in realtime.
>
> in /etc/asterisk/extensions.conf
>
> [globals]
> SEMICOLON=\;
>
> Then use ${SEMICOLON} in realitime Hacky, but it's what I'm using at
> the moment..

If you use SVN (upcoming 1.4.37, 1.6.2.14, or 1.8.0), you can encode
semicolons in the database as the literal string "^3B".

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Re: [asterisk-users] [OT-FreePBX] Outbound calls check inbound routes to see if destination is local?

[Tim Nelson]

- "Paul Belanger"  wrote:
> On Thu, Sep 16, 2010 at 11:42 AM, Tim Nelson 
> wrote:
> > First, my apologies for the OT post. Yes, I understand this is not
> the FreePBX-users mailing list. But, there are a large number of
> people that use FreePBX and I'm hoping they can be of assistance.
> >
> If you know this is off-topic, and not the FreePBX-users mailing
> list,
> why did you even make the post? Regardless that your question is not

If you were to read the first line of my post which you so thoughtfully quoted, 
the answer to your question would be undeniably clear.

> actually specific to FreePBX, they do have an large and active
> community [1].

Yes, I am aware of that. However, my experience has been a large number of 
community members there are 'point and click' junkies that don't understand the 
technology or how the software works. In my observations, the asterisk-users 
list has more technical, experienced members.

> 
> Warning people your post is off-topic and FreePBX related is a sure
> way for people to ignore your post, or reply like I did.

Exactly. I put OT and FreePBX in the subject so anyone offended or annoyed 
could "ignore your post, or reply like I did". Sound familiar?

--Tim

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[asterisk-users] Help!! Call waiting issue

[carem gyssell nieto]

 I have an incomming call but when I receive a call by a 2nd line in my
   softphone, lost the first call. Sometimes the first call is dropped, and
   sometimes the call is active,  but I can't  hear the caller.
   It's an asterisk Bug? I have asterisk 1.4.22.
   Please help!!!
   Thanks

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Re: [asterisk-users] [OT-FreePBX] Outbound calls check inbound routes to see if destination is local?

[Paul Belanger]

On Thu, Sep 16, 2010 at 11:42 AM, Tim Nelson  wrote:
> First, my apologies for the OT post. Yes, I understand this is not the 
> FreePBX-users mailing list. But, there are a large number of people that use 
> FreePBX and I'm hoping they can be of assistance.
>
If you know this is off-topic, and not the FreePBX-users mailing list,
why did you even make the post? Regardless that your question is not
actually specific to FreePBX, they do have an large and active
community [1].

Warning people your post is off-topic and FreePBX related is a sure
way for people to ignore your post, or reply like I did.

[1] http://www.freepbx.org/community

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Re: [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11

[Paul Belanger]

On Thu, Sep 16, 2010 at 12:11 PM, Jonas Kellens
 wrote:
> Is it normal that backtrace.txt is only 30K ??
>
Normal or not, simply post the results of backtrace.txt

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Re: [asterisk-users] [OT-FreePBX] Outbound calls check inbound routes to see if destination is local?

[A J Stiles]

On Thursday 16 Sep 2010, Tim Nelson wrote:
> I have a system running Asterisk 1.4.27 (see... relevance!!!) and FreePBX
> 2.6.0. There are a large number of inbound routes configured for the
> various DID's coming in via PRI, SIP, etc. If a user calls outbound to one
> of these numbers, it goes out to the PSTN (using one channel of $0.0x/min),
> then comes back in on another channel (using another $0.0x/min).
>
> Obviously, the one call is costing 2x the per minute rate when it could be
> costing nothing. Is there a way to tell FreePBX to check the inbound routes
> for a match, and if found, route locally instead of using the default PSTN
> routes?

Create a context to use as the default for any extension which can do 
a "straight through" call  (i.e. not via the PSTN);  and which routes calls 
that can be routed straight through, straight through.

Include within this, a second context for extensions which need to go out via 
the PSTN.

Have a third context for incoming calls from the PSTN, with all "inside" 
extensions in it.  Optionally include the out-via-pstn context within this; 
although, logically, if someone arrived via the PSTN, they already have PSTN 
access and so should not need to go back out via the PSTN.

Note that when an extension could match one of several possible targets, 
Asterisk always prefers the most specific  (hardest to match)  over the 
vaguest.

So you should have something like this  (very very minimal example);

[in-via-pstn]
include=>internal-phones
exten=>00.,1,Play(int-barred)
.
.
.
[straight-through]
exten=>012345678[89][0-9],1,Dial(SIP/2${EXTEN}:9)
include=>internal-phones
include=>out-via-pstn
.
.
.
[internal-phones]
exten=>[2-6]XX,1,Dial(SIP/${EXTEN})
.
.
.
[out-via-pstn]
exten=>0[123]XX.,1,Dial(${TRUNK}/${EXTEN})
exten=>07XX.,1,Dial(${MOBILES}/${EXTEN})

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Re: [asterisk-users] Purpose of qualify=yes

[Gareth Blades]

One of the main benefits of qualify=yes is to detect network problems 
with peers.
We send a lot of calls via a service provider using SIP but we have 
qualify-yes set so that if it becomes unreachable the dial fails 
immediatly without having to wait for a timeout which enables us to 
seamlessly failover to an ISDN connection.

Chris Owen wrote:
> We have a tenant who has been having issues with a congested connection and 
> in trouble shooting it we've noticed that there seems to be a lot of SIP 
> traffic even when none of the phones are doing anything.
> 
> We've determined that this traffic is mostly INFO packets generated by 
> setting qualify=2000.   I understand that 2000 ms is the default value for 
> the qualification parameter but what I'm unclear on is exactly what the 
> purpose of having asterisk qualify the phones is.
> 
> I know that in a NAT situation, qualifications can help keep UDP sessions 
> open in the firewall but in our case most phones are not behind NAT.
> 
> I realize qualifying phones is also how asterisk keeps track of who is 
> available for things like BLF but surely it doesn't need to do that every 2 
> seconds to keep the BLFs reasonably current.
> 
> So I guess my question is what is the real purpose of the qualify setting in 
> a non-NAT situation and can one safely set the qualification as something 
> higher.   I'd think something like 15 seconds would be more than enough for 
> BLFs and the like.
> 
> Chris
> 
> 
> 


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Re: [asterisk-users] Bug with Realtime?

[Danny Nicholas]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Thursday, September 16, 2010 11:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bug with Realtime?

>  Have you checked the Issue Tracker

Not yet. I wanted to see if it's just me before searching through/raising a
bug report.

Always a good idea to check it anyway;  somebody else may have put the issue
there without writing on the list.


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Re: [asterisk-users] Bug with Realtime?

[Dan Journo]

Is there any development work being done on the realtime addon? Theres been no 
updates since April.

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Re: [asterisk-users] Purpose of qualify=yes

[Peder]

qualify=2000 does not mean it sends a qualify every 2000ms, 2 seconds.  It
means that the qualify timeout is 2000ms, so if it receives a response at
2600ms, it counts that phone as down.  I believe the timing of qualifies is
still every 60 seconds, unless explicitly changed by the system admin:

http://www.voip-info.org/wiki/view/Asterisk+sip+qualify

So 20 phones with qualify is 40 packets/minute (1 packet out and 1 packet in
per phone).

I've always liked qualify as it lets me know if a phone is alive or not,
even in non NAT scenarios.  If someone calls in and says "my phone doesn't
work", I can check the qualify and if it shows it down, have them reboot.
If it shows up, then I debug them trying to place a call.  It is just easy
extra help in troubleshooting.



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Owen
Sent: Thursday, September 16, 2010 11:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Purpose of qualify=yes

On Sep 16, 2010, at 10:45 AM, Zeeshan Zakaria wrote:

> I prefer to keep qualify=on for all the extensions, as it gives you an
idea which extensions are going to give you trouble. For extensions with
qualify value greater than 300 ms you should definitely worry. For
extensions at 2000ms delay or more, turning qualify off simply means to
ignore the obvious problem. Such extensions have communication or network
issues which require serious attention. You can set this parameter to, e.g.
3000 ms or more if dealing with 2000 ms delay is unavoidable, but don't turn
it off. Afterall even at 2000 ms conversation is not truly real time and not
easy.

In our case the problem isn't that the phones are experiencing high latency
per se but rather than a full pipe plus all these SIP messages is playing
hell with the QOS stuff.

20 phones in one location times say 4 SIP packets every 2 seconds equals 40
SIP packets a second.   That normally isn't a problem but when the pipe gets
congested then we start seeing issues when a call comes in and 400 BLF
notices go out etc.  Obviously we can increase the amount of bandwidth
reserved for SIP traffic but I'm just not sure why we're sending all those
packets in the first place.

In other words, the qualify traffic is actually causing the problem, not
revealing it.

Chris



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Re: [asterisk-users] Bug with Realtime?

[Dan Journo]

>  Have you checked the Issue Tracker

Not yet. I wanted to see if it's just me before searching through/raising a bug 
report.

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Re: [asterisk-users] Bug with Realtime?

[Peder]

> But it doesnt explain why the phones that are hard coded in the sip.conf
file don't lose registration.

On a reload, it re-reads the sip.conf config file and sees the users in
there, so it doesn't flush them.  It doesn't pull down the whole SIP table
on a reload, it only loads a realtime peer config when the phone tries to
register, so realtime users disappear on a reload, but sip.conf users do
not.

PA



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Re: [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11

[Jonas Kellens]

On 09/16/2010 05:45 PM, Paul Belanger wrote:
> On Thu, Sep 16, 2010 at 8:19 AM, Jonas Kellens  
> wrote:
>
>> I get so little output :
>>
>>  
> You are still doing it incorrectly. As said, doc/backtrace.txt has all
> the required information.
>

bash-3.2# gdb -se "/usr/sbin/asterisk" -ex "bt full" -ex "thread apply 
all bt" --batch -c /tmp/core.4569 > /tmp/backtrace.txt
bash-3.2# ls -lh /tmp/
total 408M
-rw-rw-r-- 1 root root  30K Sep 16 18:09 backtrace.txt


Is it normal that backtrace.txt is only 30K ??


When I do the different steps, I have a lot of output. So I think I'm 
doing it correct now.

Jonas.

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Re: [asterisk-users] Purpose of qualify=yes

[jon pounder]

On 09/16/2010 12:01 PM, Chris Owen wrote:

well that just means you need a trunked satellite pbx where all the 
phones are, and that would take load off the main connection.

half those people have got to just be talking to each other and don't 
need to use the gateway at all.

> On Sep 16, 2010, at 10:45 AM, Zeeshan Zakaria wrote:
>
>
>> I prefer to keep qualify=on for all the extensions, as it gives you an idea 
>> which extensions are going to give you trouble. For extensions with qualify 
>> value greater than 300 ms you should definitely worry. For extensions at 
>> 2000ms delay or more, turning qualify off simply means to ignore the obvious 
>> problem. Such extensions have communication or network issues which require 
>> serious attention. You can set this parameter to, e.g. 3000 ms or more if 
>> dealing with 2000 ms delay is unavoidable, but don't turn it off. Afterall 
>> even at 2000 ms conversation is not truly real time and not easy.
>>  
> In our case the problem isn't that the phones are experiencing high latency 
> per se but rather than a full pipe plus all these SIP messages is playing 
> hell with the QOS stuff.
>
> 20 phones in one location times say 4 SIP packets every 2 seconds equals 40 
> SIP packets a second.   That normally isn't a problem but when the pipe gets 
> congested then we start seeing issues when a call comes in and 400 BLF 
> notices go out etc.  Obviously we can increase the amount of bandwidth 
> reserved for SIP traffic but I'm just not sure why we're sending all those 
> packets in the first place.
>
> In other words, the qualify traffic is actually causing the problem, not 
> revealing it.
>
> Chris
>
>
>
>


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Re: [asterisk-users] Purpose of qualify=yes

[Chris Owen]

On Sep 16, 2010, at 10:45 AM, Zeeshan Zakaria wrote:

> I prefer to keep qualify=on for all the extensions, as it gives you an idea 
> which extensions are going to give you trouble. For extensions with qualify 
> value greater than 300 ms you should definitely worry. For extensions at 
> 2000ms delay or more, turning qualify off simply means to ignore the obvious 
> problem. Such extensions have communication or network issues which require 
> serious attention. You can set this parameter to, e.g. 3000 ms or more if 
> dealing with 2000 ms delay is unavoidable, but don't turn it off. Afterall 
> even at 2000 ms conversation is not truly real time and not easy.

In our case the problem isn't that the phones are experiencing high latency per 
se but rather than a full pipe plus all these SIP messages is playing hell with 
the QOS stuff.

20 phones in one location times say 4 SIP packets every 2 seconds equals 40 SIP 
packets a second.   That normally isn't a problem but when the pipe gets 
congested then we start seeing issues when a call comes in and 400 BLF notices 
go out etc.  Obviously we can increase the amount of bandwidth reserved for SIP 
traffic but I'm just not sure why we're sending all those packets in the first 
place.

In other words, the qualify traffic is actually causing the problem, not 
revealing it.

Chris



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Re: [asterisk-users] Purpose of qualify=yes

[Chris Owen]

On Sep 16, 2010, at 10:44 AM, Steve Totaro wrote:

> The other purpose is for DCHP and the IP address of a particular phone
> may change.  If you hard code the phone and the corresponding entry in
> sip.conf, you don't need to register or use qualify.
> 
> If the phone is reachable then it will reply and the call will go
> normally.  If it doesn't reply, then on with the dialplan.

Now I'm not sure that makes sense to me.  If the IP address of the phone 
changes and the phone doesn't reregister then yes calls can't get to it but 
neither can the qualify packets.  I'm not sure how sending a qualify helps here.

Chris

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Hubris Communications Inc  www.hubris.net
-



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Re: [asterisk-users] [OT-FreePBX] Outbound calls check inbound routes to see if destination is local?

[Danny Nicholas]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Thursday, September 16, 2010 10:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] [OT-FreePBX] Outbound calls check inbound routes
to see if destination is local?

Greetings-

First, my apologies for the OT post. Yes, I understand this is not the
FreePBX-users mailing list. But, there are a large number of people that use
FreePBX and I'm hoping they can be of assistance.

I have a system running Asterisk 1.4.27 (see... relevance!!!) and FreePBX
2.6.0. There are a large number of inbound routes configured for the various
DID's coming in via PRI, SIP, etc. If a user calls outbound to one of these
numbers, it goes out to the PSTN (using one channel of $0.0x/min), then
comes back in on another channel (using another $0.0x/min).

Obviously, the one call is costing 2x the per minute rate when it could be
costing nothing. Is there a way to tell FreePBX to check the inbound routes
for a match, and if found, route locally instead of using the default PSTN
routes?

I appreciate any comments, suggestions.

--Tim

Make a calling rule to identify lines that can be reached locally.  My
assumption is that the reachable lines/extensions would be SIP, but hey,
that wouldn't be the first time I was wrong today.


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Re: [asterisk-users] Bug with Realtime?

[Danny Nicholas]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Thursday, September 16, 2010 10:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bug with Realtime?

> Noted - but if OP does a "reload" once a day, 120 seconds (2 minutes) out
of
> 1 day (14400 seconds) is 99.17% uptime; "close enough" to 99.999 percent
in
> most folks books.  What percentage of businesses use their phones 24/7?

Even if its once a month, it's still too much in my book. No wonder people
have had "bad experiences" with voip.
If I'm going to run my service on a platform that will inevitably have some
downtime, I wouldn't be a very good business man!

Let's say I add a new provider to my service and therefore have to add
another "register=>" command into sip.conf, I'd have to issue a "sip reload"
which would kill off all the realtime sip phones.

> A reload flushes the SIP registration database, so once you do a 
> reload, that phones reg is gone.

Surely you could move the "flush" part of "sip reload" into a separate
command and therefore ensure the realtime phones stay connected properly.
Still doesnt explain why, when I do a "reload", only the realtime users are
affected, and not the hard coded ones.

Dan

Just a WAG, but I'm betting that the hard-coded users are loaded into a hash
that Asterisk handles and the real-time users are either in a different hash
or not at all.  Since real-time is an "add-on" vs a "built-in" feature, this
might indeed be a "bug" or shortcoming.  Have you checked the Issue Tracker
for this?


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[asterisk-users] [OT-FreePBX] Outbound calls check inbound routes to see if destination is local?

[Tim Nelson]

Greetings-

First, my apologies for the OT post. Yes, I understand this is not the 
FreePBX-users mailing list. But, there are a large number of people that use 
FreePBX and I'm hoping they can be of assistance.

I have a system running Asterisk 1.4.27 (see... relevance!!!) and FreePBX 
2.6.0. There are a large number of inbound routes configured for the various 
DID's coming in via PRI, SIP, etc. If a user calls outbound to one of these 
numbers, it goes out to the PSTN (using one channel of $0.0x/min), then comes 
back in on another channel (using another $0.0x/min).

Obviously, the one call is costing 2x the per minute rate when it could be 
costing nothing. Is there a way to tell FreePBX to check the inbound routes for 
a match, and if found, route locally instead of using the default PSTN routes?

I appreciate any comments, suggestions.

--Tim

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Re: [asterisk-users] Purpose of qualify=yes

[Zeeshan Zakaria]

I prefer to keep qualify=on for all the extensions, as it gives you an idea
which extensions are going to give you trouble. For extensions with qualify
value greater than 300 ms you should definitely worry. For extensions at
2000ms delay or more, turning qualify off simply means to ignore the obvious
problem. Such extensions have communication or network issues which require
serious attention. You can set this parameter to, e.g. 3000 ms or more if
dealing with 2000 ms delay is unavoidable, but don't turn it off. Afterall
even at 2000 ms conversation is not truly real time and not easy.

Zeeshan A Zakaria

--
www.ilovetovoip.com

On 2010-09-16 11:38 AM, "Benny Amorsen"
>
wrote:

Chris Owen  writes:

> So I guess my question is what is the real purpose of the q...
The purpose is simply to see if the phone is available. For your
particular use it is likely best to simply turn it off completely. If a
phone disappears, its registration will eventually time out anyway.


/Benny



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Re: [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11

[Paul Belanger]

On Thu, Sep 16, 2010 at 8:19 AM, Jonas Kellens  wrote:
> I get so little output :
>
You are still doing it incorrectly. As said, doc/backtrace.txt has all
the required information.

-- 
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com

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Re: [asterisk-users] Purpose of qualify=yes

[Steve Totaro]

On Thu, Sep 16, 2010 at 11:32 AM, Benny Amorsen  wrote:
> Chris Owen  writes:
>
>> So I guess my question is what is the real purpose of the qualify
>> setting in a non-NAT situation and can one safely set the
>> qualification as something higher. I'd think something like 15 seconds
>> would be more than enough for BLFs and the like.
>
> The purpose is simply to see if the phone is available. For your
> particular use it is likely best to simply turn it off completely. If a
> phone disappears, its registration will eventually time out anyway.
>
>
> /Benny
>
>

The other purpose is for DCHP and the IP address of a particular phone
may change.  If you hard code the phone and the corresponding entry in
sip.conf, you don't need to register or use qualify.

If the phone is reachable then it will reply and the call will go
normally.  If it doesn't reply, then on with the dialplan.

Thanks,
Steve T

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Re: [asterisk-users] Bug with Realtime?

[Dan Journo]

> Noted - but if OP does a "reload" once a day, 120 seconds (2 minutes) out of
> 1 day (14400 seconds) is 99.17% uptime; "close enough" to 99.999 percent in
> most folks books.  What percentage of businesses use their phones 24/7?

Even if its once a month, it's still too much in my book. No wonder people have 
had "bad experiences" with voip.
If I'm going to run my service on a platform that will inevitably have some 
downtime, I wouldn't be a very good business man!

Let's say I add a new provider to my service and therefore have to add another 
"register=>" command into sip.conf, I'd have to issue a "sip reload" which 
would kill off all the realtime sip phones.

> A reload flushes the SIP registration database, so once you do a 
> reload, that phones reg is gone.

Surely you could move the "flush" part of "sip reload" into a separate command 
and therefore ensure the realtime phones stay connected properly.
Still doesnt explain why, when I do a "reload", only the realtime users are 
affected, and not the hard coded ones.

Dan

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Re: [asterisk-users] Bug with Realtime?

[Zeeshan Zakaria]

When making an outbound call, if sip peer is not registered, first it
registers itself, and then makes the call. This is why you don't see any
problem dialing out. For receiving, asterisk has to wait until the sip peer
registers, otherwise asterisk has nowhere to send the call.

I know the pain, as I deal with the same situation. So I don't do 'reload'
or 'sip reload' except if sip password (secret) has been changed, in which
case I prefer to use 'sip prune realtime peer ' followed by 'sip
show peer  load'. Most of the sip devices re-register every 60
seconds, or if they don't on a realtime network, depending upon the
bandwidth, they should be made to do so. Or in some cases you can send a
reboot signal to a sip device too. The bottom line is, try not to do a
'reload' as it would affect everybody else too by dropping their
registrations temporarily.

Zeeshan A Zakaria

--
www.ilovetovoip.com

On 2010-09-16 10:04 AM, "Peder"  wrote:

A reload flushes the SIP registration database, so once you do a reload,
that phones reg is gone.  If the reg is set for a short period, say 60
seconds, then in 60 seconds it will re-register and work fine.  Yes, it is a
total pain, but this is the way it has worked since day 1 for realtime.  I
agree that it seems wrong and even argued that several years ago when this
feature came out, but it is what it is.  As someone else said, the answer is
"don't do a 'reload'", do an "extensions reload" or whatever it is specific
to your changes.



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-bo...

Sent: Thursday, September 16, 2010 8:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussi...

Subject: Re: [asterisk-users] Bug with Realtime?

> That's not a bug. Only when the phone registers ...
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Re: [asterisk-users] Purpose of qualify=yes

[Benny Amorsen]

Chris Owen  writes:

> So I guess my question is what is the real purpose of the qualify
> setting in a non-NAT situation and can one safely set the
> qualification as something higher. I'd think something like 15 seconds
> would be more than enough for BLFs and the like.

The purpose is simply to see if the phone is available. For your
particular use it is likely best to simply turn it off completely. If a
phone disappears, its registration will eventually time out anyway.


/Benny


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Re: [asterisk-users] Echo on Sangoma A400 and background noise

[Moises Silva]

On Wed, Sep 15, 2010 at 6:10 PM, Al lists  wrote:

> I'm a long time user of Digium carts and stupid me i wanted to give Sangoma
> a try.
> We got Sangoma A400 with 6 FXO ports.
>
> Asterisk version: 1.4.35
> Zaptel version: 1.4.11
> Wanpipe version: 3.5.11
>
> we tried to use fxtune but looks like it wont work with Sangoma card, (
> please correct me if i'm wrong)
> Echo is really bad and also we have  background noise on all lines.
> We tried both mg2 and oslec echo canceler.
> was wondering if you have any experiense with that because Sangoma tech
> support is not helpfull, just look at their response:
>

Hello,

I am sorry you've had this bad experience and I apologize on behalf of
Sangoma if we gave you the impression of not caring about your technical
issue.

To rephrase what tech support meant. Our HWEC is known to provide better
results, but in no way that means that we will not look at your issue with
the card that does not have HWEC.

A senior tech support engineer will be contacting you soon today to follow
up on your issue appropriately.

Regards,

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | NEW 100 Renfrew Drive, Suite 100, Markham ON L3R
9R6 Canada
t. 1 905 474 1990 x128 | e. m...@sangoma.com
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Re: [asterisk-users] Bug with Realtime?

[Danny Nicholas]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack
Sent: Thursday, September 16, 2010 9:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bug with Realtime?

Danny Nicholas wrote:

If your clients can't take 2 minutes of "downtime" on a phone, they 
don't need to be on VOIP.


If VOIP ( and Asterisk ) ever really expect to be "the future of 
Telephony "  this ( attitude ) has to change

90 percent availability is unacceptable, even 95 percent,  for that 
matter even 99.99 percent availability.
Nothing short of 99.999 percent available will win.

Just one old phone man's opinion. ( along with a large number of users )

John Novack

Noted - but if OP does a "reload" once a day, 120 seconds (2 minutes) out of
1 day (14400 seconds) is 99.17% uptime; "close enough" to 99.999 percent in
most folks books.  What percentage of businesses use their phones 24/7?


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Re: [asterisk-users] Bug with Realtime?

[Danny Nicholas]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Thursday, September 16, 2010 9:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bug with Realtime?

> As someone else said, the answer is
"don't do a 'reload'", do an "extensions reload" or whatever it is specific
to your changes.

You are correct. I'm just being lazy. But I'm just worried that some time in
the future, I'll have to reload the sip config, and therefore flush out all
the realtime phones.

Can asterisk cope with 1000 phones all re-registering every 60 seconds?

Thanks
Dan

"Asterisk" absolutely can; The more valid questions are these:
Is my CPU/Bandwidth "stout" enough for 1000 extra "registrations" per
minute?
Should I make a "daemon" instead to check and register "UNKNOWN" peers?




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[asterisk-users] Indications and tonelist on a SIP channel..

[Carlos C.]

Hello All!

I want to add a silence to the beginning of a ring tonelist for a country 
inside the indications.conf file. I want that silence to be played just once, 
reason why am using an exclamation mark in front of the tone but is not 
working. Am getting the ring tone right away. I tried these combinations below 
and some others but no help:

ring = !420*40/100,!0/5000,420*40/2000,0/4000
ring = !0/5000,420*40/2000,0/4000

Is like if the exclamation mark is not been recognized. Am not sure if it has 
anything to do with the fact that this is been used on a SIP channel. I am 
using the "r" parameter with in the dial command to generate this tone 
obviously. 

Regards,

---
Carlos C.




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Re: [asterisk-users] Bug with Realtime?

[John Novack]

Danny Nicholas wrote:

If your clients can't take 2 minutes of "downtime" on a phone, they 
don't need to be on VOIP.




If VOIP ( and Asterisk ) ever really expect to be "the future of 
Telephony "  this ( attitude ) has to change

90 percent availability is unacceptable, even 95 percent,  for that 
matter even 99.99 percent availability.
Nothing short of 99.999 percent available will win.

Just one old phone man's opinion. ( along with a large number of users )

John Novack

-- 

Dog is my Co-pilot


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Re: [asterisk-users] Bug with Realtime?

[Dan Journo]

> As someone else said, the answer is
"don't do a 'reload'", do an "extensions reload" or whatever it is specific
to your changes.

You are correct. I'm just being lazy. But I'm just worried that some time in 
the future, I'll have to reload the sip config, and therefore flush out all the 
realtime phones.

Can asterisk cope with 1000 phones all re-registering every 60 seconds?

Thanks
Dan

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Re: [asterisk-users] Bug with Realtime?

[Dan Journo]

> A reload flushes the SIP registration database, so once you do a reload,
that phones reg is gone. 

Finally an answer that seemed more realistic. But it doesnt explain why the 
phones that are hard coded in the sip.conf file don't lose registration.

Any ideas?

Thanks
Dan

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Re: [asterisk-users] changing from zap to DAHDI

[Jerry Geis]

Jerry Geis wrote:
>>
>> Somewhere on your system you have a modprobe install command that's
>> running when the module is loaded.  Most likely it was installed on your
>> system by
>> http://svn.asterisk.org/view/zaptel/branches/1.4/build_tools/genmodconf?view=markup
>>  
>>
>> when you installed zaptel.
>>
>> Do you have an /etc/conf.modules file?  What does 'grep -r ztconfig
>> /etc/.' return?
>>
>>   
>
> Shaun,
>
> below is the results of the command.
>
> grep -r ztconfig /etc/.
> grep: /etc/./httpd/run/asterisk.ctl: No such device or address
> grep: /etc/./httpd/run/dbus/system_bus_socket: No such device or address
> grep: /etc/./httpd/run/acpid.socket: No such device or address
>
> Jerry
>
>
This is the new output (ztcfg is no longer mentioned) so I think that 
issue is fixed.
Now its:
+ initlog -q -c 'unload_module dahdi'
execvp: No such file or directory

---

sh -x /etc/init.d/dahdi stop
+ initdir=/etc/init.d
+ DAHDI_CFG=/usr/sbin/dahdi_cfg
+ DAHDI_CFG_CMD=/usr/sbin/dahdi_cfg
+ FXOTUNE=/usr/sbin/fxotune
+ XPP_SYNC=auto
+ DAHDI_DEV_TIMEOUT=20
+ system=redhat
+ '[' -f /etc/debian_version ']'
+ '[' redhat = redhat ']'
+ . /etc/init.d/functions
++ TEXTDOMAIN=initscripts
++ umask 022
++ PATH=/sbin:/usr/sbin:/bin:/usr/bin:/usr/X11R6/bin
++ export PATH
++ '[' -z '' ']'
++ COLUMNS=80
++ '[' -z '' ']'
+++ /sbin/consoletype
++ CONSOLETYPE=pty
++ '[' -f /etc/sysconfig/i18n -a -z '' ']'
++ . /etc/sysconfig/i18n
+++ LANG=en_US.UTF-8
+++ SUPPORTED=en_US.UTF-8:en_US:en
+++ SYSFONT=latarcyrheb-sun16
++ '[' pty '!=' pty ']'
++ '[' -n '' ']'
++ export LANG
++ '[' -z '' ']'
++ '[' -f /etc/sysconfig/init ']'
++ . /etc/sysconfig/init
+++ BOOTUP=color
+++ GRAPHICAL=yes
+++ RES_COL=60
+++ MOVE_TO_COL='echo -en \033[60G'
+++ SETCOLOR_SUCCESS='echo -en \033[0;32m'
+++ SETCOLOR_FAILURE='echo -en \033[0;31m'
+++ SETCOLOR_WARNING='echo -en \033[0;33m'
+++ SETCOLOR_NORMAL='echo -en \033[0;39m'
+++ LOGLEVEL=3
+++ PROMPT=yes
++ '[' pty = serial ']'
++ '[' color '!=' verbose ']'
++ INITLOG_ARGS=-q
+ DAHDI_MODULES_FILE=/etc/dahdi/modules
+ '[' -r /etc/dahdi/init.conf ']'
+ . /etc/dahdi/init.conf
+ '[' redhat = redhat ']'
+ LOCKFILE=/var/lock/subsys/dahdi
+ '[' '!' -x /usr/sbin/dahdi_cfg ']'
+ '[' '!' -f /etc/dahdi/system.conf ']'
+ RETVAL=0
+ case "$1" in
+ '[' redhat = debian ']'
+ '[' redhat = redhat ']'
+ action 'Unloading DAHDI hardware modules: ' unload_module dahdi
+ STRING='Unloading DAHDI hardware modules: '
+ echo -n 'Unloading DAHDI hardware modules:  '
Unloading DAHDI hardware modules:  + '[' '' '!=' '' -a -w 
/etc/rhgb/temp/rhgb-console ']'
+ shift
+ initlog -q -c 'unload_module dahdi'
execvp: No such file or directory
+ failure 'Unloading DAHDI hardware modules: '
+ rc=255
+ '[' -z '' ']'
+ initlog -q -n /etc/init.d/dahdi -s 'Unloading DAHDI hardware modules: 
' -e 2
+ '[' color '!=' verbose -a -z '' ']'
+ echo_failure
+ '[' color = color ']'
+ echo -en '\033[60G'
   + echo -n '['
[+ '[' color = color ']'
+ echo -en '\033[0;31m'
+ echo -n FAILED
FAILED+ '[' color = color ']'
+ echo -en '\033[0;39m'
+ echo -n ']'
]+ echo -ne '\r'
+ return 1
+ '[' -x /usr/bin/rhgb-client ']'
+ /usr/bin/rhgb-client --details=yes
+ return 255
+ rc=255
+ echo

+ '[' '' '!=' '' -a -w /etc/rhgb/temp/rhgb-console ']'
+ return 255
+ '[' /var/lock/subsys/dahdi '!=' '' ']'
+ '[' 0 -eq 0 ']'
+ rm -f /var/lock/subsys/dahdi
+ exit 0


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Re: [asterisk-users] Bug with Realtime?

[Danny Nicholas]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Thursday, September 16, 2010 8:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bug with Realtime?

>> That's not a bug. Only when the phone registers or performs some sort of
action 
>> (such as placing a call, etc...) does Asterisk query the database. If
your 
>> phones have a short re-registration time this becomes less of a problem.

>>How do you explain that as soon as I issue a "reload" command, the
realtime phones stop receiving calls?
>To test your theory, I rebooted the phone so that it had a fresh
>registration, I made and receives calls successfully, then issued a
'reload', then trying to dial in again, and the phone didnt ring.

>After a few seconds, the CLI says:-
>[2010-09-16 14:39:29] NOTICE[24611]: chan_sip.c:17200 sip_poke_noanswer:
Peer 'kesher_201' is now UNREACHABLE!  Last qualify: 58

>How can this not be a bug? The phone works fine for hours, and then as soon
as I issue a reload command, its UNREACHABLE.
ps. The phone can still make calls after the "reload". It just stops
receiving calls after a "reload".

>I want to move my clients over to realtime so they can manage their
accounts online, but I cant do that if they become UNREACHABLE when I do a
reload.

>Thanks
>Dan

I'm still going to defend that this is not a "bug".  It is up to you to
insure that your phones are in constant connectivity without burdening your
bandwith.  You could be aggressive and register your phone every 30 seconds;
a more realistic approach would be a register every 90-120 seconds (I have
Polycom 501's and they take at least 90 seconds to come back up on a
restart).  If your clients can't take 2 minutes of "downtime" on a phone,
they don't need to be on VOIP.


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Re: [asterisk-users] Bug with Realtime?

[Peder]

A reload flushes the SIP registration database, so once you do a reload,
that phones reg is gone.  If the reg is set for a short period, say 60
seconds, then in 60 seconds it will re-register and work fine.  Yes, it is a
total pain, but this is the way it has worked since day 1 for realtime.  I
agree that it seems wrong and even argued that several years ago when this
feature came out, but it is what it is.  As someone else said, the answer is
"don't do a 'reload'", do an "extensions reload" or whatever it is specific
to your changes.


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Thursday, September 16, 2010 8:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bug with Realtime?

> That's not a bug. Only when the phone registers or performs some sort of
action 
> (such as placing a call, etc...) does Asterisk query the database. If your

> phones have a short re-registration time this becomes less of a problem.

How do you explain that as soon as I issue a "reload" command, the realtime
phones stop receiving calls?
To test your theory, I rebooted the phone so that it had a fresh
registration, I made and receives calls successfully, then issued a
'reload', then trying to dial in again, and the phone didnt ring.

After a few seconds, the CLI says:-
[2010-09-16 14:39:29] NOTICE[24611]: chan_sip.c:17200 sip_poke_noanswer:
Peer 'kesher_201' is now UNREACHABLE!  Last qualify: 58

How can this not be a bug? The phone works fine for hours, and then as soon
as I issue a reload command, its UNREACHABLE.
ps. The phone can still make calls after the "reload". It just stops
receiving calls after a "reload".

I want to move my clients over to realtime so they can manage their accounts
online, but I cant do that if they become UNREACHABLE when I do a reload.

Thanks
Dan

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Re: [asterisk-users] changing from zap to DAHDI

[Jerry Geis]

>
> Somewhere on your system you have a modprobe install command that's
> running when the module is loaded.  Most likely it was installed on your
> system by
> http://svn.asterisk.org/view/zaptel/branches/1.4/build_tools/genmodconf?view=markup
> when you installed zaptel.
>
> Do you have an /etc/conf.modules file?  What does 'grep -r ztconfig
> /etc/.' return?
>
>   

Shaun,

below is the results of the command.

grep -r ztconfig /etc/.
grep: /etc/./httpd/run/asterisk.ctl: No such device or address
grep: /etc/./httpd/run/dbus/system_bus_socket: No such device or address
grep: /etc/./httpd/run/acpid.socket: No such device or address

Jerry


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Re: [asterisk-users] Dual WAN with load balancing

[asterisk asterisk]

Apart from that, any other tricks that I can manipulate within asterisk.
??sip.conf parameter or other??

On Thu, Sep 16, 2010 at 12:07 AM, Luki  wrote:

> > I am not sure about the problem but note that it may be related to
> incorrect
> > IP being used. Sometimes, WAN 1 and sometimes WAN 2
>
> Most likely. Get a provider that uses IP authentication rather than
> registrations, and enable access from both of your WAN IPs. All set.
>
> Luki
>
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Re: [asterisk-users] Bug with Realtime?

[Dan Journo]

> That's not a bug. Only when the phone registers or performs some sort of 
> action 
> (such as placing a call, etc...) does Asterisk query the database. If your 
> phones have a short re-registration time this becomes less of a problem.

How do you explain that as soon as I issue a "reload" command, the realtime 
phones stop receiving calls?
To test your theory, I rebooted the phone so that it had a fresh registration, 
I made and receives calls successfully, then issued a 'reload', then trying to 
dial in again, and the phone didnt ring.

After a few seconds, the CLI says:-
[2010-09-16 14:39:29] NOTICE[24611]: chan_sip.c:17200 sip_poke_noanswer: Peer 
'kesher_201' is now UNREACHABLE!  Last qualify: 58

How can this not be a bug? The phone works fine for hours, and then as soon as 
I issue a reload command, its UNREACHABLE.
ps. The phone can still make calls after the "reload". It just stops receiving 
calls after a "reload".

I want to move my clients over to realtime so they can manage their accounts 
online, but I cant do that if they become UNREACHABLE when I do a reload.

Thanks
Dan

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[asterisk-users] DTMF tones too long, for once

[Justin Sherrill]

I encountered something strange.  A local business has an ACD that, when I call 
it using a Polycom 550 connected through an Asterisk system, will respond to 
button presses only if they are short.

Calling this business with our old (non-Asterisk) phone system or with my cell 
phone works because they only generates short tones when pressing the keypad 
during a call.  The Polycom, however, generates tone for as long as you have 
your finger on the key.

I see plenty of posts out there on increasing tone length, but nothing on 
reducing it.  There may be other issues; I've called the number a few times 
when testing and received a voicemail prompt instead of their main message, a 
few times.

Justin C. Sherrill - American Rock Salt
p: 585-991-6825 f: 585-991-6926 c: 585-298-6826

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Re: [asterisk-users] Realtime semi-colon

[Steve Howes]

On 16 Sep 2010, at 12:56, Andrew Thomas wrote:
> Does anyone know how to send * a semi-colon from a realtime database.  I
> know that * uses the semi-colon as a 'seperator' - but I need to be able
> to use one in a command.  I know I can use \; in the non-realtime
> configs, but this doesn't work in realtime.

in /etc/asterisk/extensions.conf

[globals]
SEMICOLON=\;

Then use ${SEMICOLON} in realitime Hacky, but it's what I'm using at the 
moment..

S
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Re: [asterisk-users] Configure Asterisk with openssl

[Nikhil]

  Its working now when I installed openssl using yum.

yum install -y openssl-devel.

Thanks
Nikhil


On 09/16/2010 05:26 PM, Nikhil wrote:
>On 09/16/2010 04:11 PM, A J Stiles wrote:
>> On Thursday 16 Sep 2010, Nikhil wrote:
>>> Hi
>>>  I got the bellow error when I try to configure asterisk code.
>>>
>
>>> $./configure --with-ssl=/usr/local/ssl
>>> ...
>>> ...
>>> ...
>>> checking for mandatory modules:  OPENSSL... fail
>>>
>>> configure: ***
>>> configure: *** The OPENSSL installation appears to be missing or broken.
>>> configure: *** Either correct the installation, or run configure
>>> configure: *** including --without-ssl.
>>>
>>> I installed openssl in "/usr/local/ssl/" .
>>>
>>> Please help me to resolve this as I need to configure asterisk with SIP
>>> in TLS ,I am using 1.8 beta5 version . Or give  some link where I am
>>> find how to make asterisk to work with SIP on TLS.
>> Did you remember to run ldconfig after installing ssl?
>>
>> What OS are you running?  If some kind of Linux, why not just install the
>> openssl and openssl-devel packages?
>>
> I am using centOS-5 and I installed the ssl from source
> code(openssl-0.9.8k),but I think that is not the problem..Did anyone
> face the same type of error when doing configure the asterisk code?if
> yes how u solved that problem.?please help me on this.
>
> Thanks
> Nikhil
>
>


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Re: [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11

[Jonas Kellens]

On 09/16/2010 12:41 PM, Philipp von Klitzing wrote:
> Hi!
>
>> Does this shine new light to the problem ?!
>>  
> No. Once more: Go and read doc/backtrace.txt.
>
> And check if you have any meaningful information in /var/log/messages for
> the timestamp when asterisk crashed.
>
> Philipp
>

I get so little output :

Core was generated by `/usr/sbin/asterisk -f -vvvg -c'.
Program terminated with signal 6, Aborted.
#0  0x2b4f70970265 in ?? ()
(gdb) set logging on
Copying output to gdb.txt.
(gdb) bt
#0  0x2b4f70970265 in ?? ()
(gdb) bt full
#0  0x2b4f70970265 in ?? ()
No symbol table info available.
(gdb)


The generated files are also very small :

bash-3.2# ls -lh
total 408M
-rw-rw-r-- 1 root root 4.0K Sep 16 14:17 backtrace.txt
-rw--- 1 root root  42M Sep 15 21:46 core.4569
-rw-rw-r-- 1 root root 2.6K Sep 16 14:17 gdb.txt


I followed the steps in :
---
--- Getting Information After A Crash
---



Jonas.

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[asterisk-users] How to Understand a pri intense debug span X

[Danny Dias]

Hello my friends,

I would like to understand the output from "pri intense debug span X", the
Telco always says that their side is OK, but i always receive alarms,
loosing connection, take a look:

[Sep 16 13:18:19] WARNING[30364] chan_zap.c: Detected alarm on channel 1:
Recovering
[Sep 16 13:18:19] WARNING[30364] chan_zap.c: Unable to disable echo
cancellation on channel 1
[...]
[Sep 16 13:18:24] NOTICE[30363] chan_zap.c: PRI got event: No more alarm (5)
on Primary D-channel of span 1

So, i'm working with pri intense debug span X, but the output is quite
difficult to understand, what should i check from this output? Where should
i find some information in order to make a debug and talk to my telco with
"power" :)

Here is an example of pri intense debug span:

< Supervisory frame:
[Sep 16 13:59:25] < SAPI: 00  C/R: 0 EA: 0
<  TEI: 000EA: 1
[Sep 16 13:59:25] < Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
< N(R): 024 P/F: 1
< 0 bytes of data
[Sep 16 13:59:25] Handling message for SAPI/TEI=0/0
[Sep 16 13:59:25] -- ACKing all packets from 23 to (but not including) 24
[Sep 16 13:59:25] -- Since there was nothing left, stopping T200 counter
[Sep 16 13:59:25] -- Stopping T203 counter since we got an ACK
[Sep 16 13:59:25] -- Nothing left, starting T203 counter
[Sep 16 13:59:25] -- Got RR response to our frame
[Sep 16 13:59:25] -- Restarting T203 counter

Thanks in advance

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Re: [asterisk-users] Configure Asterisk with openssl

[Nikhil]

  On 09/16/2010 04:11 PM, A J Stiles wrote:
> On Thursday 16 Sep 2010, Nikhil wrote:
>>Hi
>> I got the bellow error when I try to configure asterisk code.
>>


>> $./configure --with-ssl=/usr/local/ssl
>> ...
>> ...
>> ...
>> checking for mandatory modules:  OPENSSL... fail
>>
>> configure: ***
>> configure: *** The OPENSSL installation appears to be missing or broken.
>> configure: *** Either correct the installation, or run configure
>> configure: *** including --without-ssl.
>>
>> I installed openssl in "/usr/local/ssl/" .
>>
>> Please help me to resolve this as I need to configure asterisk with SIP
>> in TLS ,I am using 1.8 beta5 version . Or give  some link where I am
>> find how to make asterisk to work with SIP on TLS.
> Did you remember to run ldconfig after installing ssl?
>
> What OS are you running?  If some kind of Linux, why not just install the
> openssl and openssl-devel packages?
>
I am using centOS-5 and I installed the ssl from source 
code(openssl-0.9.8k),but I think that is not the problem..Did anyone 
face the same type of error when doing configure the asterisk code?if 
yes how u solved that problem.?please help me on this.

Thanks
Nikhil


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[asterisk-users] Realtime semi-colon

[Andrew Thomas]

Hi list,

Does anyone know how to send * a semi-colon from a realtime database.  I
know that * uses the semi-colon as a 'seperator' - but I need to be able
to use one in a command.  I know I can use \; in the non-realtime
configs, but this doesn't work in realtime.

Cheers,
Andrew Thomas
Technical Services Manager
DataVox Ltd
Saddleworth Business Centre
Huddersfield Road
Delph, Oldham
OL3 5DF 


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Re: [asterisk-users] a2billing

[Vardan Harutyunyan]

Hello
You has installed a2b 1.7 version, and also had not do some permissions 
on folder and files.

/usr/local/src/a2billing/admin/templates_c'. Be sure $compile_dir is 
writable by the web server user. in 
/usr/local/src/a2billing/common/lib/smarty/Smarty.class.php on line 1093

I think the best place to find some help and solution for a2b - its a2b 
forum.


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Enterprise Incubator Foundation
123 Hovsep Emin Street,
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Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
www.eif-it.com

César Pinto Magán wrote:
> Hello,
> You sould go to the admin page (a2billing/admin/). There are two
> possibles web pages for a2b: the admin page and the customer page. You
> should point to the one you like in each moment :)
> César Pinto
> Alhambra-Eidos
>
> 
> *De:* asterisk-users-boun...@lists.digium.com en nombre de Flavio Miranda
> *Enviado el:* jue 16/09/2010 2:24
> *Para:* Asterisk Asterisk
> *Asunto:* [asterisk-users] a2billing
>
> Hey there,
>
> I am trying to setup a2billing on asterisk 1.6 , but ,when I try to
> access its web page I see the a2billing directories:
>
>
>   Index of /a2billing
>
> [ICO] Name Last 
> modified
>    Size
>    Description
> 
> 
> [DIR] Parent Directory   -
> [DIR] admin/ 
> 15-Sep-2010
> 19:19 -
> [DIR] agent/ 
> 15-Sep-2010
> 19:21 -
> [DIR] common/   
> 15-Sep-2010
> 19:18 -
> [DIR] customer/ 
> 15-Sep-2010 19:20 -
> 
>
> Apache/2.2.9 (Debian) PHP/5.2.6-1+lenny8 with Suhosin-Patch Server at
>
>
>
>
> Att,
>
> Flavio Roberto Miranda
> MSN:flaviormira...@hotmail.com
> Skype: flaviormiranda
>


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Re: [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11

[Philipp von Klitzing]

Hi!

> Does this shine new light to the problem ?!

No. Once more: Go and read doc/backtrace.txt.

And check if you have any meaningful information in /var/log/messages for 
the timestamp when asterisk crashed.

Philipp


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Re: [asterisk-users] Configure Asterisk with openssl

[A J Stiles]

On Thursday 16 Sep 2010, Nikhil wrote:
>   Hi
>I got the bellow error when I try to configure asterisk code.
>
> $./configure --with-ssl=/usr/local/ssl
> ...
> ...
> ...
> checking for mandatory modules:  OPENSSL... fail
>
> configure: ***
> configure: *** The OPENSSL installation appears to be missing or broken.
> configure: *** Either correct the installation, or run configure
> configure: *** including --without-ssl.
>
> I installed openssl in "/usr/local/ssl/" .
>
> Please help me to resolve this as I need to configure asterisk with SIP
> in TLS ,I am using 1.8 beta5 version . Or give  some link where I am
> find how to make asterisk to work with SIP on TLS.

Did you remember to run ldconfig after installing ssl?

What OS are you running?  If some kind of Linux, why not just install the 
openssl and openssl-devel packages?

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[asterisk-users] Configure Asterisk with openssl

[Nikhil]

 Hi
  I got the bellow error when I try to configure asterisk code.

$./configure --with-ssl=/usr/local/ssl
...
...
...
checking for mandatory modules:  OPENSSL... fail

configure: ***
configure: *** The OPENSSL installation appears to be missing or broken.
configure: *** Either correct the installation, or run configure
configure: *** including --without-ssl.

I installed openssl in "/usr/local/ssl/" .

Please help me to resolve this as I need to configure asterisk with SIP 
in TLS ,I am using 1.8 beta5 version . Or give  some link where I am 
find how to make asterisk to work with SIP on TLS.


Thanks
Nikhil
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Re: [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11

[Jonas Kellens]

Hello,

I have new information from a newly created test environment :


[Sep 16 13:41:01] -- Executing [...@macro-vakantie:1] 
MYSQL("SIP/test1-0008", "Connect connid localhost username passwd 
AsteriskHosted") in new stack
[Sep 16 13:41:01] -- Executing [...@macro-vakantie:2] 
MYSQL("SIP/test1-0008", "Query resultid 1 SELECT ast1, ast2, na, 
naID FROM vakantiedata where ID=59") in new stack

asterisk16*CLI>
Disconnected from Asterisk server
[Sep 16 13:41:01] Executing last minute cleanups
/usr/sbin/safe_asterisk: line 145:  3508 Segmentation fault  (core 
dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS} 
${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY}

Asterisk ended with exit status 139
Asterisk exited on signal 11.


Does this shine new light to the problem ?!

Kind regards,
Jonas.


On 09/15/2010 06:49 PM, jon pounder wrote:

On 09/15/2010 12:42 PM, Leif Madsen wrote:
   

On 10-09-15 05:25 AM, Jonas Kellens wrote:

 

I think I've found it :

Asterisk always reboots on this part :

[Sep 15 11:16:32] -- Goto (azura,pbx,1)
[Sep 15 11:16:32] -- Executing [...@azura:1]
NoOp("SIP/INTERTELin-", "3252480333 = pbx formule") in new stack
[Sep 15 11:16:32] -- Executing [...@azura:2]
Set("SIP/INTERTELin-", "CDR(accountcode)=AZURAin") in new stack
[Sep 15 11:16:32] -- Executing [...@azura:3]
Set("SIP/INTERTELin-", "BRON="473555006"<473555006>") in new stack
[Sep 15 11:16:32] -- Executing [...@azura:4]
Goto("SIP/INTERTELin-", "vakantie") in new stack
[Sep 15 11:16:32] -- Goto (azura,pbx,5)
[Sep 15 11:16:32] -- Executing [...@azura:5]
Macro("SIP/INTERTELin-", "vakantie,58") in new stack
[Sep 15 11:16:32] -- Executing [...@macro-vakantie:1]
MYSQL("SIP/INTERTELin-", "Connect connid localhost username
passwd AsteriskHosted") in new stack
[Sep 15 11:16:32] -- Executing [...@macro-vakantie:2]
MYSQL("SIP/INTERTELin-", "Query resultid 1 SELECT ast1 , ast2 ,
na , naID FROM vakantiedata where ID=58") in new stack
vps2301*CLI>
Disconnected from Asterisk server
[Sep 15 11:16:32] Executing last minute cleanups


Dialplan :

[macro-vakantie]
exten =>   s,1,MYSQL(Connect connid localhost username passwd AsteriskHosted)
exten =>   s,n,MYSQL(Query resultid ${connid} SELECT ast1 , ast2 , na ,
naID FROM vakantiedata where ID=${ARG1})
exten =>   s,n,MYSQL(Fetch fetchid ${resultid} AST1 AST2 NA naID )
exten =>   s,n,NoOp(vakantie-ast1 = ${AST1} vakantie-ast2 = ${AST2} na =
${NA} naID = ${naID})
exten =>   s,n,MYSQL(Clear ${resultid})
exten =>   s,n,MYSQL(Disconnect ${connid})

exten =>   s,n,NoOp(fetchid = ${fetchid})
exten =>   s,n,GoToIf($["${fetchid}"=="0"]?exit)

exten =>   s,n,NoOp()
exten =>   s,n,GoToIfTime(${AST1}?opvakantie)
exten =>   s,n,GoToIfTime(${AST2}?opvakantie)

exten =>   s,n(exit),NoOp()
exten =>   s,n,Set(vakantieresult=continue)
exten =>   s,n,MacroExit

exten =>   s,n(opvakantie),NoOp(op vakantie !)
exten =>   s,n,GoToIf($["${NA}"="hangup"]?hangup:route)


Do you guys see why Asterisk has problems with this part of the dialplan ?!
   


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Re: [asterisk-users] a2billing

[César Pinto Magán]

Hello,
 
You sould go to the admin page (a2billing/admin/). There are two possibles web 
pages for a2b: the admin page and the customer page. You should point to the 
one you like in each moment :)
 
César Pinto
Alhambra-Eidos



De: asterisk-users-boun...@lists.digium.com en nombre de Flavio Miranda
Enviado el: jue 16/09/2010 2:24
Para: Asterisk Asterisk
Asunto: [asterisk-users] a2billing


Hey there, 

  
I am trying to setup a2billing on asterisk 1.6 , but ,when I try to access its 
web page I see the a2billing directories:

Index of /a2billing

[ICO]Name Last modified 
   Size 
Description 



[DIR]Parent Directory   -  
[DIR]admin/ 15-Sep-2010 
19:19   -  
[DIR]agent/ 15-Sep-2010 
19:21   -  
[DIR]common/   15-Sep-2010 
19:18   -  
[DIR]customer/   
15-Sep-2010 19:20   -  


Apache/2.2.9 (Debian) PHP/5.2.6-1+lenny8 with Suhosin-Patch Server at 



Att,
 
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda


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Re: [asterisk-users] Speech To Text on linux with asterisk

[Nickolay V. Shmyrev]

В Чтв, 16/09/2010 в 12:44 +0530, DHAVAL INDRODIYA пишет:
> Thanks for update if a file is converted to text then where can i find
> a text file like after running 
> pocketsphinx_continuous command where text saved.

Text is in the last line:

0: we've entered the property the identification number of a
conflict



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Re: [asterisk-users] Speech To Text on linux with asterisk

[DHAVAL INDRODIYA]

Thanks for update if a file is converted to text then where can i find a
text file like after running
pocketsphinx_continuous command where text saved.

regards
dhaval

On Thu, Sep 16, 2010 at 12:29 PM, Nickolay V. Shmyrev  wrote:

> В Чтв, 16/09/2010 в 10:35 +0530, DHAVAL INDRODIYA пишет:
> >
> > Hi Nickolay,
> >
> > here i attached my file. please have a look into it.
>
> Hello DHAVAL
>
> As I wrote your file has wrong format.
>
>  $ file ask-propertyid.WAV
>   ask-propertyid.WAV: RIFF (little-endian) data, WAVE audio,
>   GSM 6.10, mono 8000 Hz
>
> See GSM 6.10 there. You need to convert it to PCM
>
>  sox ask-propertyid.WAV -e signed-integer ask-propertyid-converted.WAV
>
> Then decode.
>
>
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[asterisk-users] asterisk 1.6 and BLF

[Jonas Kellens]

Hello list,

are there special things that needs to be done when converting BLF from 
asterisk 1.4 tot 1.6.2 ?!


I have replaced call-limit with call-counter, but it seems that the 
lights on the phone no longer give the status of the extension they monitor.


On Snom phones, when the lights should be blinking (indicating a ringing 
phone) the lights are lighting up constantly (as if the extension is busy).



I have not changed my hints in the dialplan.


What other steps do I need to take ?



Kind regards,

Jonas.
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Re: [asterisk-users] Speech To Text on linux with asterisk

[Nickolay V. Shmyrev]

В Чтв, 16/09/2010 в 10:35 +0530, DHAVAL INDRODIYA пишет:
> 
> Hi Nickolay,
> 
> here i attached my file. please have a look into it.

Hello DHAVAL

As I wrote your file has wrong format.

  $ file ask-propertyid.WAV 
   ask-propertyid.WAV: RIFF (little-endian) data, WAVE audio, 
   GSM 6.10, mono 8000 Hz

See GSM 6.10 there. You need to convert it to PCM

 sox ask-propertyid.WAV -e signed-integer ask-propertyid-converted.WAV

Then decode.



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