Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls

[Khalid Touati]

I will be trying with H263+ and let you guys know, thank you for the good
news! I almost lost hope in Asterisk enabling me to use video (with
identical softphones using same codec :) )


On Tue, Sep 2, 2014 at 6:35 PM, Eric Wieling  wrote:

> "core show codecs" does not show VP8 on my Asterisk 11.  I don't recall
> why we are not using H.264.  The novelty wore off long ago and few of our
> staff use video calling anymore.
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Laimbock
> Sent: Tuesday, September 02, 2014 9:36 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls
>
> On 02-09-14 22:52, Eric Wieling wrote:
> > A co-worker was doing video, I dislike video.  The phones were Polycom
> VVX, The settings on our FreePBX box (office PBX) on the Settings /
> Asterisk SIP Settings / Video section we have Video: Enabled, H.263 and
> H.263p are the only two video codecs enabled.
>
> Thanks Eric. The obvious difference is that your co-worker was using
> H.263/H.263p while Bria uses H.264 or VP8. videosupport=yes was present
> in my sip.conf so it might be the codec. Time for more tinkering.
>
> Thanks,
> Patrick
>
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Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls

[Khalid Touati]

so it seems Asterisk Versions does not support video I guess


On Mon, Sep 1, 2014 at 9:30 AM, Khalid Touati 
wrote:

> Any article that goes through this (seems to be tedious) task to add video
> support and patents?
>
>
> On Mon, Sep 1, 2014 at 8:14 AM, Joshua Colp  wrote:
>
>> Khalid Touati wrote:
>>
>>> Hi Guys,
>>>
>>
>> Kia ora,
>>
>>
>>  Do you know of any asterisk community version that does video codec
>>> trans-coding or in other words supports video? I have 1.8.8.1 and I see
>>> h263.c format files but can't see that codec in make menuselect. it
>>> might be just a license issue (if h263 has to have license), but not
>>> sure if community versions offer video calls at all.
>>>
>>
>> Video transcoding is both usually patent encumbered as well as
>> computationally expensive. Asterisk supports passing through the video
>> untouched, but that's about it.
>>
>> Cheers,
>>
>> --
>> Joshua Colp
>> Digium, Inc. | Senior Software Developer
>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
>> Check us out at: www.digium.com & www.asterisk.org
>>
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>
>
>
> --
> Khalid Touati
>
>


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Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls

[Khalid Touati]

Any article that goes through this (seems to be tedious) task to add video
support and patents?


On Mon, Sep 1, 2014 at 8:14 AM, Joshua Colp  wrote:

> Khalid Touati wrote:
>
>> Hi Guys,
>>
>
> Kia ora,
>
>
>  Do you know of any asterisk community version that does video codec
>> trans-coding or in other words supports video? I have 1.8.8.1 and I see
>> h263.c format files but can't see that codec in make menuselect. it
>> might be just a license issue (if h263 has to have license), but not
>> sure if community versions offer video calls at all.
>>
>
> Video transcoding is both usually patent encumbered as well as
> computationally expensive. Asterisk supports passing through the video
> untouched, but that's about it.
>
> Cheers,
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
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[asterisk-users] Does Asterisk 1.8. Supports Video Calls

[Khalid Touati]

Hi Guys,
Do you know of any asterisk community version that does video codec
trans-coding or in other words supports video? I have 1.8.8.1 and I see
h263.c format files but can't see that codec in make menuselect. it might
be just a license issue (if h263 has to have license), but not sure if
community versions offer video calls at all.
Thank you!

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Re: [asterisk-users] Video Softphone for Android and PC

[khalid touati]

Thank you..I will use linphone phone then since it's there for both.

On Thu, Feb 21, 2013 at 10:18 PM, Bharat Lalcheta
wrote:

> linphone and sipdroid is there to use in android.
>
> For windows you can have many choices, openphone, linphone etc.
>
> On Fri, Feb 22, 2013 at 9:45 AM, khalid touati wrote:
>
>> Hi Guys,
>> I wanted to try video on my asterisk 1.8, I was wondering if you guys
>> know a good softphone able to make video calls and compatible with PC (win
>> 7) and Android? if yes do you guys know what codec is being used for video,
>> if any Codec should be added to Asterisk 1.8?
>> I appreciate your help!
>>
>>
>> --
>> Khalid Touati
>>
>>
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>
>
>
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[asterisk-users] Video Softphone for Android and PC

[khalid touati]

Hi Guys,
I wanted to try video on my asterisk 1.8, I was wondering if you guys know
a good softphone able to make video calls and compatible with PC (win 7)
and Android? if yes do you guys know what codec is being used for video, if
any Codec should be added to Asterisk 1.8?
I appreciate your help!


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Re: [asterisk-users] # button behavior

[khalid touati]

Thank you Jeremy! I just posted the solution :)

On Wed, Jun 27, 2012 at 4:17 PM, Jeremy Kister  wrote:

> On 6/27/2012 3:44 PM, khalid touati wrote:
>
>> #, this happened:
>> -- Started music on hold, class 'default', on SIP/USPBX2-07d5
>> --  Playing 'pbx-transfer.gsm' (language 'en')
>> and it gets disconnected. Anyone has a clue?
>>
>
>
> do you have # assigned in /etc/asterisk/features.conf ? perhaps to put the
> caller on hold ?
>
> --
>
> Jeremy Kister
> http://jeremy.kister.net./
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Re: [asterisk-users] # button behavior

[khalid touati]

That's fine guys I figured it out:
under features.conf:
[featuremap]
;blindxfer => #1; Blind transfer  (default is #) -- Make
sure to set the T and/or t option in the Dial() or Queue() app call!
blindxfer => *

I changed it to * and got rid of the pb
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[asterisk-users] # button behavior

[khalid touati]

Hi All,
We have a 1.6.2.6 Asterisk box connected to a 1.2 asterisk box, when people
dial to the conference room in 1.2 (from 1.6), of course they are prompted
for a room number and flush it by dialing # sign, the problem when they hit
#, this happened:
-- Started music on hold, class 'default', on SIP/USPBX2-07d5
--  Playing 'pbx-transfer.gsm' (language 'en')
and it gets disconnected. Anyone has a clue?
Thank you!

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[asterisk-users] Voicemail attachment format

[khalid touati]

Hi All,
I have a simple urgent question that I couldn't find the answer yet, can we
customize the voicemail attachment format *per user* in asterisk *1.2 *(like
all receive wav attch but one or two users receive attch in gsm format)? if
yes can you show me how please?

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Re: [asterisk-users] British Telecom ISDN BRI line issues

[khalid touati]

It turned out that we were just using a non matching configuration from
Dahdi tools, we were trying to use spans that has no modules inside, once
we configured only the spans that has BRI modules (under systems.conf and
dahdi-channels.conf), things started to work normally.

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Re: [asterisk-users] British Telecom ISDN BRI line issues

[khalid touati]

>Good to hear you got it working with Digium's help. One thing: if
bri_presistentlayer means that the drivers will force the >D-channel to
always be up then do not be surprised if BT disables the ISDN port. They
don't like it if a customer forces them >to power the D-channel all the
time at their expense. And with their ISDN team not contactable it may be a
bit of a >challenge >to get them to enable the port again (after you
promised not to mess with heir D-channel again...).

Hi Patrick,
it seems like you have the magic ball, I think what you described is
exactly what happened:
After we tested the server+ link and we were able to have simultaneous
calls (as expected), and knowing that this server was not touched (not even
rebooted), it is back not dialing through PTP link.The server remained
working from Friday to at least Monday then boom, when I called BT...of
course no explanation I am just wondering: their mechanism to miss up
things, is it automatic or manual ( I think automatic).
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Re: [asterisk-users] British Telecom ISDN BRI line issues

[khalid touati]

My Issie is finally fixed and I can make calls, I received actually from
digium the fix, I'll try to give as much details as I can to make sure
people who find this thread understand pb and solution.

Problem: not able to dial calls using BRI from British Telecom configured
as "system access" or PTP ("standard access which is PTMP works fine) (see
above for errors thrown)
Solution: adding this line "options wctdm24xxp bri_persistentlayer1=1" to
/etc/modprobe.d/dahdi.conf and restarting asterisk and dahdi of course.
Versions used: asterisk 1.6.2.6 dahdi 2.3.0.1 libpri 1.4.10.2
Special thanks to Patrick! thanks and good luck to all!

On Fri, May 11, 2012 at 12:13 PM, Richard Mudgett wrote:

> > Thank you for your reply Patrick!
> > for the first situation, I did try asterisk 1.6.2.6 and dahdi 2.3 but
> > with no success.
> > Can anyone suggest a combination that works till a patch is released?
>
> The patch for the layer1_presence option is in Asterisk 1.8.13.0-rc1 and
> 10.5.0-rc1.
> See chan_dahdi.conf.sample for a description of the new option.
>
> I added a comment to
> https://issues.asterisk.org/jira/browse/ASTERISK-13176
> saying which SVN revisions the patches were committed.
>
> Richard
>
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Re: [asterisk-users] British Telecom ISDN BRI line issues

[khalid touati]

Thank you for your reply Patrick!
for the first situation, I did try asterisk 1.6.2.6 and dahdi 2.3 but with
no success.
Can anyone suggest a combination that works till a patch is released?

On Thu, May 10, 2012 at 10:48 PM, Patrick Lists <
asterisk-l...@puzzled.xs4all.nl> wrote:

> Hi Khalid,
>
> Judging from that bug report I *think*:
>
> On 11-05-12 03:39, khalid touati wrote:
> > Patrick,
> > I got confused though is this true:
> > Any Asterisk soft+digium hdw => it doesn't work
>
> There seem to be combinations that do work. It is my understanding from
> that bugreport that an older libpri works with an older version of
> asterisk that does not have this issue. If your goal is to deploy the
> latest-and-greatest libpri, dahdi and asterisk 1.8 releases then it does
> not seem to work.
>
> > Any Asterisk soft+sangoma hdw => it works
>
> In my experience yes. Same goes for Eicon Diva Server cards.
>
> > Patched asterisk soft+digium hdw => it will work (per Kevin)
>
> Yes per Kevin's comment.
>
> Regards,
> Patrick
>
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Re: [asterisk-users] British Telecom ISDN BRI line issues

[khalid touati]

Patrick,
I got confused though is this true:
Any Asterisk soft+digium hdw => it doesn't work
Any Asterisk soft+sangoma hdw => it works
Patched asterisk soft+digium hdw => it will work (per Kevin)
On May 10, 2012 9:06 PM, "khalid touati"  wrote:

> Thank you Kevin! thanks Patrickhope a new release will come out soon!
>
> On Thu, May 10, 2012 at 7:37 PM, Patrick Lists <
> asterisk-l...@puzzled.xs4all.nl> wrote:
>
>> On 10-05-12 23:47, Kevin P. Fleming wrote:
>> > On 05/10/2012 03:20 PM, khalid touati wrote:
>> >> Thank you Patrick for the detailed info, it does make perfect sense to
>> >> me, I never expected that Digium cards have such an problem!
>> >
>> > There are patches in the works already (being tested by users in Europe)
>> > to deal with this layer 1 issue. Upcoming releases of DAHDI and Asterisk
>> > should have support for it.
>>
>> Thanks for the update Kevin. That's good to know. I look forward to the
>> new releases.
>>
>> Regards,
>> Patrick
>>
>>
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>
>
>
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> CCNA
>
>
>
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Re: [asterisk-users] British Telecom ISDN BRI line issues

[khalid touati]

Thank you Kevin! thanks Patrickhope a new release will come out soon!

On Thu, May 10, 2012 at 7:37 PM, Patrick Lists <
asterisk-l...@puzzled.xs4all.nl> wrote:

> On 10-05-12 23:47, Kevin P. Fleming wrote:
> > On 05/10/2012 03:20 PM, khalid touati wrote:
> >> Thank you Patrick for the detailed info, it does make perfect sense to
> >> me, I never expected that Digium cards have such an problem!
> >
> > There are patches in the works already (being tested by users in Europe)
> > to deal with this layer 1 issue. Upcoming releases of DAHDI and Asterisk
> > should have support for it.
>
> Thanks for the update Kevin. That's good to know. I look forward to the
> new releases.
>
> Regards,
> Patrick
>
>
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Re: [asterisk-users] British Telecom ISDN BRI line issues

[khalid touati]

Thank you Patrick for the detailed info, it does make perfect sense to me,
I never expected that Digium cards have such an problem!

On Thu, May 10, 2012 at 4:13 PM, Patrick Lists <
asterisk-l...@puzzled.xs4all.nl> wrote:

> On 10-05-12 21:10, khalid touati wrote:
> > Hi All,
> > I did downgrade to asterisk 1.6.2.6 and dahdi 2.3 which is working with
> > other BRI line (in our NL office), but I get this type of errores:
> >
> > -- Called G1/0788744550
> > [May 10 20:07:20] ERROR[27479]: chan_dahdi.c:12280 dahdi_pri_error: ACK
> > received for '1' outside of window of '0' to '0', restarting
> >   == Primary D-Channel on span 3 down
> > [May 10 20:07:20] WARNING[27479]: chan_dahdi.c:4158 pri_find_dchan: No
> > D-channels available!  Using Primary channel 9 as D-channel anyway!
> >   == Primary D-Channel on span 3 up
> > [May 10 20:07:20] ERROR[27480]: chan_dahdi.c:12280 dahdi_pri_error:
> > Huh!? no master found
> >
> > I didn't change my config in my previous post, anyone familiar with this
> > type of errors?
>
> No but there is a bug report with a lot of information that seems
> similar: https://issues.asterisk.org/jira/browse/14031
>
> In Europe telco's drop the D-channel (cut off power) to save on the
> electric bill. The libpri/dahdi/asterisk combo should detect a dropped
> D-channel and signal the telco to fire up the D-channel. Judging from
> that bugreport ("Unresolved") it seems Digium has still not succeeded in
> properly handling this situation.
>
> Should you not be able to resolve this issue and really require an ISDN
> BRI connection then have a look at an Eicon Diva or Sangoma card. Both
> cards+drivers properly handle a dropped D-channel. I have used Eicon
> Diva and Sangoma BRI cards in the Netherlands and they work fine. Or you
> could get a blackbox from Beronet, Patton, Audiocodes etc. If you feel
> adventurous you can also get a BRI card with a HFC-S Cologne chipset and
> get the latest and greatest isdn4k-utils, mISDN, mISDNuser and lcr from
> http://misdn.eu, build, install and configure lcr to talk to asterisk. A
> few weeks ago I set it up and did one test call and that call worked
> fine. Use at own risk :)
>
> Regards,
> Patrick
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>



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Network Administrator at Endosoft, LLC
CCNA
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Re: [asterisk-users] British Telecom ISDN BRI line issues

[khalid touati]

Hi All,
I did downgrade to asterisk 1.6.2.6 and dahdi 2.3 which is working with
other BRI line (in our NL office), but I get this type of errores:

-- Called G1/0788744550
[May 10 20:07:20] ERROR[27479]: chan_dahdi.c:12280 dahdi_pri_error: ACK
received for '1' outside of window of '0' to '0', restarting
  == Primary D-Channel on span 3 down
[May 10 20:07:20] WARNING[27479]: chan_dahdi.c:4158 pri_find_dchan: No
D-channels available!  Using Primary channel 9 as D-channel anyway!
  == Primary D-Channel on span 3 up
[May 10 20:07:20] ERROR[27480]: chan_dahdi.c:12280 dahdi_pri_error: Huh!?
no master found

I didn't change my config in my previous post, anyone familiar with this
type of errors?


On Wed, May 9, 2012 at 3:09 PM, khalid touati wrote:

> Yeah they have a wonderful policy that says "ISDN team are not
> contactable" :(   thanks a lot!!
>
>
> On Wed, May 9, 2012 at 3:06 PM, Patrick Lists <
> asterisk-l...@puzzled.xs4all.nl> wrote:
>
>> On 09-05-12 20:57, khalid touati wrote:
>> > Yeah sorry for that, I realized something is missing after I sent the
>> > email, but it is exactly what I have (other than order here, which
>> > doesn't really matter: you posted ami,te,term, I have ami,term,te).
>> > Actually I had couple technicians from digium look at it and they said
>> > BT equipements is not responding to the card within a certain range that
>> > the card is looking for (i'm not sure what range but I do believe too
>> > it's a BT issue), But I have run all the couple command that Patrick
>> > suggested (to double check), tested again and still same kind of errors.
>> > But Thank you very much Patrick for the guide, I was looking for that
>> > it's been a couple days!!
>> > I just hope someone that has the exact same issue or someone with
>> > previous BT experience see this and help :) ..we never know :) !
>>
>> Too bad you could not (yet) make it work. Hope you get somewhere with
>> BT. Once you get past the people following those silly scripts you
>> should be able to talk to someone who has a clue and resolve this issue.
>>
>> Good luck!
>>
>> Regards,
>> Patrick
>>
>> --
>> _
>> -- 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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>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> Khalid Touati
> Network Administrator at Endosoft, LLC
> CCNA
>
>
>


-- 
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Network Administrator at Endosoft, LLC
CCNA
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Re: [asterisk-users] Belgian BRI (euroisdn): what to use for a B410P

[khalid touati]

Hi Bart,
here is a working configuration in Netherlands:
/etc/dahdi/system.conf:

span = 1,1,0,ccs,ami
bchan = 1,2
hardhdlc = 3

span = 2,1,0,ccs,ami
bchan = 4,5
hardhdlc = 6

span = 3,1,0,ccs,ami
bchan = 7,8
hardhdlc = 9

span = 4,1,0,ccs,ami
bchan = 10,11
hardhdlc = 12

loadzone= nl
defaultzone= nl   (of course change those to your country initials)

/etc/asterisk/chan_dahdi.conf:

group = 1
signalling = bri_cpe_ptmp
switchtype = euroisdn
context = mainmenu
echocancel = yes
channel => 1,2,4,5,7,8,10,11

I am not using dahdi-channels, hope it helps!


On Wed, May 9, 2012 at 1:59 PM, Bart Coninckx wrote:

> Hi,
>
> I'm experiencing difficulties to get a B410P running with Asterisk 10.3.1
> and DAHDI 2.6.1.
> Am I supposed to use DAHDI for this card and ISDN BRI for my country
> (Belgium)?
>
> thx,
>
> BC
>
> --
> __**__**_
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>  
> http://lists.digium.com/**mailman/listinfo/asterisk-**users<http://lists.digium.com/mailman/listinfo/asterisk-users>
>



-- 
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Network Administrator at Endosoft, LLC
CCNA
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Re: [asterisk-users] British Telecom ISDN BRI line issues

[khalid touati]

Yeah they have a wonderful policy that says "ISDN team are not contactable"
:(   thanks a lot!!

On Wed, May 9, 2012 at 3:06 PM, Patrick Lists <
asterisk-l...@puzzled.xs4all.nl> wrote:

> On 09-05-12 20:57, khalid touati wrote:
> > Yeah sorry for that, I realized something is missing after I sent the
> > email, but it is exactly what I have (other than order here, which
> > doesn't really matter: you posted ami,te,term, I have ami,term,te).
> > Actually I had couple technicians from digium look at it and they said
> > BT equipements is not responding to the card within a certain range that
> > the card is looking for (i'm not sure what range but I do believe too
> > it's a BT issue), But I have run all the couple command that Patrick
> > suggested (to double check), tested again and still same kind of errors.
> > But Thank you very much Patrick for the guide, I was looking for that
> > it's been a couple days!!
> > I just hope someone that has the exact same issue or someone with
> > previous BT experience see this and help :) ..we never know :) !
>
> Too bad you could not (yet) make it work. Hope you get somewhere with
> BT. Once you get past the people following those silly scripts you
> should be able to talk to someone who has a clue and resolve this issue.
>
> Good luck!
>
> Regards,
> Patrick
>
> --
> _
> -- 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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Network Administrator at Endosoft, LLC
CCNA
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Re: [asterisk-users] British Telecom ISDN BRI line issues

[khalid touati]

Yeah sorry for that, I realized something is missing after I sent the
email, but it is exactly what I have (other than order here, which doesn't
really matter: you posted ami,te,term, I have ami,term,te).
Actually I had couple technicians from digium look at it and they said BT
equipements is not responding to the card within a certain range that the
card is looking for (i'm not sure what range but I do believe too it's a BT
issue), But I have run all the couple command that Patrick suggested (to
double check), tested again and still same kind of errors.
But Thank you very much Patrick for the guide, I was looking for that it's
been a couple days!!
I just hope someone that has the exact same issue or someone with previous
BT experience see this and help :) ..we never know :) !

On Wed, May 9, 2012 at 2:40 PM, Patrick Lists <
asterisk-l...@puzzled.xs4all.nl> wrote:

> On 09-05-12 19:54, khalid touati wrote:
> > Thank you for your answer,  I think I posted dhadi version and so but
> > let me add more details and recap them below:
> >
> > We are using asterisk 1.8.12.0 with dahdi 2.6.0, on CentOS 6.2. it's a
> > digium card 1HA8-0400BLF
> >
> > output of dahdi_hardware: pci::04:08.0 wctdm24xxp+  d161:8008
> > HB8-
> >
> > From BT side: it is called by BT a "system access" ISDN2 BRI (per BT NT
> > mode and signaling as PTP)
> >
> > chan_dahdi.conf
> >
> > ; Span 1: WCBRI/0/0 "HB8-" (MASTER) AMI/CCS
> > group=1,11
> > context= mainmenu
> > switchtype = euroisdn
> > signalling = bri_cpe
> > channel => 1-2
> > context = default
> > group = 63
> >
> > ; Span 2: WCBRI/0/1 "HB8-" AMI/CCS
> > group=1,12
> > context=mainmenu
> > switchtype = euroisdn
> > signalling = bri_cpe
> > channel => 4-5
> > context = default
> > group = 63
> >
> > ; Span 3: WCBRI/0/2 "HB8-" AMI/CCS
> > group=1,13
> > context=mainmenu
> > switchtype = euroisdn
> > signalling = bri_cpe
> > channel => 7-8
> > context = default
> > group = 63
> >
> > ; Span 4: WCBRI/0/3 "HB8-" AMI/CCS
> > group=1,14
> > context=mainmenu
> > switchtype = euroisdn
> > signalling = bri_cpe
> > channel => 10-11
> > context = default
> > group = 63
> >
> > the error when placing a call : *Span: 4 TEI=0 MDL-ERROR (J): N(R) error
> > in state 7(Multi-frame established)
> > *
> > Thank you!!
>
> You did not provide system.conf. Do you have something like this (may
> have errors, I did not check):
>
> loadzone = uk
> defaultzone = uk
> span => 1,1,0,ccs,ami,te,term
> bchan = 1,2
> hardhdlc = 3
>
> span => 2,2,0,ccs,ami,te,term
> bchan = 4,5
> hardhdlc = 6
>
> span => 3,3,0,ccs,ami,te,term
> bchan = 7,8
> hardhdlc = 9
>
> span => 4,4,0,ccs,ami,te,term
> bchan = 10,11
> hardhdlc = 12
>
>
> Then as root:
> modprobe wctdm23xxp
>
> And as root:
> dahdi_cfg -vvv
>
> And check if all is well (green leds, happy messages in
> /var/log/messages, etc.).
>
>
> Then in chan_dahdi.conf use something like:
>
> ;BRI Module
> group = 1
> signalling = bri_cpe
> context = incoming
> channel => 1,2,4,5,7,8,10,11
>
> Your chan_dahdi.conf has "group" and "context" multiple times and that
> does not seem right (admittedly it's been ages since I setup a Digium
> card).
>
> Hope this helps. If not follow the installation manual step for step or
> call Digium support.
>
> http://docs.digium.com/H8/hx8_series_manual.pdf
>
> Regards,
> Patrick
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
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>



-- 
Khalid Touati
Network Administrator at Endosoft, LLC
CCNA
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Re: [asterisk-users] British Telecom ISDN BRI line issues

[khalid touati]

Thank you for your answer,  I think I posted dhadi version and so but let
me add more details and recap them below:

We are using asterisk 1.8.12.0 with dahdi 2.6.0, on CentOS 6.2. it's a
digium card 1HA8-0400BLF

output of dahdi_hardware: pci::04:08.0 wctdm24xxp+  d161:8008
HB8-

>From BT side: it is called by BT a "system access" ISDN2 BRI (per BT NT
mode and signaling as PTP)

chan_dahdi.conf
; Span 1: WCBRI/0/0 "HB8-" (MASTER) AMI/CCS
group=1,11
context= mainmenu
switchtype = euroisdn
signalling = bri_cpe
channel => 1-2
context = default
group = 63

; Span 2: WCBRI/0/1 "HB8-" AMI/CCS
group=1,12
context=mainmenu
switchtype = euroisdn
signalling = bri_cpe
channel => 4-5
context = default
group = 63

; Span 3: WCBRI/0/2 "HB8-" AMI/CCS
group=1,13
context=mainmenu
switchtype = euroisdn
signalling = bri_cpe
channel => 7-8
context = default
group = 63

; Span 4: WCBRI/0/3 "HB8-" AMI/CCS
group=1,14
context=mainmenu
switchtype = euroisdn
signalling = bri_cpe
channel => 10-11
context = default
group = 63

the error when placing a call : *Span: 4 TEI=0 MDL-ERROR (J): N(R) error in
state 7(Multi-frame established)
*
Thank you!!

On Wed, May 9, 2012 at 1:22 PM, Patrick Lists <
asterisk-l...@puzzled.xs4all.nl> wrote:

> On 09-05-12 18:46, khalid touati wrote:
> > Should I understand that no Asterisk user has issues with ISDN "system
> > access" configuration from UK? or maybe no one is using Asterisk In UK
> :) ?
>
> I have no idea. But other than the error you have given very little
> information to go on. Which card are you using, what type of ISDN line
> (PTP?), what's the DAHDI or Sangoma (or...) and Asterisk configuration,
> what do the log files say, etc?
>
> Have you tried calling the vendor of the ISDN card for support?
>
> Regards,
> Patrick
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>



-- 
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Network Administrator at Endosoft, LLC
CCNA
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Re: [asterisk-users] British Telecom ISDN BRI line issues

[khalid touati]

Should I understand that no Asterisk user has issues with ISDN "system
access" configuration from UK? or maybe no one is using Asterisk In UK :) ?

On Tue, May 8, 2012 at 12:46 PM, khalid touati wrote:

> Hi All,
> I am posting this thread with the hope that someone in UK (or elsewhere)
> had a similar issue:
> Our issue is simple, we cannnot use our ISDN line, when watching asterisk
> console it gives a bunch of ISDN errors where the following is probably the
> most relevant:
>
> Span: 4 TEI=0 MDL-ERROR (J): N(R) error in state 7(Multi-frame established)
>
> We are using asterisk 1.8.12.0 with dahdi 2.6.0, on CentOS 6.2. I can post
> configuration and/or further information if needed.
>
> --
> Khalid Touati
> Network Administrator
> CCNA
>
>
>


-- 
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Network Administrator at Endosoft, LLC
CCNA
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[asterisk-users] British Telecom ISDN BRI line issues

[khalid touati]

Hi All,
I am posting this thread with the hope that someone in UK (or elsewhere)
had a similar issue:
Our issue is simple, we cannnot use our ISDN line, when watching asterisk
console it gives a bunch of ISDN errors where the following is probably the
most relevant:

Span: 4 TEI=0 MDL-ERROR (J): N(R) error in state 7(Multi-frame established)

We are using asterisk 1.8.12.0 with dahdi 2.6.0, on CentOS 6.2. I can post
configuration and/or further information if needed.

-- 
Khalid Touati
Network Administrator
CCNA
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[asterisk-users] Layer2 Down in BRI connection

[khalid touati]

Hi All,
I am having an issue with layer 2 in BRi connection configured using Misdn:
after the BRI line working fine, a technician from our phone company came
in to add another number, after testing with his ISDN phone and BRI line is
working, from our asterisk server it is not :(.
When I check ports with "misdn show port x" it says:

* Port 4 Type TE Prot. PMP L2Link DOWN L1Link:UP Blocked:0  Debug:7
(as you can see here L1Link=UP *but* L2Link=DOWN)

since we didn't change any setting from our end, it maybe an change from
our phone provider end that caused to have a *mismatch in the layer 2
connection*.

*what we tried is:*
- we did reboot our phone system several times.
- I did reconfigure the ISDN trunk setting to refresh things (*using the
same config that worked before*)
- I did use another phone system that was not used before (to eliminate the
possibility that our current phone system has hardware issues) with no
success, it *displaying the same Layer 2 issue found*.
- I turned debugging on when making calls, it shows that calls go through
as programmed but it hits a down line (because of L2 issue) :
IP0x*CLI>
-- Executing [9020823541@COUNTRY_DIAL:1] misdn_check_l2l1("SIP/8425-
00438004", "g:trunk_m2|20") in new stack

IP0x*CLI>
P[ 0] Checking Ports in group: trunk_m2
P[ 0] trying port 1
P[ 0] trying port 2
P[ 0] trying port 3
P[ 0] trying port 4
-- Executing [9020823541@COUNTRY_DIAL:2] Dial("SIP/8425-00438004",
"mISDN/g:trunk_m2/020823541") in new stack
P[ 0]  --> Group Call group: trunk_m2
P[ 1] Group [trunk_m2] Port [1]
P[ 2] Group [trunk_m2] Port [2]
P[ 3] Group [trunk_m2] Port [3]
P[ 4] Group [trunk_m2] Port [4]
P[ 4] portup:1
IP0x*CLI>
P[ 0]  --> * NEW CHANNEL dad:020823541 oad:(null)
IP0x*CLI>
P[ 4] * Queuing chan 0x43d644
IP0x*CLI>
P[ 4] read_config: Getting Config
IP0x*CLI>
P[ 4] config_jb: Called
IP0x*CLI>
P[ 4]  --> * CallGrp: PickupGrp:
IP0x*CLI>
P[ 4]  --> TON: Unknown
IP0x*CLI>
P[ 4]  --> LTON: Unknown
IP0x*CLI>
P[ 4]  --> CTON: Unknown
IP0x*CLI>
P[ 4] * CALL: g:trunk_m2/020823541
IP0x*CLI>
P[ 4]  --> * dad:9020823541 tech:mISDN/6-u3 ctx:DID_trunk_m2
IP0x*CLI>
P[ 4]  --> * adding2newbc ext 9020823541
IP0x*CLI>
P[ 4]  --> * adding2newbc callerid 8425
IP0x*CLI>
P[ 4] update_config: Getting Config
IP0x*CLI>
P[ 4]  --> pres: -1 screen: -1
IP0x*CLI>
P[ 4]  --> pres: 0
IP0x*CLI>
P[ 4]  --> PRES: Allowed (0x0)
IP0x*CLI>
P[ 4]  --> SCREEN: Unscreened (0x0)
IP0x*CLI>
P[ 4] NO OPTS GIVEN
IP0x*CLI>
P[ 4] SENDEVENT: stack->nt:0 stack->uperid:4403
IP0x*CLI>
P[ 4] I SEND:SETUP oad:8425 dad:020823541 pid:5
IP0x*CLI>
P[ 4]  --> bc_state:BCHAN_CLEANED
IP0x*CLI>
P[ 4]  --> channel:0 mode:TE cause:16 ocause:16 rad: cad:
IP0x*CLI>
P[ 4]  --> info_dad: onumplan:0 dnumplan:0 rnumplan:0 cpnnumplan:0
IP0x*CLI>
P[ 4]  --> caps:Speech pi:0 keypad: sending_complete:0
IP0x*CLI>
P[ 4]  --> screen:0 --> pres:0
IP0x*CLI>
P[ 4]  --> addr:0 l3id:90003 b_stid:0 layer_id:0
IP0x*CLI>
P[ 4]  --> facility:Fac_None out_facility:Fac_None
IP0x*CLI>
P[ 4]  --> urate:0 rate:16 mode:0 user1:0
IP0x*CLI>
P[ 4]  --> bc:1130c58 h:0 sh:0
IP0x*CLI>
P[ 4] --> new_l3id 90004
IP0x*CLI>
P[ 4] Sending msg, prim:30580 addr:41000403 dinfo:90004
IP0x*CLI>
P[ 4]  --> * SEND: State Dialing pid:5
IP0x*CLI>
-- Called g:trunk_m2/020823541
IP0x*CLI>
P[ 4] handle_frm: frm->addr:42000402 frm->prim:3f182
P[ 4]  --> lib: RELEASE_CR Ind with l3id:90004
P[ 4]  --> lib: CLEANING UP l3id: 90004
P[ 4] I IND :CLEAN_UP oad:8425 dad:020823541 pid:5 state:CALLING
P[ 4] hangup_chan called
P[ 4]  --> queue_hangup
P[ 4] release_chan: bc with l3id: 90004
P[ 4] * RELEASING CHANNEL pid:5 ctx:DID_trunk_m2 dad:020823541
oad:9020823541 state: CALLING
P[ 4]  --> * State Down
P[ 4]  --> Setting AST State to down
P[ 4] $$$ CLEANUP CALLED pid:5
P[ 4] $$$ Already cleaned up bc with stid :0
P[ 4] [Jan  3 16:59:14] DEBUG[2944]: chan_misdn.c:2525 misdn_hangup:
misdn_hangup(mISDN/6-u3)
P[ 0] misdn_hangup called, without 'chan_list' obj
  == Everyone is busy/congested at this time (1:0/0/1)
  == Auto fallthrough, channel 'SIP/8425-00438004' status is 'CHANUNAVAIL'
Idx:0 stack->cchan:0 in_use:0 Chan:1
P[ 4] Idx:1 stack->cchan:0 in_use:0 Chan:2
P[ 4] Idx:2 stack->cchan:0 in_use:0 Chan:3
P[ 0] Got empty Msg..

I appreciate any help!
-- 
Khalid
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[asterisk-users] Layer2 Down in BRI connection

[khalid touati]

Hi All,
I am having an issue with layer 2 in BRi connection configured using Misdn:
after the BRI line working fine, a technician from our phone company came
in to add another number, after testing with his ISDN phone and BRI line is
working, from our asterisk server it is not :(.
When I check ports with "misdn show port x" it says:

* Port 4 Type TE Prot. PMP L2Link DOWN L1Link:UP Blocked:0  Debug:7
(as you can see here L1Link=UP *but* L2Link=DOWN)

since we didn't change any setting from our end, it maybe an change from
our phone provider end that caused to have a *mismatch in the layer 2
connection*.

*what we tried is:*
- we did reboot our phone system several times.
- I did reconfigure the ISDN trunk setting to refresh things (*using the
same config that worked before*)
- I did use another phone system that was not used before (to eliminate the
possibility that our current phone system has hardware issues) with no
success, it *displaying the same Layer 2 issue found*.
- I turned debugging on when making calls, it shows that calls go through
as programmed but it hits a down line (because of L2 issue) :
IP0x*CLI>
-- Executing [9020823541@COUNTRY_DIAL:1]
misdn_check_l2l1("SIP/8425-00438004", "g:trunk_m2|20") in new stack

IP0x*CLI>
P[ 0] Checking Ports in group: trunk_m2
P[ 0] trying port 1
P[ 0] trying port 2
P[ 0] trying port 3
P[ 0] trying port 4
-- Executing [9020823541@COUNTRY_DIAL:2] Dial("SIP/8425-00438004",
"mISDN/g:trunk_m2/020823541") in new stack
P[ 0]  --> Group Call group: trunk_m2
P[ 1] Group [trunk_m2] Port [1]
P[ 2] Group [trunk_m2] Port [2]
P[ 3] Group [trunk_m2] Port [3]
P[ 4] Group [trunk_m2] Port [4]
P[ 4] portup:1
IP0x*CLI>
P[ 0]  --> * NEW CHANNEL dad:020823541 oad:(null)
IP0x*CLI>
P[ 4] * Queuing chan 0x43d644
IP0x*CLI>
P[ 4] read_config: Getting Config
IP0x*CLI>
P[ 4] config_jb: Called
IP0x*CLI>
P[ 4]  --> * CallGrp: PickupGrp:
IP0x*CLI>
P[ 4]  --> TON: Unknown
IP0x*CLI>
P[ 4]  --> LTON: Unknown
IP0x*CLI>
P[ 4]  --> CTON: Unknown
IP0x*CLI>
P[ 4] * CALL: g:trunk_m2/020823541
IP0x*CLI>
P[ 4]  --> * dad:9020823541 tech:mISDN/6-u3 ctx:DID_trunk_m2
IP0x*CLI>
P[ 4]  --> * adding2newbc ext 9020823541
IP0x*CLI>
P[ 4]  --> * adding2newbc callerid 8425
IP0x*CLI>
P[ 4] update_config: Getting Config
IP0x*CLI>
P[ 4]  --> pres: -1 screen: -1
IP0x*CLI>
P[ 4]  --> pres: 0
IP0x*CLI>
P[ 4]  --> PRES: Allowed (0x0)
IP0x*CLI>
P[ 4]  --> SCREEN: Unscreened (0x0)
IP0x*CLI>
P[ 4] NO OPTS GIVEN
IP0x*CLI>
P[ 4] SENDEVENT: stack->nt:0 stack->uperid:4403
IP0x*CLI>
P[ 4] I SEND:SETUP oad:8425 dad:020823541 pid:5
IP0x*CLI>
P[ 4]  --> bc_state:BCHAN_CLEANED
IP0x*CLI>
P[ 4]  --> channel:0 mode:TE cause:16 ocause:16 rad: cad:
IP0x*CLI>
P[ 4]  --> info_dad: onumplan:0 dnumplan:0 rnumplan:0 cpnnumplan:0
IP0x*CLI>
P[ 4]  --> caps:Speech pi:0 keypad: sending_complete:0
IP0x*CLI>
P[ 4]  --> screen:0 --> pres:0
IP0x*CLI>
P[ 4]  --> addr:0 l3id:90003 b_stid:0 layer_id:0
IP0x*CLI>
P[ 4]  --> facility:Fac_None out_facility:Fac_None
IP0x*CLI>
P[ 4]  --> urate:0 rate:16 mode:0 user1:0
IP0x*CLI>
P[ 4]  --> bc:1130c58 h:0 sh:0
IP0x*CLI>
P[ 4] --> new_l3id 90004
IP0x*CLI>
P[ 4] Sending msg, prim:30580 addr:41000403 dinfo:90004
IP0x*CLI>
P[ 4]  --> * SEND: State Dialing pid:5
IP0x*CLI>
-- Called g:trunk_m2/020823541
IP0x*CLI>
P[ 4] handle_frm: frm->addr:42000402 frm->prim:3f182
P[ 4]  --> lib: RELEASE_CR Ind with l3id:90004
P[ 4]  --> lib: CLEANING UP l3id: 90004
P[ 4] I IND :CLEAN_UP oad:8425 dad:020823541 pid:5 state:CALLING
P[ 4] hangup_chan called
P[ 4]  --> queue_hangup
P[ 4] release_chan: bc with l3id: 90004
P[ 4] * RELEASING CHANNEL pid:5 ctx:DID_trunk_m2 dad:020823541
oad:9020823541 state: CALLING
P[ 4]  --> * State Down
P[ 4]  --> Setting AST State to down
P[ 4] $$$ CLEANUP CALLED pid:5
P[ 4] $$$ Already cleaned up bc with stid :0
P[ 4] [Jan  3 16:59:14] DEBUG[2944]: chan_misdn.c:2525 misdn_hangup:
misdn_hangup(mISDN/6-u3)
P[ 0] misdn_hangup called, without 'chan_list' obj
  == Everyone is busy/congested at this time (1:0/0/1)
  == Auto fallthrough, channel 'SIP/8425-00438004' status is 'CHANUNAVAIL'
Idx:0 stack->cchan:0 in_use:0 Chan:1
P[ 4] Idx:1 stack->cchan:0 in_use:0 Chan:2
P[ 4] Idx:2 stack->cchan:0 in_use:0 Chan:3
P[ 0] Got empty Msg..

I appreciate any help!
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Re: [asterisk-users] India Telecom regulations

[khalid touati]

Thank you Raj,
I hope it will soon require no license as I heard there is a project to
change this law, for now I believe I will recommend our office in India to
go for license (to bridge to PSTN).
Thanks once more for your help!

2011/12/19 Raj Mathur (राज माथुर) 

> On Tuesday 20 Dec 2011, khalid touati wrote:
> > Thank you Raj,
> > so with VOIP license calls can go beyond our pbx to PSTN (india),
> > right, if so this what i needed to know to call Indian cellphone
> > from US (or  other countries)
>
> If your objective is to originate calls in the US (using whatever
> technology), route them over SIP and then terminate them to the PSTN in
> India, then yes: your Indian presence would need a VoIP licence.
> Similarly for the reverse: originate a call from Indian PSTN to your
> local office here and route it using VoIP to any destination (whether
> within India or abroad).  A licence is required in that case too.
>
> In general, interconnection of two different entities by bridging Indian
> PSTN with any other technology requires a licence.  If you're only doing
> VoIP-VoIP, or PSTN-PSTN, or bridging an Indian VoIP call to PSTN outside
> India then it's permitted in principle.  This is why, e.g., Skype is
> permitted: it doesn't connect to the Indian PSTN at any stage.
>
> Once again, IANAL and TINLA.  This is purely from my (mostly informed)
> understanding of the current laws.
>
> Regards,
>
> -- Raj
> --
> Raj Mathur  || r...@kandalaya.org   || GPG:
> http://otheronepercent.blogspot.com || http://kandalaya.org || CC68
> It is the mind that moves   || http://schizoid.in   || D17F
>
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Network Administrator at Endosoft, LLC
CCNA
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Re: [asterisk-users] India Telecom regulations

[khalid touati]

Thank you Raj,
so with VOIP license calls can go beyond our pbx to PSTN (india), right, if
so this what i needed to know to call Indian cellphone from US (or  other
countries)

On Mon, Dec 19, 2011 at 10:03 PM, Robert-IPhone wrote:

> Right check out Cordia.LT
>
>
> Sent from my iPhone 4S
>
> On Dec 19, 2011, at 9:58 PM, "Raj Mathur (राज माथुर)" <
> r...@linux-delhi.org> wrote:
>
> > On Tuesday 20 Dec 2011, Steve Edwards wrote:
> >> On Mon, 19 Dec 2011, Nick Khamis wrote:
> >>> SIP in India is illegal.
> >>
> >> What about IAX, Skype, VPN, etc?
> >
> > The only thing that is not permitted is bridging Internet calls with the
> > Indian PSTN.  In fact, that too is allowed if you have a VoIP licence
> > from the government.  Apart from that, as long as you continue using it
> > within your own organisation, any protocol is fine.
> >
> > IANAL.  TINLA.
> >
> > Regards,
> >
> > -- Raj
> > --
> > Raj Mathur  || r...@kandalaya.org   || GPG:
> > http://otheronepercent.blogspot.com || http://kandalaya.org || CC68
> > It is the mind that moves   || http://schizoid.in   || D17F
> >
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CCNA
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[asterisk-users] India Telecom regulations

[khalid touati]

Hi All,
Because I am pretty sure we have people in this DL from India, I was hoping
to get the 100% accurate information, is it legal to make calls from any
coutry to Indian mobile phones through an Asterisk server based in India?

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Re: [asterisk-users] Queuing outgoing calls

[khalid touati]

Hey Danny thanks a bunch! I really appreciate that.
Thank you Steve!

On Fri, Aug 12, 2011 at 3:05 PM, Danny Nicholas  wrote:

> The .call file can connect an internal number to an outside number
>
> Look at this sample
>
> Channel: DAHDI/R1/5551212
>
> MaxRetries: 2
>
> # Retry in 5 min
>
> RetryTime: 300
>
> WaitTime: 45
>
> Context: outgoing
>
> Extension:100
>
> Priority: 1
>
> ** **
>
> This sample call makes a call on DAHDI using Round Robin Group 1.  If the
> call can be made, it connects to internal extension 100.  So instead of your
> employee dialing 5551212 directly, they dial 1234 and enter 5551212 as the
> number to be dialed.  When a line becomes available and the call goes
> through, 100 is bridged in and the call is done
>
> ** **
>
> Exten => 1234,1,read(numtodial,enternum,10,skip,1,10)
>
> Exten => 1234,2,AGI(makecall.agi,${EXTEN},${numtodial})
>
> Exten => 1234,3,hangup()
>
> ** **
>
> Makecall.agi
>
> #!/bin/sh
>
> echo "extension: $1" > call1.tmp
>
> echo "maxtries: 3" >> call1.tmp
>
> echo "retrytime: 300" >> call1.tmp
>
> echo "Channel: DAHDI/R1/$2" >> call1.tmp
>
> echo "Priority: 1" >> call1.tmp
>
> chmod +x call1.tmp
>
> mv call1.tmp /var/spool/asterisk/outgoing
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *khalid touati
> *Sent:* Friday, August 12, 2011 9:56 AM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Queuing outgoing calls
>
> ** **
>
> Hi Danny,
> Thnks for your response but I googled "call-queueing" with no success, are
> your referring to the concept or a third party application or an Asterisk
> function..., can you please specify?
>
> On Fri, Aug 12, 2011 at 10:03 AM, Danny Nicholas 
> wrote:
>
> You can use “call-queueing” to accomplish this.  When your employee dials
> the number (555-1212 for demonstration purposes), instead of going directly
> out, the call goes to /var/spool/asterisk/outgoing as an entry.  When this
> entry comes up, the employee gets a call-back/connect to his/her party.  You
> would need to provide a 911 and/or executive loophole however.
>
>  
>
> ** **
>
> ** **
>
>
> --
> _________
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>
>
>
> --
> Khalid Touati
> Network Administrator at Endosoft, LLC
> CCNA
>
> 
>
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Re: [asterisk-users] Queuing outgoing calls

[khalid touati]

Hi Danny,
Thnks for your response but I googled "call-queueing" with no success, are
your referring to the concept or a third party application or an Asterisk
function..., can you please specify?

On Fri, Aug 12, 2011 at 10:03 AM, Danny Nicholas  wrote:

> You can use “call-queueing” to accomplish this.  When your employee dials
> the number (555-1212 for demonstration purposes), instead of going directly
> out, the call goes to /var/spool/asterisk/outgoing as an entry.  When this
> entry comes up, the employee gets a call-back/connect to his/her party.  You
> would need to provide a 911 and/or executive loophole however.
>
> ** **
>
>
>
> --
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[asterisk-users] Queuing outgoing calls

[khalid touati]

Hi All,
Usually we need to queue calls coming from outside our system and for that
we use queues.conf, in this case we have a lot of employees that are using
POTS (Dahdi or zap channels) and we want to make them go by order since we
have limited outgoing lines, does anybody have a clue what to use in this
case, is queues.conf will still be useful in the case figure?

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Re: [asterisk-users] menu issue

[khalid touati]

Hi I wanted to help out with dial plan, but it's not obvious what you want
to achieve, also I do recommend to read the chapter that talks about
contexts and dialplan from future of asterisk book. but if you're in rush
just try to make clear how you want your system to behave and i'll be glad
to help.

On Tue, Jun 21, 2011 at 9:22 AM, salaheddine elharit <
salah.elharit...@gmail.com> wrote:

>  Hello
>
>
>
> I have created the menu below, with this menu when I call 520460XXX I can
> hear the welcome message [home] context and when I press the # I can go to
> the [menu] context and hear the menu message
>
>
>
> When I press 1 in order to go to the [call] I can hear the call message
>
>
> Now I have 2 client sip 222 and 223 with x-lite, how can I do in order to
> receive the call in 223 with [call] and receive a call in 222 with [support]
>
> thanks and best regards
>
>
> exten => 520460XXX,1,Ringing()
> exten => 520460XXX,2,Wait(4)
> exten => 520460XXX,3,Goto(home,s,1)
>
> [home]
> exten => s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
> exten => s,2,Background(${sounds_path}welcome)
> exten => s,3,WaitExten(10)
>
> exten => #,1,Goto(menu,s,1)
> exten => i,1,Playback(${sounds_path}error-key)
> exten => t,1,Goto(home,s,1)
>
> [menu]
> exten => s,1,Background(${sounds_path}menu)
>
> exten => 0,1,Goto(menu,s,1)
> exten => 1,1,Goto(call,s,1)
> exten => 2,1,Goto(support,s,1)
>
> exten => i,1,Playback(${sounds_path}error-key)
> exten => i,2,Goto(call,s,1)
> exten => t,1,Goto(call,s,1)
> [call]
> exten => s,1,Background(${sounds_path}call)
>
> exten => 0,1,Goto(menu,s,1)
> exten => 223,1,Dial(SIP/${EXTEN},20,tr)
>
> exten => i,1,Playback(${sounds_path}error-key)
> exten => t,1,Goto(appel,s,1)
>
>
> [support]
> exten => s,1,GoToIfTime(09:00-17:00|mon-fri|*|*?s,4)
> exten => s,2,Playback(${sounds_path}no-relation-support)
> exten => s,3,Goto(menu,s,1)
> exten => s,4,Playback(${sounds_path}relation-support)
> exten => s,5,Queue(default)
> exten => t,1,Hangup()
>
>
>  2011/6/20 Warren Selby 
>
>>   On Mon, Jun 20, 2011 at 12:17 PM, salaheddine elharit <
>> salah.elharit...@gmail.com> wrote:
>>
>>> [home]
>>> exten => s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
>>> exten => s,2,Background(${sounds_path}welcome)
>>> exten => #,1,Goto(menu,s,1)
>>> exten => i,1,Playback(${sounds_path}error-key)
>>> exten => t,1,Goto(home,s,1)
>>>
>>>
>> You need to add the following to the [home] context:
>>
>> exten => s,3,WaitExten(10)
>>
>> which will cause the call to wait 10 seconds for input, otherwise it will
>> timeout and go to the 't' extension.  The way you currently have it, the
>> call will end after the Background() app finishes playing because it has no
>> additional steps and nothing that will tell it to go to the 't' extension.
>>
>> Also, consider switching your dialplan priorities away from "1,2,3..." and
>> go to "1,n,n,n..." as this reduces headaches in the longrun.
>>
>> --
>> Thanks,
>> --Warren Selby, dCAP
>> http://www.SelbyTech.com 
>>
>>
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>
>
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Re: [asterisk-users] Problem with ReceiveFAX app from FFA

[khalid touati]

@ Bryant: thanks so much for the interesting figure of use.


> Why do so may people think their problems are unique. Many people use FFA
> and spandsp. They all come across this. The issue is widely known, well
> understood, and not at all strange once you think about it.
>
> Steve
>
>
 @ Steve: don't get that mad dude, my impression is only My impression and
it only affects me, so nothing to worry about, i'd rather discuss asterisk
issues instead of discussing my impression, but thanks for your help.

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Re: [asterisk-users] Problem with ReceiveFAX app from FFA

[khalid touati]

Ok, for the variables, I can retrieve some of them like the caller number
and so on (I would assume that all the variables that last for duration of
call are there), but I still think that I sould not use the h extension to
continue after ReceiveFAX use, it's like not a lot of people use FFA,
moreover very few came accross such an issue which is fine.
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Re: [asterisk-users] Problem with ReceiveFAX app from FFA

[khalid touati]

Hi Guys,
I solved temporarely my issue by kind of tricking Asterisk, I used the
following line instead of the old:
exten => h,n,System('/usr/local/
bin/fax2mail -p -f "${FAXFILENOEXT}" --cid-number ${CALLERID(num)}
--cid-name "${CALLERID(name)}" --dest-name "Sir/Madam"')
now when it hang up I receive my fax through email, and let me tell you
(first time using Free Fax from Asterisk) ReceiveFAX catch well faxes, just
a couple tries but got them all, let's see with more faxes what will happen.


On Sun, Jun 19, 2011 at 12:24 PM, khalid touati wrote:

>
> Hi all,
> I am running to the following problem, when using the below dialplan to
> receive fax, everything works perfect till this line
> exten => receive,n,ReceiveFAX(${FAXFILE}):
> and then the following line cannot be executed, it's like asterisk can't go
> back to dialplan and continue, the good news is when i check what is
> received in my fax folder i find that the file is a valid one (not corrupted
> or empty), also when I use another way to execute fax2mail, it's working
> perfect and sent right to my email (test was done with the same file
> received by ReceiveFAX), so I belive the problem is just when transiting
> from the line that call ReceiveFAX and the line that call fax2mail (but both
> work not in order or separately), the debugging of PRI channel is below
> dialplan (maybe i will need to turn on debugging in receiveFAX app as well),
> please advise!
>
> I am using asterisk 1.6.2.11, FAX For Asterisk Components:
> Applications: 1.6.2.0_1.2.1
> Digium FAX Driver: 1.6.2.0_1.2.1 (optimized for barcelona_64)
>
> [fax-rx]
> exten => receive,1,NoOp( FAX RECEIVE )
> exten => receive,n,Set(GLOBAL(FAXCOUNT)=$[ ${GLOBAL(FAXCOUNT)} + 1 ])
> exten => receive,n,Set(FAXCOUNT=${GLOBAL(FAXCOUNT)})
> exten =>
> receive,n,Set(FAXFILE=/var/spool/asterisk/fax/${STRFTIME(,,%F_%T_${CALLERID(num)})}.tif)
> exten =>
> receive,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${STRFTIME(,,%F_%T_${CALLERID(num)})})
> exten => receive,n,Set(GLOBAL(LASTFAXCALLERNUM)=${CALLERID(num)})
> exten => receive,n,Set(GLOBAL(LASTFAXCALLERNAME)=${CALLERID(name)})
> exten => receive,n,NoOp( SETTING FAXOPT )
> exten => receive,n,Set(FAXOPT(ecm)=yes)
> exten => receive,n,Set(FAXOPT(headerinfo)=MY FAXBACK RX)
> exten => receive,n,Set(FAXOPT(localstationid)=15184893772)
> exten => receive,n,Set(FAXOPT(maxrate)=14400)
> exten => receive,n,Set(FAXOPT(minrate)=2400)
> exten => receive,n,NoOp(FAXOPT(ecm) : ${FAXOPT(ecm)})
> exten => receive,n,NoOp(FAXOPT(headerinfo) : ${FAXOPT(headerinfo)})
> exten => receive,n,NoOp(FAXOPT(localstationid) : ${FAXOPT(localstationid)})
> exten => receive,n,NoOp(FAXOPT(maxrate) : ${FAXOPT(maxrate)})
> exten => receive,n,NoOp(FAXOPT(minrate) : ${FAXOPT(minrate)})
> exten => receive,n,NoOp( RECEIVING FAX : ${FAXFILE} )
> exten => receive,n,ReceiveFAX(${FAXFILE})
> exten => receive,n,System('/usr/local/bin/fax2mail -p -f "${FAXFILENOEXT}"
> --cid-number ${CALLERID(num)} --cid-name "${CALLERID(name)}" --dest-name
> "Sir/Madam"')
>
> *debug:*
>
> pbx3*CLI>
> -- Executing [receive@fax-rx:20] ReceiveFAX("DAHDI/1-1",
> "/var/spool/asterisk/fax/2011-06-18_12:52:44_8009806858.tif") in new stack
> q931.c:5088 q931_connect: Call 14288 enters state 8 (Connect Request).
> Hold state: Idle
> pbx3*CLI>
> pbx3*CLI>
> > DL-DATA request
> pbx3*CLI>
> > Protocol Discriminator: Q.931 (8)  len=14
> pbx3*CLI>
> > TEI=0 Call Ref: len= 2 (reference 14288/0x37D0) (Sent to originator)
> pbx3*CLI>
> > Message Type: CONNECT (7)
> TEI=0 Transmitting N(S)=1, window is open V(A)=0 K=7
> pbx3*CLI>
> pbx3*CLI>
> > Protocol Discriminator: Q.931 (8)  len=14
> pbx3*CLI>
> > TEI=0 Call Ref: len= 2 (reference 14288/0x37D0) (Sent to originator)
> > Message Type: CONNECT (7)
> pbx3*CLI>
> > [18 03 a9 83 81]
> > Channel ID (len= 5) [ Ext: 1  IntID: Implicit  Other(PRI)  Spare: 0
> Exclusive  Dchan: 0
>
> pbx3*CLI>
> >   ChanSel: As indicated in following octets
> >   Ext: 1  Coding: 0  Number Specified  Channel Type:
> 3
> pbx3*CLI>
> >   Ext: 1  Channel: 1 Type: CPE]
> > [1e 02 81 82]
> pbx3*CLI>
> > Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)
> 0: 0  Location: Private network serving the local user (1)
> pbx3*CLI>
> >   Ext: 1  Progress Description: Called
> equipment is non-ISDN. (2) ]
> pbx3*CLI>
> -- 

[asterisk-users] Problem with ReceiveFAX app from FFA

[khalid touati]

Hi all,
I am running to the following problem, when using the below dialplan to
receive fax, everything works perfect till this line
exten => receive,n,ReceiveFAX(${FAXFILE}):
and then the following line cannot be executed, it's like asterisk can't go
back to dialplan and continue, the good news is when i check what is
received in my fax folder i find that the file is a valid one (not corrupted
or empty), also when I use another way to execute fax2mail, it's working
perfect and sent right to my email (test was done with the same file
received by ReceiveFAX), so I belive the problem is just when transiting
from the line that call ReceiveFAX and the line that call fax2mail (but both
work not in order or separately), the debugging of PRI channel is below
dialplan (maybe i will need to turn on debugging in receiveFAX app as well),
please advise!

I am using asterisk 1.6.2.11, FAX For Asterisk Components:
Applications: 1.6.2.0_1.2.1
Digium FAX Driver: 1.6.2.0_1.2.1 (optimized for barcelona_64)

[fax-rx]
exten => receive,1,NoOp( FAX RECEIVE )
exten => receive,n,Set(GLOBAL(FAXCOUNT)=$[ ${GLOBAL(FAXCOUNT)} + 1 ])
exten => receive,n,Set(FAXCOUNT=${GLOBAL(FAXCOUNT)})
exten =>
receive,n,Set(FAXFILE=/var/spool/asterisk/fax/${STRFTIME(,,%F_%T_${CALLERID(num)})}.tif)
exten =>
receive,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${STRFTIME(,,%F_%T_${CALLERID(num)})})
exten => receive,n,Set(GLOBAL(LASTFAXCALLERNUM)=${CALLERID(num)})
exten => receive,n,Set(GLOBAL(LASTFAXCALLERNAME)=${CALLERID(name)})
exten => receive,n,NoOp( SETTING FAXOPT )
exten => receive,n,Set(FAXOPT(ecm)=yes)
exten => receive,n,Set(FAXOPT(headerinfo)=MY FAXBACK RX)
exten => receive,n,Set(FAXOPT(localstationid)=15184893772)
exten => receive,n,Set(FAXOPT(maxrate)=14400)
exten => receive,n,Set(FAXOPT(minrate)=2400)
exten => receive,n,NoOp(FAXOPT(ecm) : ${FAXOPT(ecm)})
exten => receive,n,NoOp(FAXOPT(headerinfo) : ${FAXOPT(headerinfo)})
exten => receive,n,NoOp(FAXOPT(localstationid) : ${FAXOPT(localstationid)})
exten => receive,n,NoOp(FAXOPT(maxrate) : ${FAXOPT(maxrate)})
exten => receive,n,NoOp(FAXOPT(minrate) : ${FAXOPT(minrate)})
exten => receive,n,NoOp( RECEIVING FAX : ${FAXFILE} )
exten => receive,n,ReceiveFAX(${FAXFILE})
exten => receive,n,System('/usr/local/bin/fax2mail -p -f "${FAXFILENOEXT}"
--cid-number ${CALLERID(num)} --cid-name "${CALLERID(name)}" --dest-name
"Sir/Madam"')

*debug:*

pbx3*CLI>
-- Executing [receive@fax-rx:20] ReceiveFAX("DAHDI/1-1",
"/var/spool/asterisk/fax/2011-06-18_12:52:44_8009806858.tif") in new stack
q931.c:5088 q931_connect: Call 14288 enters state 8 (Connect Request).  Hold
state: Idle
pbx3*CLI>
pbx3*CLI>
> DL-DATA request
pbx3*CLI>
> Protocol Discriminator: Q.931 (8)  len=14
pbx3*CLI>
> TEI=0 Call Ref: len= 2 (reference 14288/0x37D0) (Sent to originator)
pbx3*CLI>
> Message Type: CONNECT (7)
TEI=0 Transmitting N(S)=1, window is open V(A)=0 K=7
pbx3*CLI>
pbx3*CLI>
> Protocol Discriminator: Q.931 (8)  len=14
pbx3*CLI>
> TEI=0 Call Ref: len= 2 (reference 14288/0x37D0) (Sent to originator)
> Message Type: CONNECT (7)
pbx3*CLI>
> [18 03 a9 83 81]
> Channel ID (len= 5) [ Ext: 1  IntID: Implicit  Other(PRI)  Spare: 0
Exclusive  Dchan: 0

pbx3*CLI>
>   ChanSel: As indicated in following octets
>   Ext: 1  Coding: 0  Number Specified  Channel Type: 3
pbx3*CLI>
>   Ext: 1  Channel: 1 Type: CPE]
> [1e 02 81 82]
pbx3*CLI>
> Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  0:
0  Location: Private network serving the local user (1)
pbx3*CLI>
>   Ext: 1  Progress Description: Called
equipment is non-ISDN. (2) ]
pbx3*CLI>
-- Channel 'DAHDI/1-1' receiving FAX
'/var/spool/asterisk/fax/2011-06-18_12:52:44_8009806858.tif'
pbx3*CLI>
-- Channel 'DAHDI/1-1' FAX session '7' started
pbx3*CLI>  < Protocol Discriminator: Q.931 (8)  len=5
< TEI=0 Call Ref: len= 2 (reference 14288/0x37D0) (Sent from originator)
< Message Type: CONNECT ACKNOWLEDGE (15)
pbx3*CLI>
Received message for call 0x2aaac80812d0 on link 0x2aaac8035bb8 TEI/SAPI 0/0
q931.c:7785 post_handle_q931_message: Call 14288 enters state 10 (Active).
Hold state: Idle
pbx3*CLI>
-- FAX handle 0: [ 104.899463 ], entering CLOSING state
-- FAX handle 0: [ 104.899528 ], entering CLOSING state
pbx3*CLI>  < Protocol Discriminator: Q.931 (8)  len=9
< TEI=0 Call Ref: len= 2 (reference 14288/0x37D0) (Sent from originator)
< Message Type: DISCONNECT (69)
< [08 02 82 90]
pbx3*CLI>
< Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0
Location: Public network serving the local user (2)
<  Ext: 1  Cause: Normal Clearing (16), class = Normal Event
(1) ]
Received message for call 0x2aaac80812d0 on link 0x2aaac8035bb8 TEI/SAPI 0/0
-- Processing IE 8 (cs0, Cause)
pbx3*CLI>
-- Found active call: 0x2aaac80812d0 cref:14288
q931.c:7994 post_handle_q931_message: Call 14288 enters state 12 (Disconnect
Indication).

Re: [asterisk-users] Can I use phone line to recive faxes?

[khalid touati]

Yeah I am using a TDM410P, thanks for the answer.

On Fri, Jun 3, 2011 at 4:30 AM, A J Stiles wrote:

>  On Thursday 02 Jun 2011, khalid touati wrote:
> > Hi Guys,
> > Actually My question is as in the subject, may I use a regular phone line
> > to receive faxes with FFA (Fax For Asterisk), I am using asterisk
> 1.6.2.8.
>
> Yes, you can.  BUT, you will need some sort of FXO interface  (allows the
> computer to connect to the telephone socket on the wall),  which is
> supported
> by DAHDI (or its predecesor, Zaptel).
>
> --
> AJS
>
> Answers come *after* questions.
>
> --
> _
> -- 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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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



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Re: [asterisk-users] SIP Register DOS attack

[khalid touati]

Also you guys may need to use:
sip.conf
[general]
allowguest=no
*alwaysauthreject = yes*



On Thu, Jun 2, 2011 at 1:01 PM, Al lists  wrote:

> I'll check this option and see if it helps next time,
> just to clarify, there were no actual calls in place, just DOS register
> attack.
>
>
>   On Wed, Jun 1, 2011 at 12:22 PM, Ira  wrote:
>
>>   At 10:56 AM 6/1/2011, you wrote:
>>
>> Do you have:
>>
>> sip.conf
>> [general]
>> allowguest=no
>>
>>
>> So because of this I decided to type "sip show channels" into my Asterisk
>> and got this:
>>
>> Peer User/ANRCall ID  Format Hold  Last
>> Message  Expiry  Peer
>> 216.xxx.69.xxx   (None)  f2d8db55-0a7edd  (nothing)  NoRx:
>> OPTIONS   
>> 216.xxx.69.xxx   (None)  2ce0b9a5-6de7f4  (nothing)  NoRx:
>> OPTIONS   
>> 64.xxx.41.xxx6314098389  2a482e4b684a59a  (nothing)
>> No  
>> 192.168.233.xxx  (None) ioh3fna2aw.n4mz  (nothing)  NoRx:
>> REGISTER  
>> 4 active SIP dialogs
>>
>> I have allowguest=no and all of those IPs are either my providers or a SIP
>> phone on my network so why would it show  as the peer?
>>
>> I'm running Asterisk SVN-trunk-r319759M  if that matters.
>>
>> Ira
>>
>> --
>> _
>> -- 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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>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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[asterisk-users] Can I use phone line to recive faxes?

[khalid touati]

Hi Guys,
Actually My question is as in the subject, may I use a regular phone line to
receive faxes with FFA (Fax For Asterisk), I am using asterisk 1.6.2.8.
-- 
Khalid
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Re: [asterisk-users] Call failed becaus of SIP tanslate

[khalid touati]

Ok, it seems like I don't have g729 codec intsalled, can I install this
codec within asterisk 1.2 or just 1.4 and 1.6 are supported?


On Fri, Nov 12, 2010 at 2:56 PM, khalid touati wrote:

> Hi Guys,
> I have a the following issue when I ma trying to place a call through my
> voip provider, I am using an asterisk 1.2.21.1, do you have an idea what
> could fix this issue (as you can see when the other party answered, the call
> get dropped because of probably sip incompatibility)
>
> Nov 12 14:31:30 WARNING[21432]: chan_sip.c:2587 sip_write: Asked to
> transmit frame type 256, while native formats is 4 (read/write = 4/4)
> Nov 12 14:31:30 WARNING[21432]: chan_sip.c:2587 sip_write: Asked to
> transmit frame type 4, while native formats is 256 (read/write = 256/256)
> Nov 12 14:31:30 WARNING[21432]: chan_sip.c:2587 sip_write: Asked to
> transmit frame type 256, while native formats is 4 (read/write = 4/4)
> Nov 12 14:31:30 WARNING[21432]: chan_sip.c:2587 sip_write: Asked to
> transmit frame type 4, while native formats is 256 (read/write = 256/256)
> Nov 12 14:31:30 WARNING[21432]: chan_sip.c:2587 sip_write: Asked to
> transmit frame type 256, while native formats is 4 (read/write = 4/4)
> -- SIP/to-my-voip-11b955c0 answered SIP/8021-52514588
> Nov 12 14:31:30 WARNING[21432]: channel.c:2781 ast_channel_make_compatible:
> No path to translate from SIP/8021-52514588(4) to
> SIP/to-my-voip-11b955c0(256)
> Nov 12 14:31:30 WARNING[21432]: app_dial.c:1628 dial_exec_full: Had to drop
> call because I couldn't make SIP/8021-52514588 compatible with
> SIP/to-my-voip-11b955c0
>
> Thank you for any help!
>
> --
> Abdullah
>



-- 
Abdullah
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[asterisk-users] Call failed becaus of SIP tanslate

[khalid touati]

Hi Guys,
I have a the following issue when I ma trying to place a call through my
voip provider, I am using an asterisk 1.2.21.1, do you have an idea what
could fix this issue (as you can see when the other party answered, the call
get dropped because of probably sip incompatibility)

Nov 12 14:31:30 WARNING[21432]: chan_sip.c:2587 sip_write: Asked to transmit
frame type 256, while native formats is 4 (read/write = 4/4)
Nov 12 14:31:30 WARNING[21432]: chan_sip.c:2587 sip_write: Asked to transmit
frame type 4, while native formats is 256 (read/write = 256/256)
Nov 12 14:31:30 WARNING[21432]: chan_sip.c:2587 sip_write: Asked to transmit
frame type 256, while native formats is 4 (read/write = 4/4)
Nov 12 14:31:30 WARNING[21432]: chan_sip.c:2587 sip_write: Asked to transmit
frame type 4, while native formats is 256 (read/write = 256/256)
Nov 12 14:31:30 WARNING[21432]: chan_sip.c:2587 sip_write: Asked to transmit
frame type 256, while native formats is 4 (read/write = 4/4)
-- SIP/to-my-voip-11b955c0 answered SIP/8021-52514588
Nov 12 14:31:30 WARNING[21432]: channel.c:2781 ast_channel_make_compatible:
No path to translate from SIP/8021-52514588(4) to
SIP/to-my-voip-11b955c0(256)
Nov 12 14:31:30 WARNING[21432]: app_dial.c:1628 dial_exec_full: Had to drop
call because I couldn't make SIP/8021-52514588 compatible with
SIP/to-my-voip-11b955c0

Thank you for any help!

-- 
Abdullah
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Re: [asterisk-users] Needed in Dallas, Texas, Network Voice Engineer and Field Service Install Technician

[khalid touati]

Yeah, his "crap" may be interesting to someone, if it's not the case for I'd
say it's simply not for me !! and i see no commercial advances here! I may
be wrong..

On Thu, Oct 21, 2010 at 11:19 AM, Fred Posner  wrote:

> On Oct 21, 2010, at 11:11 AM, Steve Howes wrote:
>
> > On 21 Oct 2010, at 15:56, JR Richardson wrote:
> >> These are full time positions in Dallas, no telecommuters please.
> >
> > A very vast majority of people on here are not in Dallas (and indeed
> probably a majority in the US). So stop filling their mailboxes with this
> crap.
> >
> > Incase you hadn't noticed "Asterisk Users Mailing List - Non-Commercial
> Discussion"
> >
> > S
>
> JR is pretty active here. No reason to be pissy about it. I'd rather have
> his "crap" than your tantrum on someone who contributes to the project.
>
> ---fred
> http://qxork.com
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



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Re: [asterisk-users] IAX2 works one direction, but not the other...

[khalid touati]

Hi Cassius,
it may be slightly offsubject but i did connect two 1.6 asterisk boxes, and
the only issue i had is these two statements missing:
calltokenoptional=209.16.236.73/255.255.255.0
requirecalltoken=no

hope it helps!

On Tue, Oct 19, 2010 at 12:54 PM, Paul Belanger <
paul.belan...@polybeacon.com> wrote:

> On Mon, Oct 18, 2010 at 7:41 PM, Cassius Smith 
> wrote:
> > Any hints for me?
>
> Server Ottawa (192.168.1.190)
>
> register => 
> Ottawa:ottawaisc...@192.168.1.196
>
> [Toronto]
> type=peer
> host=dynamic
> username=Toronto
> secret=TorontoIsFine
>
> Server Toronto (192.168.1.196)
>
> register => 
> Toronto:torontoisf...@192.168.1.190
>
> [Ottawa]
> type=peer
> host=dynamic
> username=Ottawa
> secret=OttawaIsCool
>
> --
> Paul Belanger | dCAP
> Polybeacon | Consultant
> Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
> blog.polybeacon.com
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>



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Re: [asterisk-users] SIP flood attacK

[khalid touati]

actually same thing happened to us a year ago (under asterisk 1.2) we solved
the same day discovered by putting both:

allowguest=no
alwaysauthreject = yes





On Sun, Oct 3, 2010 at 7:17 PM, Barry Miller wrote:

> On Sun, Oct 03, 2010 at 02:19:35PM -0600, Greg Saunders wrote:
> > Hello all. I was recently the victim of a SIP flood attack. I'm wondering
> > what is the best method to prevent such things in the future.
>
> In sip.conf:
>[general]
>alwaysauthreject = yes
>
> The attacking program is probably svwar.py (part of SIPVicious).  It
> will give up as soon as it realizes it can't tell the difference
> between attempting to register an invalid extension and a valid one
> (with an arbitrary password).
>
> It's the default in 1.8, but the option goes back at least to 1.4.
>
> --
> Barry
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Not able to join conference

[khalid touati]

Hi List,
After looking at the debug output, it turned out that the user (my boss :D)
is pressing the button twice (unless phone is faulty), so instead of trying
to log to conf # 3 it goes to conf # 33:

pbx2*CLI> Oct  1 08:05:39 DEBUG[23283]: chan_zap.c:3666 zt_handle_dtmfup:
DTMF digit: 3 on Zap/8-1

pbx2*CLI> Oct  1 08:05:39 DEBUG[23283]: chan_zap.c:3666 zt_handle_dtmfup:
DTMF digit: 3 on Zap/8-1

pbx2*CLI> Oct  1 08:05:40 DEBUG[23283]: chan_zap.c:3666 zt_handle_dtmfup:
DTMF digit: # on Zap/8-1

pbx2*CLI> Oct  1 08:05:40 DEBUG[23283]: app_meetme.c:1654 find_conf: *Building
dynamic conference '33'*


what i want to do now is to make the user aware that he typed the wrong
number, so i looked for an option in MeetMe that announces the conf #, but
unfortunately it's not there, do you know any way to reach my goal?

thanks for any help!
**


On Tue, Sep 21, 2010 at 8:47 AM, khalid touati wrote:

>
>
> On Tue, Sep 21, 2010 at 8:33 AM, Andrew Thomas  wrote:
>
>> I was wondering what happened if YOU put that number in. Does it put
>> everyone in to the same conference?
>>
>> That would, at least, prove that the MeetMe app was working as it should
>> (unless you've tried this already).
>>
>>
>> yes, as i said, it will place all caller in conf no 500. and it's not
>> supposed to work like that, for the meetme app, it's working fine except
>> this issue and i cannot even reproduce the issue.
>>
>>
>>
>>
>> -----Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid
>> touati
>> Sent: 20 September 2010 14:06
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] Not able to join conference
>>
>>
>> it's going to put you in conf no 500 without prompting you to enter a
>> conference number I guess, but i don't it's going to solve my issue.
>> actually I'm atill wondering is there a way to debug just Meetme app
>> output or the only way is turn the whole debug thing on?
>>
>>
>> On Mon, Sep 20, 2010 at 4:11 AM, Andrew Thomas 
>> wrote:
>>
>> What happens if you put in a 'room' number?
>>
>> Eg: exten => 8080,3,MeetMe(500|MDci)
>>
>>
>>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>>
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid
>> touati
>> Sent: 17 September 2010 14:24
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>
>> Subject: [asterisk-users] Not able to join conference
>>
>>
>> Hi All,
>> We are running to a weird problem, we're using asterisk 1.2 as a
>> production server (I'm wiling to move very soon to more recent version)
>> and our problem is when somebody try to join a conference he's told that
>> he's the only one in the conference but in fact there is some 3 or 5 or
>> whatever people in that same conference, after several tries he
>> can/cannot enter the conference and meet with the people already in,
>>
>> here is the lines corresponding to conf in the dialplan, that would be a
>>
>> big help if you guys can help diagnose the issue.
>>
>>
>> exten => 8080,1,Answer
>> exten => 8080,2,Wait,1
>> exten => 8080,3,MeetMe(|MDci)
>>
>>
>>
>>  If you have received this communication in error we would appreciate
>> you advising us either by telephone or return of e-mail. The contents
>> of this message, and any attachments, are the property of DataVox,
>> and are intended for the confidential use of the named recipient only.
>> If you are not the intended recipient, employee or agent responsible
>> for delivery of this message to the intended recipient, take note that
>> any dissemination, distribution or copying of this communication and
>> its attachments is strictly prohibited, and may be subject to civil or
>> criminal action for which you may be liable.
>> Every effort has been made to ensure that this e-mail or any attachments
>> are free from viruses. While the company has taken every reasonable
>> precaution to minimise this risk, neither company, nor the sender can
>> accept liability for any damage which you sustain as a result of
>> viruses.
>> It is recommended that you should carry out your own virus checks
>> before opening any attachments.
>>
>> Registered in England. No. 27459085.
>>
>>
>>
>> --
>>
>> ___

Re: [asterisk-users] Unable to load fax modules

[khalid touati]

Hi Guys,
Sorry Kevin for that it was not on purpose (i didn't pay attention to what
"reply" is putting as emails).
actually I feel so dump, i didn't pay attention at all when i was
downloading, but thanks a lot. i did install the right version and it's
showing up info about modules, so it's fine.

David:
i see what you mean, you're right it's there,thank you for your help!


On Thu, Sep 30, 2010 at 12:09 PM, David Backeberg wrote:

> On Thu, Sep 30, 2010 at 11:46 AM, khalid touati 
> wrote:
> > thanks for replies,
> > I am using Asterisk 1.6.2.11
> > and components res_fax-1.4_1.2.1-x86_64 and
> > res_fax_digium-1.4_1.2.1-barcelona_64.
> > (amd 64 bit machine)
> > actually I am not aware that there is version which include fax.
> > for rebuilding with manager support that would be great if you could give
> me
> > a link to know about that, cause i've never done it !
>
> If you're building from source, do a make menuconfig
>
> then you get the ability to do select/deselct on the individual
> applications.
>
> One is called app_fax
> That contains the built-in fax support.
>
> It requires that you have already pre-built SpanDSP.
>
> And also, it looks like you cannot NOT compile in manager support, so
> my original idea seems wrong.
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>



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Re: [asterisk-users] Unable to load fax modules

[khalid touati]

thanks for replies,
I am using Asterisk 1.6.2.11
and components res_fax-1.4_1.2.1-x86_64 and
res_fax_digium-1.4_1.2.1-barcelona_64.
(amd 64 bit machine)

David:
actually I am not aware that there is version which include fax.
for rebuilding with manager support that would be great if you could give me
a link to know about that, cause i've never done it !

Thank you guys!

On Thu, Sep 30, 2010 at 11:23 AM, Kevin P. Fleming wrote:

> On 09/30/2010 09:51 AM, khalid touati wrote:
> > Hi List,
> > I did follow the procedure to install Free Fax for Asterisk successfully
> > till i came accross this isssue: i can't load the fax module:
> >
> > pbx3*CLI> module load res_fax_digium.so
> > Unable to load module res_fax_digium.so
> > Command 'module load res_fax_digium.so' failed.
> > [Sep 30 10:50:12] WARNING[5427]: loader.c:429 load_dynamic_module: Error
> > loading module 'res_fax_digium.so':
> > /usr/lib/asterisk/modules/res_fax_digium.so: undefined symbol:
> manager_event
> > [Sep 30 10:50:12] WARNING[5427]: loader.c:797 load_resource: Module
> > 'res_fax_digium.so' could not be loaded.
> >
> > any help will be much appreciated!!
>
> It will be very hard to help you with the information you provided; at a
> minimum we need to know what version of Asterisk and of the FAX modules
> you tried to use.
>
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> skype: kpfleming | jabber: kflem...@digium.com
> Check us out at www.digium.com & www.asterisk.org
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
>
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



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[asterisk-users] Unable to load fax modules

[khalid touati]

Hi List,
I did follow the procedure to install Free Fax for Asterisk successfully
till i came accross this isssue: i can't load the fax module:

pbx3*CLI> module load res_fax_digium.so
Unable to load module res_fax_digium.so
Command 'module load res_fax_digium.so' failed.
[Sep 30 10:50:12] WARNING[5427]: loader.c:429 load_dynamic_module: Error
loading module 'res_fax_digium.so':
/usr/lib/asterisk/modules/res_fax_digium.so: undefined symbol: manager_event
[Sep 30 10:50:12] WARNING[5427]: loader.c:797 load_resource: Module
'res_fax_digium.so' could not be loaded.

any help will be much appreciated!!

-- 
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-- 
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[asterisk-users] Successive Dial apps give hang up within 30s!!

[khalid touati]

Hi All,
I am using an Asterisk 1.6.2.6, and when I use this part of the dialplan:
exten  => 8355,1,Dial(SIP/${EXTEN}&IAX2/${EXTEN},18,tTWwr)
exten => 8355,n,Dial(IAX2/8366,48,tTWwr)
(i made that simple to exhibit issue)

I got just 1 ring in 8366 extension before it hangup, what i noticed is the
total time spent on ringing is 30s that means if i use 12s in the first dial
i get 18s left in the second dial used (and so on)
i looked to the debug and cut the spot where it weirdly hang up:

localhost*CLI>
[Sep 29 20:02:32] DEBUG[21183]: chan_iax2.c:9726 socket_process: Packet
arrived out of order (expecting 3, got 2) (frametype = 6, subclass = 12)
localhost*CLI>
[Sep 29 20:02:32] DEBUG[21183]: chan_iax2.c:9733 socket_process: Acking
anyway
localhost*CLI>
[Sep 29 20:02:34] DEBUG[21184]: chan_iax2.c:10261 socket_process:
Immediately destroying 3477, having received hangup
localhost*CLI>
[Sep 29 20:02:34] DEBUG[5752]: channel.c:1760 ast_hangup: Hanging up channel
'IAX2/8366-8884'
[Sep 29 20:02:34] DEBUG[5752]: chan_iax2.c:4978 iax2_hangup: We're hanging
up IAX2/8366-8884 now...
-- Hungup 'IAX2/8366-8884'
[Sep 29 20:02:34] DEBUG[5752]: app_dial.c:2315 dial_exec_full: Exiting with
DIALSTATUS=CANCEL.
[Sep 29 20:02:34] DEBUG[5752]: pbx.c:4306 __ast_pbx_run: Spawn extension
(OFFICE_EXTEN,8355,2) exited non-zero on 'IAX2/USPBX2-3477'
  == Spawn extension (OFFICE_EXTEN, 8355, 2) exited non-zero on
'IAX2/USPBX2-3477'
[Sep 29 20:02:34] DEBUG[5752]: channel.c:1655 ast_softhangup_nolock:
Soft-Hanging up channel 'IAX2/USPBX2-3477'
[Sep 29 20:02:34] DEBUG[5752]: channel.c:1760 ast_hangup: Hanging up channel
'IAX2/USPBX2-3477'
[Sep 29 20:02:34] DEBUG[5752]: chan_iax2.c:4978 iax2_hangup: We're hanging
up IAX2/USPBX2-3477 now...
[Sep 29 20:02:34] DEBUG[5752]: chan_iax2.c:4995 iax2_hangup: Really
destroying IAX2/USPBX2-3477 now...
[Sep 29 20:02:34] DEBUG[5752]: chan_iax2.c:2540 sched_delay_remove: schedule
decrement of callno used for 208.79.76.130 in 60 seconds
-- Hungup 'IAX2/USPBX2-3477'

does anyone can guess what is hepening?
Thanks you!

-- 
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Re: [asterisk-users] Can't turn debug on in a 1.2 box

[khalid touati]

Thank you guys for the responses, actually that piece that was missing is
"logger reload", only when I issue this command i can see the dedug turning
on, i though "reload" is enough to reload everything but it seems like
"logger reload" is specifically needed to apply changes, Thanks a lot Steve
and Paul!

On Fri, Sep 24, 2010 at 6:54 PM, Steve Edwards wrote:

> > On Thu, Sep 23, 2010 at 10:06 AM, khalid touati 
> wrote:
>
> >> do you guys know how i can turn debug on or just know why it's not
> >> getting enabled?
>
> On Fri, 24 Sep 2010, Paul Belanger wrote:
>
> > *CLI> set debug 15
> > *CLI> reload
>
> If you change these lines in the '[logfiles]' section of logger.conf and
> enter 'logger reload' at the Asterisk CLI, you will get more than enough
> debugging info on the console and in your syslog file (probably
> /var/log/messages).
>
> console =
> debug,dtmf,error,event,notice,verbose,warning
> syslog.local0   =
> debug,dtmf,error,event,notice,verbose,warning
>
> Please remember to change them back and reload when you have identified
> your problem(s).
>
> --
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline  Fax: +1-760-731-3000
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



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[asterisk-users] Can't turn debug on in a 1.2 box

[khalid touati]

Hi Guys,
i could turn debug on in a asterisk 1.6 box (by enabling debug in
logger.conf and core set debug to > 0), but my issue is i cannot enable
debugging in a 1.2 box by doing the same 2 steps, also this is a production
server so i can't restart with debug enabled, do you guys know how i can
turn debug on or just know why it's not getting enabled?
Thanks a lot for your help!
-- 
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-- 
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Re: [asterisk-users] Not able to join conference

[khalid touati]

On Tue, Sep 21, 2010 at 8:33 AM, Andrew Thomas  wrote:

> I was wondering what happened if YOU put that number in. Does it put
> everyone in to the same conference?
>
> That would, at least, prove that the MeetMe app was working as it should
> (unless you've tried this already).
>
>
> yes, as i said, it will place all caller in conf no 500. and it's not
> supposed to work like that, for the meetme app, it's working fine except
> this issue and i cannot even reproduce the issue.
>
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid
> touati
> Sent: 20 September 2010 14:06
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Not able to join conference
>
>
> it's going to put you in conf no 500 without prompting you to enter a
> conference number I guess, but i don't it's going to solve my issue.
> actually I'm atill wondering is there a way to debug just Meetme app
> output or the only way is turn the whole debug thing on?
>
>
> On Mon, Sep 20, 2010 at 4:11 AM, Andrew Thomas 
> wrote:
>
> What happens if you put in a 'room' number?
>
> Eg: exten => 8080,3,MeetMe(500|MDci)
>
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
>
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid
> touati
> Sent: 17 September 2010 14:24
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Subject: [asterisk-users] Not able to join conference
>
>
> Hi All,
> We are running to a weird problem, we're using asterisk 1.2 as a
> production server (I'm wiling to move very soon to more recent version)
> and our problem is when somebody try to join a conference he's told that
> he's the only one in the conference but in fact there is some 3 or 5 or
> whatever people in that same conference, after several tries he
> can/cannot enter the conference and meet with the people already in,
>
> here is the lines corresponding to conf in the dialplan, that would be a
>
> big help if you guys can help diagnose the issue.
>
>
> exten => 8080,1,Answer
> exten => 8080,2,Wait,1
> exten => 8080,3,MeetMe(|MDci)
>
>
>
>  If you have received this communication in error we would appreciate
> you advising us either by telephone or return of e-mail. The contents
> of this message, and any attachments, are the property of DataVox,
> and are intended for the confidential use of the named recipient only.
> If you are not the intended recipient, employee or agent responsible
> for delivery of this message to the intended recipient, take note that
> any dissemination, distribution or copying of this communication and
> its attachments is strictly prohibited, and may be subject to civil or
> criminal action for which you may be liable.
> Every effort has been made to ensure that this e-mail or any attachments
> are free from viruses. While the company has taken every reasonable
> precaution to minimise this risk, neither company, nor the sender can
> accept liability for any damage which you sustain as a result of
> viruses.
> It is recommended that you should carry out your own virus checks
> before opening any attachments.
>
> Registered in England. No. 27459085.
>
>
>
> --
>
> _
> -- 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
>
>
>
>
> --
> Abdullah
>
> --
> _
> -- 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] Not able to join conference

[khalid touati]

it's going to put you in conf no 500 without prompting you to enter a
conference number I guess, but i don't it's going to solve my issue.
actually I'm atill wondering is there a way to debug just Meetme app output
or the only way is turn the whole debug thing on?

On Mon, Sep 20, 2010 at 4:11 AM, Andrew Thomas  wrote:

> What happens if you put in a 'room' number?
>
> Eg: exten => 8080,3,MeetMe(500|MDci)
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid
> touati
> Sent: 17 September 2010 14:24
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Not able to join conference
>
>
> Hi All,
> We are running to a weird problem, we're using asterisk 1.2 as a
> production server (I'm wiling to move very soon to more recent version)
> and our problem is when somebody try to join a conference he's told that
> he's the only one in the conference but in fact there is some 3 or 5 or
> whatever people in that same conference, after several tries he
> can/cannot enter the conference and meet with the people already in,
> here is the lines corresponding to conf in the dialplan, that would be a
> big help if you guys can help diagnose the issue.
>
> exten => 8080,1,Answer
> exten => 8080,2,Wait,1
> exten => 8080,3,MeetMe(|MDci)
>
>
>  If you have received this communication in error we would appreciate
> you advising us either by telephone or return of e-mail. The contents
> of this message, and any attachments, are the property of DataVox,
> and are intended for the confidential use of the named recipient only.
> If you are not the intended recipient, employee or agent responsible
> for delivery of this message to the intended recipient, take note that
> any dissemination, distribution or copying of this communication and
> its attachments is strictly prohibited, and may be subject to civil or
> criminal action for which you may be liable.
> Every effort has been made to ensure that this e-mail or any attachments
> are free from viruses. While the company has taken every reasonable
> precaution to minimise this risk, neither company, nor the sender can
> accept liability for any damage which you sustain as a result of viruses.
> It is recommended that you should carry out your own virus checks
> before opening any attachments.
>
> Registered in England. No. 27459085.
>
>
>
> --
> _
> -- 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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Re: [asterisk-users] Not able to join conference

[khalid touati]

Hi Guys,
Paul
you meant a debug file while the problem is happening, actually the thing is
i cannot even reproduce this issue, I'll keep trying though, but is there a
way to debug just Meetme app output?

On Fri, Sep 17, 2010 at 1:04 PM, Danny Nicholas  wrote:

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul
> Belanger
> Sent: Friday, September 17, 2010 11:57 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Not able to join conference
>
> On Fri, Sep 17, 2010 at 9:24 AM, khalid touati 
> wrote:
> > in the dialplan, that would be a big help if you guys can help diagnose
> the
> > issue.
> >
> A debug log of the actually problem will be more helpful.
>
> --
> Paul Belanger | dCAP
> Polybeacon | Consultant
> Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
> blog.polybeacon.com
>
> Or at least some CLI output, since there are only a few (hundred/thousand?)
> 1.2 users.
>
>
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[asterisk-users] Not able to join conference

[khalid touati]

Hi All,
We are running to a weird problem, we're using asterisk 1.2 as a production
server (I'm wiling to move very soon to more recent version) and our problem
is when somebody try to join a conference he's told that he's the only one
in the conference but in fact there is some 3 or 5 or whatever people in
that same conference, after several tries he can/cannot enter the conference
and meet with the people already in, here is the lines corresponding to conf
in the dialplan, that would be a big help if you guys can help diagnose the
issue.

exten => 8080,1,Answer
exten => 8080,2,Wait,1
exten => 8080,3,MeetMe(|MDci)
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Re: [asterisk-users] Soft-phone on Black Berry

[khalid touati]

thank you guys for your responses!
sorry, actually  i was not accurate in asking this question, my search is
restricted to soft-phone to use within Black Berry and integrate with
Asterisk.
It seems like the cheapest solution available now that you can integrate
with asterisk and install in Black Berries is the famous Skype (of course
after buying the connector).
.

2010/7/15 Danny Nicholas 

>  --
>
> n   GI (NOT?) YF
>
> n   Try this link
>
> n   http://voip.about.com/od/mobilevoip/a/BlackBerryVoIP.htm
>
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[asterisk-users] Soft-phone on Black Berry

[khalid touati]

Hi All,
i have a question, is there any soft-phone available for Black Berry use,
I've been told there is a firefly one, but when i looked, i found nothing,
is any body has an update on this please?
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Re: [asterisk-users] Fax for Asterisk, capable of receiving from website but not from fax machine !!

[khalid touati]

>'Fax for Asterisk' is a commercial application sold by Digium. This is not
their official support channel. Since you paid for the >product, why not
contact them directly about your problem?

i did get this version for free after buying a (actually several) digium
telephony card, but i realized that they're not supporting the free version
after talking and emailing them, actually i was calling Digium support for
all the past year and i can say that (for me) it was good:4/5 satisfaction,
but this time with fax, i didn't get much help, i was redirected to the
community and that why i posted. by the way is there a reliable alternative?
is for 1.6 rfax is doing good (if anyone worked with it)?
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Re: [asterisk-users] Fax for Asterisk, capable of receiving from website but not from fax machine !!

[khalid touati]

so nobody seems to like dealing with fax!!

2010/7/12 khalid touati 

> Hi Guys,
> i am using the latest version of asterisk 1.4 (1.4.33.1), dahdi (2.3.0.1)
> and FFA (Applications: 1.4_1.2.0, Digium FAX Driver: 1.4_1.2.0). the issue
> i'm having is that i'm able to receive faxes from a website (that offer this
> service) but not able to receive from a regular fax machine (that is working
> perfect).
>
> [fax-rx]
>
> exten => receive,1,NoOp( FAX RECEIVE ) exten =>
> receive,n,Set(GLOBAL(FAXCOUNT)=$[ ${GLOBAL(FAXCOUNT)} + 1 ]) exten =>
> receive,n,Set(FAXCOUNT=${GLOBAL(FAXCOUNT)})
>
> exten =>
> receive,n,Set(FAXFILE=${STRFTIME(${EPOCH},GMT-5,%F_%T_${CALLERIDNUM})}.tif)
>
> exten =>
> receive,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},GMT-5,%F_%T_${CALLERIDNUM})})
>
> exten => receive,n,Set(GLOBAL(LASTFAXCALLERNUM)=${CALLERID(num)})
>
> exten => receive,n,Set(GLOBAL(LASTFAXCALLERNAME)=${CALLERID(name)})
>
> exten => receive,n,NoOp( SETTING FAXOPT ) exten =>
> receive,n,Set(FAXOPT(ecm)=yes) exten => receive,n,Set(FAXOPT(headerinfo)=MY
> FAXBACK RX) exten => receive,n,Set(FAXOPT(localstationid)=15184893772)
>
> exten => receive,n,Set(FAXOPT(maxrate)=14400)
>
> exten => receive,n,Set(FAXOPT(minrate)=2400)
>
> exten => receive,n,NoOp(FAXOPT(ecm) : ${FAXOPT(ecm)}) exten =>
> receive,n,NoOp(FAXOPT(headerinfo) : ${FAXOPT(headerinfo)}) exten =>
> receive,n,NoOp(FAXOPT(localstationid) : ${FAXOPT(localstationid)}) exten =>
> receive,n,NoOp(FAXOPT(maxrate) : ${FAXOPT(maxrate)}) exten =>
> receive,n,NoOp(FAXOPT(minrate) : ${FAXOPT(minrate)}) exten =>
> receive,n,NoOp( RECEIVING FAX : ${FAXFILE} ) exten =>
> receive,n,ReceiveFAX(/var/spool/asterisk/fax/${FAXFILE})
>
> exten => receive,n,System('/usr/local/bin/fax2mail -p -f "${FAXFILENOEXT}"
> --cid-number ${CALLERID(num)} --cid-name "${CALLERID(name)}" --dest-name
> "Sir/Madam"')
>
>
>
> a previous debugging showed:
>
>
> *- for a fax from myfax.com that was received successfully:*
>
> pbx1*CLI>
>
>> Channel 'DAHDI/1-1' fax session '53', [ 034.021683 ], channel
> sent 59 frames (1180 ms) of energy.
>
> pbx1*CLI>
>
> -- Channel 'DAHDI/1-1' fax session '53', [ 040.489601 ],
> STAT_EVT_HW_CLOSE  st: WT_HW_CLSrt: WCLSNCLS
>
> pbx1*CLI>
>
> -- Channel 'DAHDI/1-1' fax session '53', [ 040.489798 ],
> STAT_SES_COMPLETE
>
> pbx1*CLI>
>
> -- Channel 'DAHDI/1-1' fax session '53' is complete, result: 'SUCCESS'
> (FAX_SUCCESS), error: 'NO_ERROR', pages: 2, resolution: '204x196', transfer
> rate: '14400', remoteSID: 'FAX'
>
> pbx1*CLI>
>
> -- Executing [rece...@fax-rx:21] System("DAHDI/1-1",
> "/usr/local/bin/fax2mail -p -f
> "/var/spool/asterisk/fax/2010-05-18_03:59:42_" --cid-number  --cid-name ""
> --dest-name "Sir/Madam"") in new stack
>
>
>
> pbx1*CLI>
>
>   == Auto fallthrough, channel 'DAHDI/1-1' status is 'UNKNOWN'
>
>
>
> pbx1*CLI>
>
> -- Hungup 'DAHDI/1-1'
>
>
>
> -* for a fax from regular machine that failed:*
>
> *
> *
>
> pbx1*CLI>
>
>> Channel 'DAHDI/1-1' fax session '54', [ 032.782251 ], channel
> sent 3 frames (60 ms) of energy.
>
>
>
> pbx1*CLI>
>
> -- Channel 0/1, span 1 got hangup request, cause 16
>
>
>
> pbx1*CLI>
>
> [May 17 19:02:41] NOTICE[1316]: res_fax.c:993 generic_fax_exec: Channel
> 'DAHDI/1-1' did not return a frame; probably hung up.
>
>
>
> pbx1*CLI>
>
> -- Channel 'DAHDI/1-1' fax session '54', [ 038.131701 ],
> STAT_EVT_HW_CLOSE  st: WT_HW_CLSrt: WCLSNCLS
>
> pbx1*CLI>
>
> -- Channel 'DAHDI/1-1' fax session '54', [ 038.131879 ],
> STAT_SES_COMPLETE
>
> pbx1*CLI>
>
> -- Channel 'DAHDI/1-1' fax session '54' is complete, result: 'SUCCESS'
> (FAX_SUCCESS), error: 'NO_ERROR', pages: 1, resolution: '204x196', transfer
> rate: '14400', remoteSID: '518 489 3772'
>
>
>
> pbx1*CLI>
>
>   == Spawn extension (fax-rx, receive, 20) exited non-zero on 'DAHDI/1-1'
>
>
>
> pbx1*CLI>
>
> -- Hungup 'DAHDI/1-1'
>
>
> I would really appreciate any help! thanks!
>
> --
> Abdullah
>



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Re: [asterisk-users] Re : Re : Re : Communication IAX2 >SIP>IAX2

[khalid touati]

Glad you found the issue, sorry for not being able to help.

2010/7/9 Paul Belanger 

> On Fri, Jul 9, 2010 at 7:38 AM, Adil Zaaraoui 
> wrote:
> > ok it works i had a problem with a syntax:
> > i had to wrire:
> > exten =>_!X.,n(external),Dial(SIP/011212664800...@pstn2,,S(20))
> >
> Correct,
>
> Dial(SIP/lo...@pstn2/011212664800450,,S(20))
>
> Is not valid syntax
>
> --
> Paul Belanger | dCAP
> Polybeacon | Consultant
> Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
> blog.polybeacon.com
>
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Re: [asterisk-users] Communication IAX2 >SIP>IAX2

[khalid touati]

2010/7/8 Kyle Kienapfel 

> On Wed, Jul 7, 2010 at 10:34 AM, Adil Zaaraoui 
> wrote:
> > Dear list.
> >
> > Is it possible to use both IAX2 and SIP protocole during a dial?
> >
> > Illustration:
> >
> > I have peer A communicate with my Asterisk using IAX2 protocole.
> > I have peer B communicate with my Asterisk using SIP protocole.
> >
> > A and B are both registred to the same Asterisk.
> >
> > So is it possible that peer A communicate with peer B and vice versa?
> >
> > if yes how can i achieve that?
> >
> > Best regards
> >
>

yes it is possible, all you have to to is use the Dial application according
to what technology your peer is using, and asterisk do the translation for
you automatically:

exten = XXX,1,Dial(IAX2/A,20,) and exten = XXX,1,Dial(SIP/B,20,)

you can also use one Dial that try to reach this extension using both
technologies.

exten = XXX,1,Dial(IAX2/A&SIP/A,,) (or if A=555 for ex: exten =
555,Dial(IAX2/${EXTEN}&SIP/${EXTEN},,)

you can find this and more on the future of telephony book.

Good luck!!

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Re: [asterisk-users] Time variables in system application

[khalid touati]

thank you Barry, you're right, it is also working.
well, happy that i have a bunch of choices that work (after wrong output).
thanks for all!

2010/6/2 Barry Miller 

> On Wed, Jun 02, 2010 at 10:26:12AM -0400, khalid touati wrote:
> > Hi Guys,
> > for people who may have the same issue:
> > i was just not using STRFTIME the right way, after consulting docs, i'm
> > using it like this:
> > exten =>
> >
> ,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},America/New_York,%F_%T)})
> >
> > instead of this:
> > exten =>
> >
> ,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},GMT-5,%F_%T)})
> > and it's displaying the right time now!!
>
> If all you want is your system's idea of the current local time, you can
> simplify it to:
>
> exten =>
> ,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${STRFTIME(,,%F_%T)})
>
> Sorry for replying so late.  I somehow missed this thread back in April.
>
> --
> Barry
>
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Re: [asterisk-users] Time variables in system application

[khalid touati]

thanks i'll keep that in mind.

2010/6/2 Roderick A. Anderson 

> khalid touati wrote:
> > Hi Guys,
> > for people who may have the same issue:
> > i was just not using STRFTIME the right way, after consulting docs, i'm
> > using it like this:
> > exten =>
> >
> ,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},America/New_York,%F_%T)})
> >
> > instead of this:
> > exten =>
> >
> ,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},GMT-5,%F_%T)})
> > and it's displaying the right time now!!
>
> You might need to put America/New_York in quotes -- 'America/New_York'
> or "America/New_York".
>
> This based on my experience with the Perl DateTime module.  Of course
> YMMV and you have it working now.  :-)
>
>
> \\||/
> Rod
> --
> >
> > 2010/4/13 Danny Nicholas mailto:da...@debsinc.com>>
> >
> > My "derailed" train of thought came from OP's mention of Centos 5.3
> > - I have
> > to do a "hwclock -s" on my 5.3 box at least daily to keep a
> > reasonable time.
> >
> > -Original Message-
> > From: asterisk-users-boun...@lists.digium.com
> > <mailto:asterisk-users-boun...@lists.digium.com>
> > [mailto:asterisk-users-boun...@lists.digium.com
> > <mailto:asterisk-users-boun...@lists.digium.com>] On Behalf Of
> Tilghman
> > Lesher
> > Sent: Tuesday, April 13, 2010 2:58 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [asterisk-users] Time variables in system application
> >
> > On Tuesday 13 April 2010 14:00:36 Danny Nicholas wrote:
> >  > Just what I thought - guess that's the X'th time I wuz wrong
> today.
> >
> > The only difference between what I think you're calling the system
> time
> > (output of date) and Asterisk is that Asterisk uses a different
> > (internal)
> > library to convert the epoch-based time into a broken-out date.  Both
> > are using exactly the same value internally, however.  Hardware clock
> is
> > generally how system time is set initially at boot, though with NTP
> > servers
> > and system skew, it's possible for the two values to drift apart
> > over time.
> >
> > --
> > Tilghman Lesher
> > Digium, Inc. | Senior Software Developer
> > twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
> > Check us out at: www.digium.com <http://www.digium.com> &
> > www.asterisk.org <http://www.asterisk.org>
> >
> > --
> > _
> > -- 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
> >
> >
> > --
> > _
> > -- 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
> >
> >
> >
> >
> > --
> > Abdullah
> >
>
>
> --
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Re: [asterisk-users] Time variables in system application

[khalid touati]

Hi Guys,
for people who may have the same issue:
i was just not using STRFTIME the right way, after consulting docs, i'm
using it like this:
exten =>
,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},America/New_York,%F_%T)})

instead of this:
exten =>
,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},GMT-5,%F_%T)})
and it's displaying the right time now!!




2010/4/13 Danny Nicholas 

> My "derailed" train of thought came from OP's mention of Centos 5.3 - I
> have
> to do a "hwclock -s" on my 5.3 box at least daily to keep a reasonable
> time.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
> Lesher
> Sent: Tuesday, April 13, 2010 2:58 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Time variables in system application
>
> On Tuesday 13 April 2010 14:00:36 Danny Nicholas wrote:
> > Just what I thought - guess that's the X'th time I wuz wrong today.
>
> The only difference between what I think you're calling the system time
> (output of date) and Asterisk is that Asterisk uses a different (internal)
> library to convert the epoch-based time into a broken-out date.  Both
> are using exactly the same value internally, however.  Hardware clock is
> generally how system time is set initially at boot, though with NTP servers
> and system skew, it's possible for the two values to drift apart over time.
>
> --
> Tilghman Lesher
> Digium, Inc. | Senior Software Developer
> twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
> Check us out at: www.digium.com & www.asterisk.org
>
> --
> _
> -- 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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[asterisk-users] Faxes from website works, but from regular don't: cause 16

[khalid touati]

Hi Guys,
I'm having a non-obvious issue, i am using Fax for asterisk to receive
faxes, so when i test using a website that send faxes it's working great:
the fax is received and the fax2mail app is called and i get it in my email
box. but when i try using a regular fax machine everything in logs (turned
on debug) but all of the sudden a line appear saying:
Channel 0/1, span 1 got hangup request, cause 16
and then the fax2mail is not called for some reason and [image: :(] no fax
received, can you help me guys with that?
thanks!!!

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Re: [asterisk-users] Code in extensions.conf to leave a voicemailin another PBX ?!

[khalid touati]

Hi Guys,
first of all, thanks Danny for your support trying to help is a big help
itself.
so the thing is:
from pbx1 to pbx2 which was able to leave VM, it was set up like this:
exten => 8021,1,Dial(IAX2/pbx2/${EXTEN},30,tTWwr)

but from pbx2 to pbx1 which was not able to leave VM, it was setup like
this:
exten => 8093,1,Dial(IAX2/pbx1/${EXTEN},20,tTWwr)

that seems to me suly, but though i wnet ahead and modified the only
difference which is the ring time from 20 to 30, and IT WORKED!!!
i wasted some time going over values, and it seems like it's working for 21
but not for 20, maybe a pro can give us precise explanation, but at least i
can leave a VM now :)!

2010/5/5 Danny Nicholas 

>  This is a little over my head, but the message indicates that you don’t
> have a fully authorized connection.  Can you post the iax.conf snippets
> relevant to the call?
>
>
>  --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *khalid touati
> *Sent:* Wednesday, May 05, 2010 8:36 AM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Code in extensions.conf to leave a
> voicemailin another PBX ?!
>
>
>
> Thank you Danny, but it says in the link that it's an iptables issue,
> though i allowed everything on this network interface and even stopped
> iptables but still i have this issue.
>
> 2010/5/4 Danny Nicholas 
>
> See if this helps
>
> http://www.voipuser.org/forum_topic_3921.html
>
>
>
>
>  --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *khalid touati
> *Sent:* Tuesday, May 04, 2010 11:35 AM
>
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>
> *Subject:* Re: [asterisk-users] Code in extensions.conf to leave a voice
> mailin another PBX ?!
>
>
>
> Hi Guys,
> so when i dial from an asterisk 1.2 to asterisk 1.4 i get the following
> warning:
> WARNING[640]: file.c:738 ast_readaudio_callback: Failed to write frame
> is anyone familiar with?
>
> 2010/4/29 khalid touati 
>
> Hi Guys,
> Danny: as i said from pbx1 (1.4) to pbx2 (1.2) it's working fine.
> Peder: i just didn't want to put a lot of lines, (by the way it's dialing
> talking fine), but here you are:
>
> [macro-stdexten]
>
> exten => s,n,Dial(SIP/${ARG1}&IAX2/${ar...@${arg1},20,tTrWw);Ring
> phone for 20 seconds
>
>
> exten => s,n,Goto(s-${DIALSTATUS},1)
>
> exten => s-NOANSWER,1,Voicemail(u${ARG1})
> exten => s-NOANSWER,2,Goto(default,s,1)
>
> exten => s-BUSY,1,Voicemail(b${ARG1})
> exten => s-BUSY,2,Goto(default,s,1)
>
> exten => _s-.,1,Goto(s-NOANSWER,1)
>
> exten => a,1,VoicemailMain(${ARG1})
>
>   2010/4/29 Peder 
>
> In PBX1, where are you actually dialing the phone?  The first line of the
> macro just says “goto dialstatus” with no Dial statement.
>
>
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *khalid touati
>
>
> *Sent:* Thursday, April 29, 2010 2:03 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>
> *Subject:* [asterisk-users] Code in extensions.conf to leave a voice mail
> in another PBX ?!
>
>
>
> Hi Guys,
> i spent some time to figure this out (since i love how dialplan is written)
> but i decided to ask for your help guys.
>
> i have two asterisk servers one is 1.2 the other is 1.4, from 1.4 (pbx1) to
> 1.2 (pbx2) i can leave a voice mail without any pb, but from pbx2 to pbx1 it
> just hang up.
>
> in pbx2 extensions.conf:
> i am using: exten => 8029,1,Dial(IAX2/pbx1/${EXTEN},20,tTWwr)
>
> in pbx1, i have:
> exten => 8029,1,Macro(stdexten,8029)
> and in stdexten macro:
>
> exten => s,n,Goto(s-${DIALSTATUS},1)
> exten => s-NOANSWER,1,Voicemail(u${ARG1})
> exten => s-NOANSWER,2,Goto(default,s,1)
>
> exten => s-BUSY,1,Voicemail(b${ARG1})
> exten => s-BUSY,2,Goto(default,s,1)
>
> exten => _s-.,1,Goto(s-NOANSWER,1)
> exten => a,1,VoicemailMain(${ARG1})
>
> when calling from 8021(pbx2) to 8029(pbx1) i get on CLI pbx1:
>
> -- Executing [...@macro-stdexten:6] Goto("IAX2/pbx2-15464", "s-NOANSWER|1")
> in new stack
> -- Goto (macro-stdexten,s-NOANSWER,1)
> -- Executing [s-noans...@macro-stdexten:1]
> VoiceMail("IAX2/pbx2-15464", "u8029") in new stack
> *[Apr 29 14:36:35] WARNING[7307]: file.c:738 ast_readaudio_callback:
> Failed to write frame*
> --  Playing
> '/var/sp

Re: [asterisk-users] Code in extensions.conf to leave a voice mailin another PBX ?!

[khalid touati]

Thank you Danny, but it says in the link that it's an iptables issue, though
i allowed everything on this network interface and even stopped iptables but
still i have this issue.

2010/5/4 Danny Nicholas 

>  See if this helps
>
> http://www.voipuser.org/forum_topic_3921.html
>
>
>
>
>  --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *khalid touati
> *Sent:* Tuesday, May 04, 2010 11:35 AM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Code in extensions.conf to leave a voice
> mailin another PBX ?!
>
>
>
> Hi Guys,
> so when i dial from an asterisk 1.2 to asterisk 1.4 i get the following
> warning:
> WARNING[640]: file.c:738 ast_readaudio_callback: Failed to write frame
> is anyone familiar with?
>
> 2010/4/29 khalid touati 
>
> Hi Guys,
> Danny: as i said from pbx1 (1.4) to pbx2 (1.2) it's working fine.
> Peder: i just didn't want to put a lot of lines, (by the way it's dialing
> talking fine), but here you are:
>
> [macro-stdexten]
>
> exten => s,n,Dial(SIP/${ARG1}&IAX2/${ar...@${arg1},20,tTrWw);Ring
> phone for 20 seconds
>
>
> exten => s,n,Goto(s-${DIALSTATUS},1)
>
> exten => s-NOANSWER,1,Voicemail(u${ARG1})
> exten => s-NOANSWER,2,Goto(default,s,1)
>
> exten => s-BUSY,1,Voicemail(b${ARG1})
> exten => s-BUSY,2,Goto(default,s,1)
>
> exten => _s-.,1,Goto(s-NOANSWER,1)
>
> exten => a,1,VoicemailMain(${ARG1})
>
>
>   2010/4/29 Peder 
>
> In PBX1, where are you actually dialing the phone?  The first line of the
> macro just says “goto dialstatus” with no Dial statement.
>
>
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *khalid touati
>
>
> *Sent:* Thursday, April 29, 2010 2:03 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>
> *Subject:* [asterisk-users] Code in extensions.conf to leave a voice mail
> in another PBX ?!
>
>
>
> Hi Guys,
> i spent some time to figure this out (since i love how dialplan is written)
> but i decided to ask for your help guys.
>
> i have two asterisk servers one is 1.2 the other is 1.4, from 1.4 (pbx1) to
> 1.2 (pbx2) i can leave a voice mail without any pb, but from pbx2 to pbx1 it
> just hang up.
>
> in pbx2 extensions.conf:
> i am using: exten => 8029,1,Dial(IAX2/pbx1/${EXTEN},20,tTWwr)
>
> in pbx1, i have:
> exten => 8029,1,Macro(stdexten,8029)
> and in stdexten macro:
>
> exten => s,n,Goto(s-${DIALSTATUS},1)
> exten => s-NOANSWER,1,Voicemail(u${ARG1})
> exten => s-NOANSWER,2,Goto(default,s,1)
>
> exten => s-BUSY,1,Voicemail(b${ARG1})
> exten => s-BUSY,2,Goto(default,s,1)
>
> exten => _s-.,1,Goto(s-NOANSWER,1)
> exten => a,1,VoicemailMain(${ARG1})
>
> when calling from 8021(pbx2) to 8029(pbx1) i get on CLI pbx1:
>
> -- Executing [...@macro-stdexten:6] Goto("IAX2/pbx2-15464", "s-NOANSWER|1")
> in new stack
> -- Goto (macro-stdexten,s-NOANSWER,1)
> -- Executing [s-noans...@macro-stdexten:1]
> VoiceMail("IAX2/pbx2-15464", "u8029") in new stack
> *[Apr 29 14:36:35] WARNING[7307]: file.c:738 ast_readaudio_callback:
> Failed to write frame*
> --  Playing
> '/var/spool/asterisk/voicemail/default/8029/unavail' (language 'en')
>   == Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on
> 'IAX2/pbx2-15464' in macro 'stdexten'
>   == Spawn extension (default, 8029, 1) exited non-zero on
> 'IAX2/pbx2-15464'
> -- Hungup 'IAX2/pbx2-15464'
>
> any other ideas how to be able to leave a voice mail from 1.2 to 1.4 or fix
> the issue I'm having, thanks a lot!
>
> --
> Abdullah
>
>
>
> --
>
>
> _
> -- 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
>
>
>
>
> --
> Abdullah
>
>
>
>
> --
> Abdullah
>
> --
> _
> -- 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
>



-- 
Abdullah
-- 
_
-- 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

Re: [asterisk-users] Code in extensions.conf to leave a voice mail in another PBX ?!

[khalid touati]

Hi Guys,
so when i dial from an asterisk 1.2 to asterisk 1.4 i get the following
warning:
WARNING[640]: file.c:738 ast_readaudio_callback: Failed to write frame
is anyone familiar with?

2010/4/29 khalid touati 

> Hi Guys,
> Danny: as i said from pbx1 (1.4) to pbx2 (1.2) it's working fine.
> Peder: i just didn't want to put a lot of lines, (by the way it's dialing
> talking fine), but here you are:
>
> [macro-stdexten]
>
> exten => s,n,Dial(SIP/${ARG1}&IAX2/${ar...@${arg1},20,tTrWw);Ring
> phone for 20 seconds
>
> exten => s,n,Goto(s-${DIALSTATUS},1)
>
> exten => s-NOANSWER,1,Voicemail(u${ARG1})
> exten => s-NOANSWER,2,Goto(default,s,1)
>
> exten => s-BUSY,1,Voicemail(b${ARG1})
> exten => s-BUSY,2,Goto(default,s,1)
>
> exten => _s-.,1,Goto(s-NOANSWER,1)
>
> exten => a,1,VoicemailMain(${ARG1})
>
>
>
> 2010/4/29 Peder 
>
>>  In PBX1, where are you actually dialing the phone?  The first line of
>> the macro just says “goto dialstatus” with no Dial statement.
>>
>>
>>
>>
>>
>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *khalid touati
>>
>> *Sent:* Thursday, April 29, 2010 2:03 PM
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* [asterisk-users] Code in extensions.conf to leave a voice mail
>> in another PBX ?!
>>
>>
>>
>> Hi Guys,
>> i spent some time to figure this out (since i love how dialplan is
>> written) but i decided to ask for your help guys.
>>
>> i have two asterisk servers one is 1.2 the other is 1.4, from 1.4 (pbx1)
>> to 1.2 (pbx2) i can leave a voice mail without any pb, but from pbx2 to pbx1
>> it just hang up.
>>
>> in pbx2 extensions.conf:
>> i am using: exten => 8029,1,Dial(IAX2/pbx1/${EXTEN},20,tTWwr)
>>
>> in pbx1, i have:
>> exten => 8029,1,Macro(stdexten,8029)
>> and in stdexten macro:
>>
>> exten => s,n,Goto(s-${DIALSTATUS},1)
>> exten => s-NOANSWER,1,Voicemail(u${ARG1})
>> exten => s-NOANSWER,2,Goto(default,s,1)
>>
>> exten => s-BUSY,1,Voicemail(b${ARG1})
>> exten => s-BUSY,2,Goto(default,s,1)
>>
>> exten => _s-.,1,Goto(s-NOANSWER,1)
>> exten => a,1,VoicemailMain(${ARG1})
>>
>> when calling from 8021(pbx2) to 8029(pbx1) i get on CLI pbx1:
>>
>> -- Executing [...@macro-stdexten:6] Goto("IAX2/pbx2-15464", "s-NOANSWER|1")
>> in new stack
>> -- Goto (macro-stdexten,s-NOANSWER,1)
>> -- Executing [s-noans...@macro-stdexten:1]
>> VoiceMail("IAX2/pbx2-15464", "u8029") in new stack
>> *[Apr 29 14:36:35] WARNING[7307]: file.c:738 ast_readaudio_callback:
>> Failed to write frame*
>> --  Playing
>> '/var/spool/asterisk/voicemail/default/8029/unavail' (language 'en')
>>   == Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on
>> 'IAX2/pbx2-15464' in macro 'stdexten'
>>   == Spawn extension (default, 8029, 1) exited non-zero on
>> 'IAX2/pbx2-15464'
>> -- Hungup 'IAX2/pbx2-15464'
>>
>> any other ideas how to be able to leave a voice mail from 1.2 to 1.4 or
>> fix the issue I'm having, thanks a lot!
>>
>> --
>> Abdullah
>>
>> --
>>
>> _
>> -- 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
>>
>
>
>
> --
> Abdullah
>



-- 
Abdullah
-- 
_
-- 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

Re: [asterisk-users] Code in extensions.conf to leave a voice mail in another PBX ?!

[khalid touati]

Hi Guys,
Danny: as i said from pbx1 (1.4) to pbx2 (1.2) it's working fine.
Peder: i just didn't want to put a lot of lines, (by the way it's dialing
talking fine), but here you are:

[macro-stdexten]

exten => s,n,Dial(SIP/${ARG1}&IAX2/${ar...@${arg1},20,tTrWw);Ring phone
for 20 seconds
exten => s,n,Goto(s-${DIALSTATUS},1)

exten => s-NOANSWER,1,Voicemail(u${ARG1})
exten => s-NOANSWER,2,Goto(default,s,1)

exten => s-BUSY,1,Voicemail(b${ARG1})
exten => s-BUSY,2,Goto(default,s,1)

exten => _s-.,1,Goto(s-NOANSWER,1)

exten => a,1,VoicemailMain(${ARG1})



2010/4/29 Peder 

>  In PBX1, where are you actually dialing the phone?  The first line of the
> macro just says “goto dialstatus” with no Dial statement.
>
>
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *khalid touati
>
> *Sent:* Thursday, April 29, 2010 2:03 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Code in extensions.conf to leave a voice mail
> in another PBX ?!
>
>
>
> Hi Guys,
> i spent some time to figure this out (since i love how dialplan is written)
> but i decided to ask for your help guys.
>
> i have two asterisk servers one is 1.2 the other is 1.4, from 1.4 (pbx1) to
> 1.2 (pbx2) i can leave a voice mail without any pb, but from pbx2 to pbx1 it
> just hang up.
>
> in pbx2 extensions.conf:
> i am using: exten => 8029,1,Dial(IAX2/pbx1/${EXTEN},20,tTWwr)
>
> in pbx1, i have:
> exten => 8029,1,Macro(stdexten,8029)
> and in stdexten macro:
>
> exten => s,n,Goto(s-${DIALSTATUS},1)
> exten => s-NOANSWER,1,Voicemail(u${ARG1})
> exten => s-NOANSWER,2,Goto(default,s,1)
>
> exten => s-BUSY,1,Voicemail(b${ARG1})
> exten => s-BUSY,2,Goto(default,s,1)
>
> exten => _s-.,1,Goto(s-NOANSWER,1)
> exten => a,1,VoicemailMain(${ARG1})
>
> when calling from 8021(pbx2) to 8029(pbx1) i get on CLI pbx1:
>
> -- Executing [...@macro-stdexten:6] Goto("IAX2/pbx2-15464", "s-NOANSWER|1")
> in new stack
> -- Goto (macro-stdexten,s-NOANSWER,1)
> -- Executing [s-noans...@macro-stdexten:1]
> VoiceMail("IAX2/pbx2-15464", "u8029") in new stack
> *[Apr 29 14:36:35] WARNING[7307]: file.c:738 ast_readaudio_callback:
> Failed to write frame*
> --  Playing
> '/var/spool/asterisk/voicemail/default/8029/unavail' (language 'en')
>   == Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on
> 'IAX2/pbx2-15464' in macro 'stdexten'
>   == Spawn extension (default, 8029, 1) exited non-zero on
> 'IAX2/pbx2-15464'
> -- Hungup 'IAX2/pbx2-15464'
>
> any other ideas how to be able to leave a voice mail from 1.2 to 1.4 or fix
> the issue I'm having, thanks a lot!
>
> --
> Abdullah
>
> --
> _
> -- 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
>



-- 
Abdullah
-- 
_
-- 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

[asterisk-users] Code in extensions.conf to leave a voice mail in another PBX ?!

[khalid touati]

Hi Guys,
i spent some time to figure this out (since i love how dialplan is written)
but i decided to ask for your help guys.

i have two asterisk servers one is 1.2 the other is 1.4, from 1.4 (pbx1) to
1.2 (pbx2) i can leave a voice mail without any pb, but from pbx2 to pbx1 it
just hang up.

in pbx2 extensions.conf:
i am using: exten => 8029,1,Dial(IAX2/pbx1/${EXTEN},20,tTWwr)

in pbx1, i have:
exten => 8029,1,Macro(stdexten,8029)
and in stdexten macro:

exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Voicemail(u${ARG1})
exten => s-NOANSWER,2,Goto(default,s,1)

exten => s-BUSY,1,Voicemail(b${ARG1})
exten => s-BUSY,2,Goto(default,s,1)

exten => _s-.,1,Goto(s-NOANSWER,1)
exten => a,1,VoicemailMain(${ARG1})

when calling from 8021(pbx2) to 8029(pbx1) i get on CLI pbx1:

-- Executing [...@macro-stdexten:6] Goto("IAX2/pbx2-15464", "s-NOANSWER|1") in
new stack
-- Goto (macro-stdexten,s-NOANSWER,1)
-- Executing [s-noans...@macro-stdexten:1] VoiceMail("IAX2/pbx2-15464",
"u8029") in new stack
*[Apr 29 14:36:35] WARNING[7307]: file.c:738 ast_readaudio_callback: Failed
to write frame*
--  Playing
'/var/spool/asterisk/voicemail/default/8029/unavail' (language 'en')
  == Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on
'IAX2/pbx2-15464' in macro 'stdexten'
  == Spawn extension (default, 8029, 1) exited non-zero on 'IAX2/pbx2-15464'
-- Hungup 'IAX2/pbx2-15464'

any other ideas how to be able to leave a voice mail from 1.2 to 1.4 or fix
the issue I'm having, thanks a lot!

-- 
Abdullah
-- 
_
-- 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

Re: [asterisk-users] Interpbx connection

[khalid touati]

Steve,
You're completely right!! it seems like my colleague gave me a wrong info
(probably a firewall issue), i was also curious (before i read your
response) so i tried this in my network and really it has nothing to do with
call setup or peer authentication, sorry for the wrong info Guys!

2010/4/19 Steve Edwards 

> Un-top-posting...
>
> > 2010/4/14 khalid touati 
>
> >   i've connecting two pbx server successfully for several times using
> the following config:
> >
> >   register => USPBX:myp...@122.11.176.35
> >
> >   [PBX1]
> >   type=friend
> >   host=122.11.176.35
> >   trunk=yes
> >   sercret=mypass
> >   context=external
> >   deny=0.0.0.0/0.0.0.0
> >   permit=122.11.176.35/255.255.255.240
> >   insecure=very
> >   allow=all
> >   nat=yes
> >   qualify=yes
> >   canreinvite=no
> >
> >   in the other and it's the analog.
> >
> >   but now i can only dial from one end, and the other en d is giving
> me this error.
> >
> >   Apr 14 16:44:21 ERROR[26502]: chan_sip.c:6659 register_verify: Peer
> 'PBX1' is trying to register, but not
> >   configured as host=dynamic
> >
> >   when dialing a fast busy signal and it sauys in the CLI:
> CONGESTION. any help please!!!
> >
> >   --
> >   Abdullah
>
> On Mon, 19 Apr 2010, khalid touati wrote:
>
> > for people's future references: we found out that the option in DIAL
> > application in the extensions.conf has to be the same from both side,
> > the issue was India server was using "tr" while US server was using
> > "TWw" so we made them both using "tr" and that solved the issue, i guess
> > if one side is set to "trTWw" that would work regardless of the other
> > side but didn't try though. have a headeache-free experience with
> > asterisk "the future of telephony" :)!
>
> Dial() options don't have any relationship to registration failures --
> they happen at different times.
>
> Registration failures may cause dial() failures.
>
> I don't understand the relationship between ringing, transfer and
> recording options and dial() returning congestion. I'd suggest
> investigating exactly which combination is causing congestion before
> concluding it is unrelated to the registration failure.
>
> --
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline  Fax: +1-760-731-3000
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
>
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



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Re: [asterisk-users] Interpbx connection

[khalid touati]

Hi Guys,
for people's future references: we found out that the option in DIAL
application in the extensions.conf has to be the same from both side, the
issue was India server was using "tr" while US server was using "TWw" so we
made them both using "tr" and that solved the issue, i guess if one side is
set to "trTWw" that would work regardless of the other side but didn't try
though. have a headeache-free experience with asterisk "the future of
telephony" :)!

2010/4/14 khalid touati 

> Hi Guys,
> i've connecting two pbx server successfully for several times using the
> following config:
>
> register => USPBX:myp...@122.11.176.35 
>
> [PBX1]
> type=friend
> host=122.11.176.35
> trunk=yes
> sercret=mypass
> context=external
> deny=0.0.0.0/0.0.0.0
> permit=122.11.176.35/255.255.255.240
> insecure=very
> allow=all
> nat=yes
> qualify=yes
> canreinvite=no
>
> in the other and it's the analog.
>
> but now i can only dial from one end, and the other en d is giving me this
> error.
>
> Apr 14 16:44:21 ERROR[26502]: chan_sip.c:6659 register_verify: Peer 'PBX1'
> is trying to register, but not configured as host=dynamic
>
> when dialing a fast busy signal and it sauys in the CLI: CONGESTION. any
> help please!!!
>
> --
> Abdullah
>



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[asterisk-users] Interpbx connection

[khalid touati]

Hi Guys,
i've connecting two pbx server successfully for several times using the
following config:

register => USPBX:myp...@122.11.176.35 

[PBX1]
type=friend
host=122.11.176.35
trunk=yes
sercret=mypass
context=external
deny=0.0.0.0/0.0.0.0
permit=122.11.176.35/255.255.255.240
insecure=very
allow=all
nat=yes
qualify=yes
canreinvite=no

in the other and it's the analog.

but now i can only dial from one end, and the other en d is giving me this
error.

Apr 14 16:44:21 ERROR[26502]: chan_sip.c:6659 register_verify: Peer 'PBX1'
is trying to register, but not configured as host=dynamic

when dialing a fast busy signal and it sauys in the CLI: CONGESTION. any
help please!!!

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Re: [asterisk-users] Time variables in system application

[khalid touati]

i believe not only today :D, but thank u anyway for the spirit of helping
people!!

2010/4/13 Danny Nicholas 

> Just what I thought - guess that's the X'th time I wuz wrong today.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
> Lesher
> Sent: Tuesday, April 13, 2010 1:58 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Time variables in system application
>
> On Tuesday 13 April 2010 13:27:30 Danny Nicholas wrote:
> > You are apparently in U.S. Central Time zone.Asterisk uses the
> hardware
> > clock
>
> What makes you think Asterisk uses the hardware clock?
>
> --
> Tilghman Lesher
> Digium, Inc. | Senior Software Developer
> twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
> Check us out at: www.digium.com & www.asterisk.org
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>
>
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>



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Re: [asterisk-users] iptables miss up phone calls if not used properly

[khalid touati]

DNS!! i believe it has to do with call setup and rtp protocol cause all
devices shows as sip peers at the call time, but not 100% sure. any iptables
plz :) !

2010/4/13 Gordon Henderson

>

> On Tue, 13 Apr 2010, khalid touati wrote:
>
> > Hi Guys,
> > i wanted to share this with u and ask for little help at the same time:
> > i used iptables to secure my server, so i wnet ahead and blocked avery
> thing
> > except a couple of domain protocols and UDP ports of SIP, IAX2 and that
> > range 15000 to 2, tested it and OK. when in production, the calls
> were
> > taking a huge time 7s to be established and somtimes after call setup
> people
> > cannot hear ech other (but not all the time which weird), so iptables can
> > miss up performance if not set correctly (even if it's working, stuff
> like
> > this can happen). so if any body have some lines of iptables that secure
> > server and don't cause performence trouble to phone calls please share
> with
> > me (i am using Centos 5.3 asterisk 1.4.24).
>
> You've probably blocks too much and it's stopping DNS working properly.
>
> Gordon
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Time variables in system application

[khalid touati]

>You are apparently in U.S. Central Time zone.Asterisk uses the hardware
clock and >system() uses the system clock, so these are probably out of
sync.  Try doing

>Date and

>Hwclock

>From a command prompt.
thanks, here is the output of the two clocks you mentioned they dispaly same
info (slight diff on in 24 and other 12 format)!! if any body know what's
the issue, i will be grateful!

[r...@pbx1 bin]# hwclock
Tue 13 Apr 2010 02:40:16 PM EDT  -0.000607 seconds
[r...@pbx1 bin]# date
Tue Apr 13 14:41:11 EDT 2010



>
>
>  --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *khalid touati
> *Sent:* Tuesday, April 13, 2010 1:08 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Time variables in system application
>
>
>
> Hi Guys,
> i have a weird thing here: when using time variables (%F & %T) in a shell
> script, out of dial plan (particularly system() app); it displays the right
> time (same as output of date), but when same variables are used in system()
> application it displays a wrong time/date (ahead of 6 hours). I am using a
> centos 5.3, can anyone help me fix this?
>
> --
> Abdullah
>
> --
> _
> -- 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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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



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[asterisk-users] iptables miss up phone calls if not used properly

[khalid touati]

Hi Guys,
i wanted to share this with u and ask for little help at the same time:
i used iptables to secure my server, so i wnet ahead and blocked avery thing
except a couple of domain protocols and UDP ports of SIP, IAX2 and that
range 15000 to 2, tested it and OK. when in production, the calls were
taking a huge time 7s to be established and somtimes after call setup people
cannot hear ech other (but not all the time which weird), so iptables can
miss up performance if not set correctly (even if it's working, stuff like
this can happen). so if any body have some lines of iptables that secure
server and don't cause performence trouble to phone calls please share with
me (i am using Centos 5.3 asterisk 1.4.24).
Thanks!

-- 
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-- 
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[asterisk-users] Time variables in system application

[khalid touati]

Hi Guys,
i have a weird thing here: when using time variables (%F & %T) in a shell
script, out of dial plan (particularly system() app); it displays the right
time (same as output of date), but when same variables are used in system()
application it displays a wrong time/date (ahead of 6 hours). I am using a
centos 5.3, can anyone help me fix this?

-- 
Abdullah
-- 
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Re: [asterisk-users] How set debug file for RxFax application

[khalid touati]

any clue Guys???!!!

2010/4/5 khalid touati 

> Hi Juan,
> my system is an asterisk 1.2 on gentoo, it is configured to receive faxes
> through rxfax and then to use fax2email to convert the tiff to pdf and send
> it to front desk:
>
> exten =>
> 3772,1,Set(FAXFILE=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},GMT-5,%F_%T_${CALLERIDNUM}.tif)})
> exten =>
> 3772,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},GMT-5,%F_%T_${CALLERIDNUM})})
>
> exten => 3772,n,rxfax(${FAXFILE}|debug)
> exten => 3772,n,System('/usr/local/bin/fax2mail -p -f "${FAXFILENOEXT}"
> --cid-number ${CALLERIDNUM} --cid-name "${CALLERIDNAME}" --dest-name
> "Sir/Madam"')
>
> fax does not fail all the time, when the fax is failing it says in the fax
> log :
> Fax file /var/spool/asterisk/fax/2010-04-05_10:48:20_3103789385.tif not
> found.
> that means no tif was generated.
> i went ahead and turned debugging on span (ISDN pri line), i did compare
> the debug info of a failed fax and a successful fax and they look exactly
> the same. that's why i am looking to debug the rxfax or any other party that
> get involved and i forget to check.
>
> 2010/4/5 "Juan E. Rodríguez" 
>
>  How is your system configured?
>> Debug output of faild faxes?
>>
>> This kind of information is needed to help you!
>>
>> Regards,
>> Juan
>>
>>
>> khalid touati wrote:
>>
>> can anyone help me out in this, a big number of my faxes are lost
>> everyday! i would really appreciate any help on how i can tweak asterisk
>> (rxfax) to receive all faxes!
>>
>> 2010/4/2 khalid touati 
>>
>>> i went ahead and i used this line:exten => 3772,n,rxfax(${FAXFILE}|debug)
>>> as it says in the rxfax tutorial (also because i'm not sure that FAXOPT is
>>> supported by asterisk 1.2), but no output neither in the CLI or in a file.
>>> so is there any body who knows about that?
>>> i'll appreciate any help!
>>>
>>> 2010/4/2 khalid touati 
>>>
>>> thank you guys for responses,
>>>> Danny-, am i going to receive debug info in the CLI or in a default file
>>>> (/var/log /*)? is FAXOPT supported whitin asterisk 1.2 (sorry I forgot to
>>>> mention that i am using 1.2)?
>>>>
>>>>  2010/4/2 Tzafrir Cohen 
>>>>
>>>>>  On Fri, Apr 02, 2010 at 10:11:59AM -0400, khalid touati wrote:
>>>>>
>>>>> > Hi Guys,
>>>>> > do any body know how to receive debug info on RxFAX application? i am
>>>>> > experiencing a lot of fax failures and can't guess the reason behind.
>>>>> > Thank you very much for any help!
>>>>>
>>>>>They have a hard-wired log file. Make sure Asterisk can write to
>>>>> 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
>>>>>
>>>>> --
>>>>> _
>>>>> -- 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
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Abdullah
>>>>
>>>
>>>
>>>
>>> --
>>> Abdullah
>>>
>>
>>
>>
>> --
>> Abdullah
>>
>>
>> --
>> _
>> -- 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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>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> Abdullah
>



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Re: [asterisk-users] Continuous bothering message -- Remote UNIXconnection disconnected

[khalid touati]

you did  it Bradley! that was two instances running (i don't know when and
how), i gotta confess i'm not that celever to check basic things first.
Zakaria thank you for your help, i'm not using Munin or any cron job when i
looked for sshd sessions i could find just the ones i am using, so that was
it two instances.
Thanks to all of you guys, that was a ggod help!

2010/4/5 Watkins, Bradley 

>
>
>  --
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *khalid touati
> *Sent:* Monday, April 05, 2010 10:29 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Continuous bothering message -- Remote
> UNIXconnection disconnected
>
>  Hi Guys,
> i have a small issue but bothering me, after restarting asterisk (version
> 1.4 running on centos) i have the following message that comes repeatedly
> when i am connected to the CLI:
> -- Remote UNIX connection
> -- Remote UNIX connection disconnected
> -- Remote UNIX connection
> -- Remote UNIX connection disconnected
>
> does any one know how to stop this or if it's a sign of a more serious
> issue?
> i would appreciate any help, thanks!
>
>
>
> I would make sure that you don't have an extra copy of safe_asterisk
> running.
>
> - Brad
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
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Re: [asterisk-users] How set debug file for RxFax application

[khalid touati]

Hi Juan,
my system is an asterisk 1.2 on gentoo, it is configured to receive faxes
through rxfax and then to use fax2email to convert the tiff to pdf and send
it to front desk:

exten =>
3772,1,Set(FAXFILE=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},GMT-5,%F_%T_${CALLERIDNUM}.tif)})
exten =>
3772,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},GMT-5,%F_%T_${CALLERIDNUM})})
exten => 3772,n,rxfax(${FAXFILE}|debug)
exten => 3772,n,System('/usr/local/bin/fax2mail -p -f "${FAXFILENOEXT}"
--cid-number ${CALLERIDNUM} --cid-name "${CALLERIDNAME}" --dest-name
"Sir/Madam"')

fax does not fail all the time, when the fax is failing it says in the fax
log :
Fax file /var/spool/asterisk/fax/2010-04-05_10:48:20_3103789385.tif not
found.
that means no tif was generated.
i went ahead and turned debugging on span (ISDN pri line), i did compare the
debug info of a failed fax and a successful fax and they look exactly the
same. that's why i am looking to debug the rxfax or any other party that get
involved and i forget to check.

2010/4/5 "Juan E. Rodríguez" 

>  How is your system configured?
> Debug output of faild faxes?
>
> This kind of information is needed to help you!
>
> Regards,
> Juan
>
>
> khalid touati wrote:
>
> can anyone help me out in this, a big number of my faxes are lost everyday!
> i would really appreciate any help on how i can tweak asterisk (rxfax) to
> receive all faxes!
>
> 2010/4/2 khalid touati 
>
>> i went ahead and i used this line:exten => 3772,n,rxfax(${FAXFILE}|debug)
>> as it says in the rxfax tutorial (also because i'm not sure that FAXOPT is
>> supported by asterisk 1.2), but no output neither in the CLI or in a file.
>> so is there any body who knows about that?
>> i'll appreciate any help!
>>
>> 2010/4/2 khalid touati 
>>
>> thank you guys for responses,
>>> Danny-, am i going to receive debug info in the CLI or in a default file
>>> (/var/log /*)? is FAXOPT supported whitin asterisk 1.2 (sorry I forgot to
>>> mention that i am using 1.2)?
>>>
>>>  2010/4/2 Tzafrir Cohen 
>>>
>>>>  On Fri, Apr 02, 2010 at 10:11:59AM -0400, khalid touati wrote:
>>>>
>>>> > Hi Guys,
>>>> > do any body know how to receive debug info on RxFAX application? i am
>>>> > experiencing a lot of fax failures and can't guess the reason behind.
>>>> > Thank you very much for any help!
>>>>
>>>>They have a hard-wired log file. Make sure Asterisk can write to 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
>>>>
>>>> --
>>>> _
>>>> -- 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
>>>>
>>>
>>>
>>>
>>> --
>>> Abdullah
>>>
>>
>>
>>
>> --
>> Abdullah
>>
>
>
>
> --
> Abdullah
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
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Re: [asterisk-users] Continuous bothering message -- Remote UNIX connection disconnected

[khalid touati]

Thank you so much Zakaria for this valuable infos, i am not using freePBX
but i do think that happened because i did restart asterisk while it was
connected to CLI using another session of putty. but now i have just one
session, so you think maybe if  i try to find and kill that sshd process,
that will solve the issue!?

2010/4/5 Zeeshan Zakaria 

> Setting verbosity to 0 doesn't make it go away, just stops displaying it on
> the CLI, and so does it stops displaying a lot of other useful information
> which you might actually need to see in your CLI. So first try to figure out
> which computer on your network is running "asterisk -rx" command, and stop
> it there. If you are using FreePBX on one of the computers, it is very
> common that somebody leaves it in its home screen, resulting it to keep
> sending "asterisk -rx" command to the asterisk server. Solution in this case
> is to navigate away from the home screen of FreePBX. Verbose should stay at
> 3 if you want to see what is happening on your asterisk server.
>
> Zeeshan A Zakaria
>
> --
> Sent from my Android phone with K-9 Mail.
>
> On 2010-04-05 11:09 AM, "khalid touati"  wrote:
>
> Thank you Danny, i did set verbosity to 1 and it's gone, thanks to all of
> you guys!!
>
> 2010/4/5 David Gibbons 
>
>> >
>> > You probably have a cron job running that executes ‘asterisk –rx’
>> >
>> >
>> >
>> > -Dave
>> >
>> >
>> >
>> > From...
>>
>> > --
>> > _
>> > -- Bandwidth and Colo...
>>
>
>
>
> --
> Abdullah
>
> --
> _
> -- 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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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> --
> _
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Re: [asterisk-users] How set debug file for RxFax application

[khalid touati]

can anyone help me out in this, a big number of my faxes are lost everyday!
i would really appreciate any help on how i can tweak asterisk (rxfax) to
receive all faxes!

2010/4/2 khalid touati 

> i went ahead and i used this line:exten => 3772,n,rxfax(${FAXFILE}|debug)
> as it says in the rxfax tutorial (also because i'm not sure that FAXOPT is
> supported by asterisk 1.2), but no output neither in the CLI or in a file.
> so is there any body who knows about that?
> i'll appreciate any help!
>
> 2010/4/2 khalid touati 
>
> thank you guys for responses,
>> Danny-, am i going to receive debug info in the CLI or in a default file
>> (/var/log /*)? is FAXOPT supported whitin asterisk 1.2 (sorry I forgot to
>> mention that i am using 1.2)?
>>
>>  2010/4/2 Tzafrir Cohen 
>>
>>> On Fri, Apr 02, 2010 at 10:11:59AM -0400, khalid touati wrote:
>>>
>>> > Hi Guys,
>>> > do any body know how to receive debug info on RxFAX application? i am
>>> > experiencing a lot of fax failures and can't guess the reason behind.
>>> > Thank you very much for any help!
>>>
>>> They have a hard-wired log file. Make sure Asterisk can write to 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
>>>
>>> --
>>> _
>>> -- 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
>>>
>>
>>
>>
>> --
>> Abdullah
>>
>
>
>
> --
> Abdullah
>



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Re: [asterisk-users] Continuous bothering message -- Remote UNIX connection disconnected

[khalid touati]

Thank you Danny, i did set verbosity to 1 and it's gone, thanks to all of
you guys!!

2010/4/5 David Gibbons 

>  You probably have a cron job running that executes ‘asterisk –rx’
>
>
>
> -Dave
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *khalid touati
> *Sent:* Monday, April 05, 2010 10:29 AM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Continuous bothering message -- Remote UNIX
> connection disconnected
>
>
>
> Hi Guys,
>
> i have a small issue but bothering me, after restarting asterisk (version
> 1.4 running on centos) i have the following message that comes repeatedly
> when i am connected to the CLI:
> -- Remote UNIX connection
> -- Remote UNIX connection disconnected
> -- Remote UNIX connection
> -- Remote UNIX connection disconnected
>
> does any one know how to stop this or if it's a sign of a more serious
> issue?
> i would appreciate any help, thanks!
>
>
> --
> Abdullah
>
> --
> _
> -- 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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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



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[asterisk-users] Continuous bothering message -- Remote UNIX connection disconnected

[khalid touati]

Hi Guys,
i have a small issue but bothering me, after restarting asterisk (version
1.4 running on centos) i have the following message that comes repeatedly
when i am connected to the CLI:
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected

does any one know how to stop this or if it's a sign of a more serious
issue?
i would appreciate any help, thanks!


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Re: [asterisk-users] How set debug file for RxFax application

[khalid touati]

i went ahead and i used this line:exten => 3772,n,rxfax(${FAXFILE}|debug) as
it says in the rxfax tutorial (also because i'm not sure that FAXOPT is
supported by asterisk 1.2), but no output neither in the CLI or in a file.
so is there any body who knows about that?
i'll appreciate any help!

2010/4/2 khalid touati 

> thank you guys for responses,
> Danny-, am i going to receive debug info in the CLI or in a default file
> (/var/log /*)? is FAXOPT supported whitin asterisk 1.2 (sorry I forgot to
> mention that i am using 1.2)?
>
> 2010/4/2 Tzafrir Cohen 
>
>> On Fri, Apr 02, 2010 at 10:11:59AM -0400, khalid touati wrote:
>>
>> > Hi Guys,
>> > do any body know how to receive debug info on RxFAX application? i am
>> > experiencing a lot of fax failures and can't guess the reason behind.
>> > Thank you very much for any help!
>>
>> They have a hard-wired log file. Make sure Asterisk can write to 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
>>
>> --
>> _
>> -- 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
>>
>
>
>
> --
> Abdullah
>



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Re: [asterisk-users] How set debug file for RxFax application

[khalid touati]

thank you guys for responses,
Danny-, am i going to receive debug info in the CLI or in a default file
(/var/log /*)? is FAXOPT supported whitin asterisk 1.2 (sorry I forgot to
mention that i am using 1.2)?

2010/4/2 Tzafrir Cohen 

> On Fri, Apr 02, 2010 at 10:11:59AM -0400, khalid touati wrote:
> > Hi Guys,
> > do any body know how to receive debug info on RxFAX application? i am
> > experiencing a lot of fax failures and can't guess the reason behind.
> > Thank you very much for any help!
>
> They have a hard-wired log file. Make sure Asterisk can write to 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
>
> --
> _
> -- 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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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



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[asterisk-users] How set debug file for RxFax application

[khalid touati]

Hi Guys,
do any body know how to receive debug info on RxFAX application? i am
experiencing a lot of fax failures and can't guess the reason behind.
Thank you very much for any help!

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Re: [asterisk-users] Background noise

[khalid touati]

:) all users are having the same issue, even those connected to this server
from abroad!

2010/3/29 Philipp von Klitzing 

> > i have the same model polycom phone configured with another server
> > (asterisk 1.4), and guess what no noise at all. any guess!
>
> Replace the handset?
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Background noise

[khalid touati]

Hi!
so after using the "sip show channels" commands i can see that most of the
the communication are under ulaw format and one or two are under gsm, do you
guys know a way to force the system to use only ulaw, is it a good idea and
is it gonna solve my static noise issue?
actually, the caller and the called hear the static noise just when handset
is used, once i use the speaker it's gone.
actual situation:
-once i use the handset to dial i can here the noise.
-i have the same model polycom phone configured with another server
(asterisk 1.4), and guess what no noise at all.
any guess!


2010/3/27 khalid touati 

> Thank you very mutch Philip, i'll use these commands and get back with the
> output.
>
> 2010/3/26 Philipp von Klitzing 
>
> Hi!
>>
>> > it should be some commands that can give me a better idea about the
>> > codecs, if anyone know them, please help!
>>
>> Use "sip show channels" and "iax show channels" and look at the Format
>> column.
>>
>> About the Polycom devices: Others will have to help you there. I have no
>> good guess why you might have the issue only on speakerphone, but not in
>> handset mode. Could it maybe be some kind of electrical grounding issue
>> (instead of something caused by transcoding)?
>>
>> Philipp
>>
>>
>> --
>> _
>> -- 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
>>
>
>
>
> --
> Abdullah
>



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Re: [asterisk-users] Background noise

[khalid touati]

Thank you very mutch Philip, i'll use these commands and get back with the
output.

2010/3/26 Philipp von Klitzing 

> Hi!
>
> > it should be some commands that can give me a better idea about the
> > codecs, if anyone know them, please help!
>
> Use "sip show channels" and "iax show channels" and look at the Format
> column.
>
> About the Polycom devices: Others will have to help you there. I have no
> good guess why you might have the issue only on speakerphone, but not in
> handset mode. Could it maybe be some kind of electrical grounding issue
> (instead of something caused by transcoding)?
>
> Philipp
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>



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Re: [asterisk-users] Background noise

[khalid touati]

Hi Philip,
So i looked at the codecs in the device (polycom) it says only G.711 and
ulaw can be used, i made an internal call using two phones that are
configured just with sip (so IAX not involved) but the static noise is
there, i typed show sip peer  and this is the only thing i got:
 Codecs   : 0x8000e (gsm|ulaw|alaw|h263)
 Codec Order  : (none)
it should be some commands that can give me a better idea about the codecs,
if anyone know them, please help!

2010/3/25 Philipp von Klitzing 

> Hi!
>
> > i have recently connected my (working) asterisk 1.2 server, with two
> > 1.4 asterisk servers (one using SIP the other using IAX), since then
> > (i believe) people starts complaining about a high background noise
>
> The best idea is probably to start out by looking at the codecs. If you
> happen to have either gsm or g726 somewhere along your media path then I
> would strongly suggest you start to get rid of them so see if that
> eliminates the issue. g726 is one big mess, and gsm had some compiler
> issues in the past.
>
> The next thing to look at is if the issue only appears when IAX is
> involved.
>
> Maybe looking at the Polycom firmware, and release notes in particular
> (no clue how detailed those are), would also be a good idea.
>
> Philipp
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>



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[asterisk-users] Background noise

[khalid touati]

Hi Guys,
i have recently connected my (working) asterisk 1.2 server, with two 1.4
asterisk servers (one using SIP the other using IAX), since then (i believe)
people starts complaining about a high background noise when using the
handset on Polycom phones (but when using the speaker it's fine, and i
noticed that my self), my question is, can anybody tell me any step to begin
diagnosing the issue, to be honest i don't know from where to begin!!

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Re: [asterisk-users] InterPBX communication using SIP

[khalid touati]

OK Guys i got fixed the phones i was using were registered in both servers
which is not good, once i removed them it started working!

2010/3/4 khalid touati 

> Hi Guys,
> i am using the following config in pbx1:
> register => pbx1:endop...@172.16.200.175 
> [pbx2]
> type=friend
> host=dynamic
> trunk=yes
> sercret=password
> context=[default]
> deny=0.0.0.0/0.0.0.0
> permit=172.16.200.175/255.255.255.128
>
> in pbx2:
> register => pbx2:endop...@172.16.200.176 
> [pbx1]
> type=friend
> host=dynamic
> trunk=yes
> sercret=password
> context=[default]
> deny=0.0.0.0/0.0.0.0
> permit=172.16.200.176/255.255.255.128
>
> and i get the following in pbx1:
> -- Executing [18...@default:1] Dial("SIP/8029-b7413678",
> "SIP/pbx2/8021||TWw") in new stack
> -- Called pbx2/8021
> [Mar  4 16:49:13] WARNING[3392]: chan_sip.c:12679 handle_response_invite:
> Received response: "Forbidden" from '"Khalid Touati" <
> sip:8...@172.16.200.176 >;tag=as1dcf5ff2'
> -- SIP/pbx2-09cf4468 is circuit-busy
>   == Everyone is busy/congested at this time (1:0/1/0)
>   == Auto fallthrough, channel 'SIP/8029-b7413678' status is 'CONGESTION'
>
> though i am using the same config in IAX and it's working fine, also it's
> in the same context (so i believe it's a context issue).
>
>
> --
> Abdullah
>



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[asterisk-users] InterPBX communication using SIP

[khalid touati]

Hi Guys,
i am using the following config in pbx1:
register => pbx1:endop...@172.16.200.175 
[pbx2]
type=friend
host=dynamic
trunk=yes
sercret=password
context=[default]
deny=0.0.0.0/0.0.0.0
permit=172.16.200.175/255.255.255.128

in pbx2:
register => pbx2:endop...@172.16.200.176 
[pbx1]
type=friend
host=dynamic
trunk=yes
sercret=password
context=[default]
deny=0.0.0.0/0.0.0.0
permit=172.16.200.176/255.255.255.128

and i get the following in pbx1:
-- Executing [18...@default:1] Dial("SIP/8029-b7413678",
"SIP/pbx2/8021||TWw") in new stack
-- Called pbx2/8021
[Mar  4 16:49:13] WARNING[3392]: chan_sip.c:12679 handle_response_invite:
Received response: "Forbidden" from '"Khalid Touati" <
sip:8...@172.16.200.176 >;tag=as1dcf5ff2'
-- SIP/pbx2-09cf4468 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
  == Auto fallthrough, channel 'SIP/8029-b7413678' status is 'CONGESTION'

though i am using the same config in IAX and it's working fine, also it's in
the same context (so i believe it's a context issue).


-- 
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Re: [asterisk-users] Dial cellphone from one PBX1 to PBX2? is it possible?

[khalid touati]

OK! you're right Mr William, that worked, unfortunately i don't know if
there is any points to affect to somebody who helped you (like rating in
other forums) :(, but one more request can you recommand me a book to master
and tweak dialplans?

2010/1/29 William Stillwell (Lists) 

>  You are using contexts..
>
>
>
> Look @ destination pbx, you should see  something like this:
>
>
>
> Rejected connect attempt from , request ‘ext@ conext>’ does not exist
>
>
>
> If you didn’t put a context under the peer, it uses the default one in the
> iax.conf file which is normally [default]
>
>
>
>
>
>
>
>
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *khalid touati
> *Sent:* Friday, January 29, 2010 11:54 AM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Dial cellphone from one PBX1 to PBX2? is
> it possible?
>
>
>
> Hi William,
>
> I appreciate your answer, though can you make things more clear for me:
>
> 1- i am not using extensions when registering PBX boxes in IAX files.
>
> 2- is inbounx context in the call sender PBX (pbx1) and outbound context is
> in the call receiver (or dialer) PBX (pbx2)?
>
> 3- i am using two identical dialplan's is this gonna confuse
> the communication process (contextes's name are duplicated over the two
> servers)
>
>
>
> thank you very much for making it clear for me!
>
> 2010/1/28 William Stillwell (Lists) 
>
> Your inbound context needs to have access to your outbound context.
>
>
>
> [iax-inbound]
>
>
>
> Include => outbound-conext
>
>
>
>
>
> [outbound-context]
>
>
>
> Exten => _1NXXNXX,1,Dial(DAHDI\g1\${EXTEN})
>
>
>
>
>
>
>
> Something like that.
>
>
>
>
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *khalid touati
> *Sent:* Thursday, January 28, 2010 3:29 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Dial cellphone from one PBX1 to PBX2? is it
> possible?
>
>
>
> Hi Guys,
>
> i am using two PBX's i can call from pbx1 a cellphone tied to pbx2 that
> way:
>
> 1) use a phone in PBX1
>
> 2) call extension in PBX2
>
> 3) extensions in PBX2 ring to a cellphone (as this specific ext is tied to
> a cellphone)
>
>
>
> my questions now is : am i gonna be able to dial from an IPphone registered
> within PBX1 to a cellphone by using the Trunk (Zap or dahdi) of PBX2?
> anybody know
>
> IPphone->PBX1-IAX>PBX2PRI
> line--->cellphone???
>
> thank you for you help guys!!
> --
> Abdullah
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
> --
> Abdullah
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Abdullah
-- 
_
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Re: [asterisk-users] Dial cellphone from one PBX1 to PBX2? is it possible?

[khalid touati]

Hi William,
I appreciate your answer, though can you make things more clear for me:
1- i am not using extensions when registering PBX boxes in IAX files.
2- is inbounx context in the call sender PBX (pbx1) and outbound context is
in the call receiver (or dialer) PBX (pbx2)?
3- i am using two identical dialplan's is this gonna confuse
the communication process (contextes's name are duplicated over the two
servers)

thank you very much for making it clear for me!

2010/1/28 William Stillwell (Lists) 

>  Your inbound context needs to have access to your outbound context.
>
>
>
> [iax-inbound]
>
>
>
> Include => outbound-conext
>
>
>
>
>
> [outbound-context]
>
>
>
> Exten => _1NXXNXX,1,Dial(DAHDI\g1\${EXTEN})
>
>
>
>
>
>
>
> Something like that.
>
>
>
>
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *khalid touati
> *Sent:* Thursday, January 28, 2010 3:29 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Dial cellphone from one PBX1 to PBX2? is it
> possible?
>
>
>
> Hi Guys,
>
> i am using two PBX's i can call from pbx1 a cellphone tied to pbx2 that
> way:
>
> 1) use a phone in PBX1
>
> 2) call extension in PBX2
>
> 3) extensions in PBX2 ring to a cellphone (as this specific ext is tied to
> a cellphone)
>
>
>
> my questions now is : am i gonna be able to dial from an IPphone registered
> within PBX1 to a cellphone by using the Trunk (Zap or dahdi) of PBX2?
> anybody know
>
> IPphone->PBX1-IAX>PBX2PRI
> line--->cellphone???
>
> thank you for you help guys!!
> --
> Abdullah
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Abdullah
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Dial cellphone from one PBX1 to PBX2? is it possible?

[khalid touati]

Hi Guys,
i am using two PBX's i can call from pbx1 a cellphone tied to pbx2 that way:
1) use a phone in PBX1
2) call extension in PBX2
3) extensions in PBX2 ring to a cellphone (as this specific ext is tied to a
cellphone)

my questions now is : am i gonna be able to dial from an IPphone registered
within PBX1 to a cellphone by using the Trunk (Zap or dahdi) of PBX2?
anybody know
IPphone->PBX1-IAX>PBX2PRI
line--->cellphone???
thank you for you help guys!!
-- 
Abdullah
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] [inter-pbx commnication] trying to make PBX1 talk to PBX2

[khalid touati]

Hi William,
thank you very much for your response, actually i used the same config but i
removed the mention of the context, and it went through!

2010/1/26 William Stillwell (Lists) 

>  This is how I did it..
>
>
>
> I have to Servers, SRV1 and SRV2
>
>
>
> In SRV1 iax.conf
>
>
>
> [SRV1-SRV2]
>
> type=peer
>
> username=SRV1-SRV2
>
> secret=Password1
>
> host=
>
> qualify=yes
>
>
>
> [SRV2-SRV1]
>
> type=user
>
> username=SRV2-SRV1
>
> secret=Password2
>
> context=from-iax
>
> host=
>
> quailfy=yes
>
>
>
>
>
> If I need to make calls on other box, I do Dial(IAX2/SRV1-SRV2/XX)
> where X is in destination “from-iax” context
>
>
>
> On SRV2 iax.conf
>
>
>
> [SRV1-SRV2]
>
> type=user
>
> username= SRV1-SRV2
>
> secret=Password1
>
> host=
>
> context=from-iax
>
> qualify=yes
>
>
>
> [SRV2-SRV1]
>
> type=peer
>
> username= SRV2-SRV1
>
> secret=Password2
>
> host=
>
> qualify=yes
>
>
>
> And calls from Here to There are Dial(IAX2/SRV2-SRV1/X) where  is
> in destination “from-iax” context
>
>
>
>
>
>
>
>
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *khalid touati
> *Sent:* Tuesday, January 26, 2010 10:11 AM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] [inter-pbx commnication] trying to make PBX1
> talk to PBX2
>
>
>
> Hi All,
> i want to make an extension from pbx1 able to tlak to another extension
> from pbx2 or use pbx2's trunk to dial outside calls.
>
>
> so i edited in both servers accordinally the iax.conf:
>
> ..
>
> ..
>
> ..
>
> when i type "iax2 show peers" i notice that pbx's are registred. of course
> still didn't attend my goal, do anybody have an idea how to make this
> happend?!
>
> --
> Abdullah
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Abdullah
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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