Re: [asterisk-users] Movistar sip Mexico

[Alyed]

Have you followed the instructions in:
https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway
and: http://www.voip-info.org/wiki/view/Asterisk+T.38+Gateway
??

If possible try with a different ATA since it seems not all of them work
fine with fax pass trough.

Alyed

2013/11/21 Damian Gonzalez 

> Hi,
>
> I have Asterisk 10.12.1. I can not figure out the solution.
>
> Thank you for your help.
>
> Best Regards
>
>
> On Thu, Nov 21, 2013 at 7:07 PM, Alyed  wrote:
>
>> Which version of Asterisk are you using?
>>
>> According to http://www.voip-info.org/wiki/view/Asterisk%20T.38 unless
>> you are using Asterisk 10, there's quite some patching (or buying) you'll
>> need to be doing.
>>
>> Alyed
>>
>>
>> 2013/11/21 Bryant Zimmerman 
>>
>>> Can you funnel them through a specific inbound dial context. Then force
>>> a re-invite to g729?
>>>
>>> Thanks
>>>
>>> Bryant Zimmerman (ZK Tech Inc.)
>>> 616-855-1030 Ext. 2003
>>>
>>>
>>> --
>>> *From*: "Damian Gonzalez" 
>>> *Sent*: Thursday, November 21, 2013 8:25 AM
>>> *To*: "Asterisk Users Mailing List - Non-Commercial Discussion" <
>>> asterisk-users@lists.digium.com>
>>> *Subject*: Re: [asterisk-users] Movistar sip Mexico
>>>
>>>
>>> Any posible solution?
>>>
>>>
>>> On Wed, Nov 20, 2013 at 6:03 PM, Kristian Kielhofner 
>>> wrote:
>>>
>>>> It is possible that Asterisk requires an rtpmap even for static payload
>>>> types (I'm not sure about this).  The INVITE from your provider omits
>>>> rtpmap for payload type 18 (G729) which is perfectly valid.
>>>>
>>>>
>>>> On Wed, Nov 20, 2013 at 2:56 PM, Damian Gonzalez >>> > wrote:
>>>>
>>>>> Hello,
>>>>>
>>>>> Thanks for the quickly response. I have only G729 in the peer but I
>>>>> have t38pt_udptl= yes
>>>>>
>>>>> If I put t38pt_udptl=no , asterisk reject the call with 488 code.
>>>>>
>>>>> The problem is that Movistar send T38 codec in all calls and I need
>>>>> ignore only if in the SDP I have G729 and T38 (18 and 101), but if I have
>>>>> only T38 I have to negociate a fax call.
>>>>>
>>>>> Thanks.
>>>>>
>>>>>
>>>>> On Wed, Nov 20, 2013 at 4:46 PM, Alyed  wrote:
>>>>>
>>>>>> Think you only need to make sure you have in your sip.conf file these
>>>>>> configs:
>>>>>>
>>>>>> [your-device-name]
>>>>>> .
>>>>>> .
>>>>>> disallow=all
>>>>>> allow=g729
>>>>>> .
>>>>>> .
>>>>>>
>>>>>>
>>>>>> Alyed
>>>>>>
>>>>>> 2013/11/20 Damian Gonzalez 
>>>>>>
>>>>>>> Hello,
>>>>>>>
>>>>>>> I have a problem with movistar in Mexico with a sip calls. Movistar
>>>>>>> send to me T38 and G729 in the INVITE and they say that I have to ignore
>>>>>>> T38 and use G729 in the voice call.
>>>>>>>
>>>>>>> When a fax call is made Movistar send only T38 in the INVITE.
>>>>>>>
>>>>>>> Invite example:
>>>>>>>
>>>>>>> v=0
>>>>>>> o=GDL-BMSW-12D 19913379 19899826 IN IP4 192.168.1.2
>>>>>>> s=sip call
>>>>>>> c=IN IP4 192.168.1.2
>>>>>>> t=0 0
>>>>>>> m=audio 6370 RTP/AVP 18 101
>>>>>>> a=fmtp:18 annexb=yes
>>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>> a=fmtp:101 0-15
>>>>>>> a=ptime:20
>>>>>>> m=image 6372 udptl t38
>>>>>>> a=T38FaxVersion:0
>>>>>>> a=T38FaxMaxBuffer:1100
>>>>>>> a=T38FaxMaxDatagram:612
>>>>>>> a=T38MaxBitRate:14400
>>>>>>> a=T38FaxRateManagement:transferredTCF
>>>>>>> a=T38FaxUdpEC:t38UDPRedundancy
>>>>>>>
>>>>>>> How can I  ignore T38 and use only G729 for 

Re: [asterisk-users] Movistar sip Mexico

[Alyed]

Which version of Asterisk are you using?

According to http://www.voip-info.org/wiki/view/Asterisk%20T.38 unless you
are using Asterisk 10, there's quite some patching (or buying) you'll need
to be doing.

Alyed


2013/11/21 Bryant Zimmerman 

> Can you funnel them through a specific inbound dial context. Then force a
> re-invite to g729?
>
> Thanks
>
> Bryant Zimmerman (ZK Tech Inc.)
> 616-855-1030 Ext. 2003
>
>
> --
> *From*: "Damian Gonzalez" 
> *Sent*: Thursday, November 21, 2013 8:25 AM
> *To*: "Asterisk Users Mailing List - Non-Commercial Discussion" <
> asterisk-users@lists.digium.com>
> *Subject*: Re: [asterisk-users] Movistar sip Mexico
>
>
> Any posible solution?
>
>
> On Wed, Nov 20, 2013 at 6:03 PM, Kristian Kielhofner wrote:
>
>> It is possible that Asterisk requires an rtpmap even for static payload
>> types (I'm not sure about this).  The INVITE from your provider omits
>> rtpmap for payload type 18 (G729) which is perfectly valid.
>>
>>
>> On Wed, Nov 20, 2013 at 2:56 PM, Damian Gonzalez 
>> wrote:
>>
>>> Hello,
>>>
>>> Thanks for the quickly response. I have only G729 in the peer but I have
>>> t38pt_udptl= yes
>>>
>>> If I put t38pt_udptl=no , asterisk reject the call with 488 code.
>>>
>>> The problem is that Movistar send T38 codec in all calls and I need
>>> ignore only if in the SDP I have G729 and T38 (18 and 101), but if I have
>>> only T38 I have to negociate a fax call.
>>>
>>> Thanks.
>>>
>>>
>>> On Wed, Nov 20, 2013 at 4:46 PM, Alyed  wrote:
>>>
>>>> Think you only need to make sure you have in your sip.conf file these
>>>> configs:
>>>>
>>>> [your-device-name]
>>>> .
>>>> .
>>>> disallow=all
>>>> allow=g729
>>>> .
>>>> .
>>>>
>>>>
>>>> Alyed
>>>>
>>>> 2013/11/20 Damian Gonzalez 
>>>>
>>>>> Hello,
>>>>>
>>>>> I have a problem with movistar in Mexico with a sip calls. Movistar
>>>>> send to me T38 and G729 in the INVITE and they say that I have to ignore
>>>>> T38 and use G729 in the voice call.
>>>>>
>>>>> When a fax call is made Movistar send only T38 in the INVITE.
>>>>>
>>>>> Invite example:
>>>>>
>>>>> v=0
>>>>> o=GDL-BMSW-12D 19913379 19899826 IN IP4 192.168.1.2
>>>>> s=sip call
>>>>> c=IN IP4 192.168.1.2
>>>>> t=0 0
>>>>> m=audio 6370 RTP/AVP 18 101
>>>>> a=fmtp:18 annexb=yes
>>>>> a=rtpmap:101 telephone-event/8000
>>>>> a=fmtp:101 0-15
>>>>> a=ptime:20
>>>>> m=image 6372 udptl t38
>>>>> a=T38FaxVersion:0
>>>>> a=T38FaxMaxBuffer:1100
>>>>> a=T38FaxMaxDatagram:612
>>>>> a=T38MaxBitRate:14400
>>>>> a=T38FaxRateManagement:transferredTCF
>>>>> a=T38FaxUdpEC:t38UDPRedundancy
>>>>>
>>>>> How can I  ignore T38 and use only G729 for this call?.
>>>>>
>>>>> Thanks for your help.
>>>>>
>>>>> Damian
>>>>>
>>>>>
>>>>> --
>>>>>
>>>>>
>>>>> --
>>>>> _
>>>>> -- 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 --
>>>> 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/aste

Re: [asterisk-users] Movistar sip Mexico

[Alyed]

Think you only need to make sure you have in your sip.conf file these
configs:

[your-device-name]
.
.
disallow=all
allow=g729
.
.


Alyed

2013/11/20 Damian Gonzalez 

> Hello,
>
> I have a problem with movistar in Mexico with a sip calls. Movistar send
> to me T38 and G729 in the INVITE and they say that I have to ignore T38 and
> use G729 in the voice call.
>
> When a fax call is made Movistar send only T38 in the INVITE.
>
> Invite example:
>
> v=0
> o=GDL-BMSW-12D 19913379 19899826 IN IP4 192.168.1.2
> s=sip call
> c=IN IP4 192.168.1.2
> t=0 0
> m=audio 6370 RTP/AVP 18 101
> a=fmtp:18 annexb=yes
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
> m=image 6372 udptl t38
> a=T38FaxVersion:0
> a=T38FaxMaxBuffer:1100
> a=T38FaxMaxDatagram:612
> a=T38MaxBitRate:14400
> a=T38FaxRateManagement:transferredTCF
> a=T38FaxUdpEC:t38UDPRedundancy
>
> How can I  ignore T38 and use only G729 for this call?.
>
> Thanks for your help.
>
> Damian
>
>
> --
>
>
> --
> _
> -- 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] External sip phones register with the servers IP...

[Alyed]

Please post one of your sip.conf phone configs, so we can have a look.

Alyed


2013/8/2 Carlos Chavez 

> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> On 8/1/13 9:17 PM, Michael L. Young wrote:
> > - Original Message -
> >> From: "Carlos Chavez"  To:
> >> asterisk-users@lists.digium.com Sent: Thursday, August 1, 2013
> >> 8:41:19 PM Subject: [asterisk-users] External sip phones register
> >> with the servers IP...
> >>
> >> We have just updated our office server to Asterisk 11.4.0 from
> >> 1.8.15 and internally everything is working fine.  The problem we
> >> are having is that we cannot use any external phone connected
> >> through the Internet.  This used to work fine with 1.8 but since
> >> the upgrade whenever you register any phone from an outside
> >> network the phone tries to register using the servers internal
> >> IP.
> >>
> >> I endo up having something like this:
> >>
> >> Sending to 187.163.93.235:58545 (no NAT) -- Registered SIP '2003'
> >> at 192.168.2.50:58545 Reliably Transmitting (no NAT) to
> >> 192.168.2.50:58545: OPTIONS sip:2003@192.168.2.50:58545;ob
> >> SIP/2.0 Via: SIP/2.0/UDP
> >> 192.168.2.50:5060;branch=z9hG4bK5f2019c0 Max-Forwards: 70 From:
> >> "asterisk" ;tag=as4ed13172 To:
> >>  Contact:
> >>  Call-ID:
> >> 46fd0ef840d6781d219269ae415e156e@192.168.2.50:5060 CSeq: 102
> >> OPTIONS User-Agent: Asterisk PBX 11.4.0 Date: Fri, 02 Aug 2013
> >> 00:27:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
> >> SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer
> >> Content-Length: 0
> >>
> >> I really cannot understand what is wrong, I have checked my
> >> sip.conf configuration and it is the same as in past versions.
> >> externaddr and localnet are set to the proper values.  Any
> >> ideas?
> >
> > Did you look at the CHANGES file?  There are new settings for NAT.
> > If you are using the same settings as in 1.8, there is a posiblity
> > that you will have problems depending on what settings you have
> > (which you did not include in this message).
> >
> > Also, I would recommend 11.5 since there was a one-way audio issue
> > fixed related to using the two new NAT settings.
> >
> I have tried with all nat variations and I get the same result.  I
> upgraded to 11.5 yesterday but same problem.  External phones still
> register using the servers internal IP.  IAX works fine.
>
> - --
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez Prats
> Director de Tecnología
> +52-55-91169161 ext 2001
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> =j+t/
> -END PGP SIGNATURE-
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> _
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Re: [asterisk-users] DAHDI 1.4 on Kernel 3.0

[Alyed]

Thanks a lot for the link and the tip. Have been trying it these days and
think it wil work on my system.

Thanks again Shaun.

2012/11/8 Shaun Ruffell 

> On Tue, Nov 06, 2012 at 06:49:09PM -0600, Alyed wrote:
> > Hello listers,
> >
> > I'm trying to run DAHDI 1.4 on a 3.0 Debian Kernel in an embedded system,
> > but have faced lots of problems mainly because it has lots of functions
> > looking for the PCI.
> >
> > Have seen so many problems, I'm in fact thinking it cannot be possibly
> done
> > (at least not in a couple of weeks, by one only man). Has anyone out
> there
> > had any experience on something like this? or can someone shed some light
> > on how to overcome this issues?
> >
> > Any ideas are very welcome
>
> There isn't a 1.4 version of DAHDI. However version v2.6.0 will not
> build any PCI drivers if the Kernel does not have the PCI bus
> configured.
>
> [1] http://svnview.digium.com/svn/dahdi?view=revision&revision=10397
>
> --
> Shaun Ruffell
> Digium, Inc. | Linux Kernel Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: www.digium.com & www.asterisk.org
>
> --
> _
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[asterisk-users] DAHDI 1.4 on Kernel 3.0

[Alyed]

Hello listers,

I'm trying to run DAHDI 1.4 on a 3.0 Debian Kernel in an embedded system,
but have faced lots of problems mainly because it has lots of functions
looking for the PCI.

Have seen so many problems, I'm in fact thinking it cannot be possibly done
(at least not in a couple of weeks, by one only man). Has anyone out there
had any experience on something like this? or can someone shed some light
on how to overcome this issues?

Any ideas are very welcome
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[asterisk-users] Distortion and block on analog lines

[Alyed]

Hi listers!

Have a problem with distortion in some analog lines. When some call comes in
from PSTN the sound is really distorte, nothing can be understanded, but
Internal calls work ok.

Funny thing is that when I start/stop asterisk,dahdi, and wanrouter services
eveything goes fine again. This is happening every week or so. I'm using
asterisk 1.4.36, dahdi linux 2.2.0.2 and wanpipe 3.4.9
stable
version, as you can guess I'm using Sangoma cards, specifically A400BRMDE

Sometimes it also happens that the lines block, so I'm unable to make
outbound calls using those lines.

both problems solve after services restart

any ideas?
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Re: [asterisk-users] Press twice *

[Alyed]

try it with "_" in front of the "*"

exten => _**,1,.


Alyed

2010/6/4 Danny Nicholas 

>  Probably going to have to use read to detect this..
>
>
>  --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Anahi Ludueña
> *Sent:* Friday, June 04, 2010 4:59 PM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] Press twice *
>
>
>
> Hi people, I need to detect when the user presses twice *...
> In the dialplan I added the following, but it doesn't work.
> Could you help me with that?
>
> exten => **,1,.
>
>
>  *
> --
> *
>
> *Anahi Ludueña*
>
>
>
>
>
>  --
>
> ¿Un navegador seguro buscando estás? ¡Protegete ya en
> www.ayudartepodria.com! <http://www.ayudartepodria.com>
>
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Re: [asterisk-users] Pattern matching - how to ignore numbers after 10 digits

[Alyed]

I guess it's the "!", sometimes it has a funny behaviour.

try changing ("." instead of "!" and an "X" less)
exten => 
_91XX!,1,DIAL(SIP/${EXTEN:1...@ia.ntelos.net)
; long distance
to
exten => 
_91X.,1,DIAL(SIP/${EXTEN:1...@ia.ntelos.net)
; long distance


I always use "." and never had a problem.

Alyed


2010/5/27 Eddie Mikell 

> All:
>
> Yesterday I discovered something interesting.  I dialed 1800ANCESTRY
> from the asterisk system I am testing and got the number doesn't exist
> message.  I then dialed the same number from our old system and it went
> through.
>
> I realized that the "Y" in ancestry made the number too long, and went
> back to my dialplan.
>
> How do I ignore numbers that are too long?  Obviously, I've done
> something wrong in my pattern matching.
>
> outgoing part of extensions.conf
>
> exten => 
> _91XX!,1,DIAL(SIP/${EXTEN:1...@ia.ntelos.net)
> ; long distance
> exten => 
> _9765XXX,1,DIAL(SIP/${EXTEN:1...@ia.ntelos.net)
> ; local
> exten => 
> _9XXX,1,DIAL(SIP/${EXTEN:1...@ia.ntelos.net)
> ; local
> exten => 
> _9011XXX!,1,DIAL(SIP/${EXTEN:1...@ia.ntelos.net)
> ; international
> exten => _911,1,DIAL(SIP/${ext...@ia.ntelos.net )
> ; emergency
>
> Thanks!
>
> Eddie Mikell
>
>
>
>
>
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[asterisk-users] Aastra i740 and Asterisk

[Alyed]

Hi listers!

Just ran across a customer who wants to replace an Aastra Nexspan with an
Asterisk 1.6.X, wants also to connect it to a MOCS (Microsoft Office
Comunications Server) though that's not my real concern right now.

I got one of his phones (Aastra conexity i740) and though I have been able
to change te phone's IP, GW and mask parameters, have not yet a clue on how
to make it register with asterisk.

Has anyone out there got some experience dealing with something similar??

Thanks!

Alyed
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Re: [asterisk-users] meetme / upgrade to 1.6.2.6

[Alyed]

You are right if going from 1.4.X to 1.6.2.X or similar that's the best, but
if not moving from revision, then I don't think you need to remake the
sample files.

Alyed

2010/4/19 Carlos Chavez 

> On Mon, 2010-04-19 at 11:19 -0500, Alyed wrote:
> > If that's the case what I usually do is just stop asterisk, delete the
> > contents of /usr/lib/asterisk/modules/ (back it up first!) and compile
> > the new version (don't run make samples if you want to preserve your
> > old .conf files).
> >
> > When using extra modules (like G729 codec)  be sure to follow their
> > instructions for asterisk upgrade.
> >
>When upgrading from a different revision of Asterisk I would not
> recommend keeping your old config files in place. It is better to
> remove/rename (and backup) /etc/asterisk and /usr/lib/asterisk/modules
> and do a new make; make install; make samples.  After that compare your
> config files with the new samples and insert your modifications.  This
> is because the keywords/format in the configuration files changes from
> version to version and keeping your old config files may cause Asterisk
> to behave strangely if it is missing an important parameter.
>
> --
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez Prats
> Director de Tecnología
> +52-55-91169161 ext 2001
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] meetme / upgrade to 1.6.2.6

[Alyed]

If that's the case what I usually do is just stop asterisk, delete the
contents of /usr/lib/asterisk/modules/ (back it up first!) and compile the
new version (don't run make samples if you want to preserve your old .conf
files).

When using extra modules (like G729 codec)  be sure to follow their
instructions for asterisk upgrade.

Alyed


2010/4/18 Tonty T 

> You got him wrong.
>
> He actually want to know the steps to upgrade to version 1.6.2 so he do can
> a conference bridge using confbridge instead of of meetme because he does
> not have dahdi installed.
>
> He just want to know how to upgrade from an older version to version 1.6.2
>
>
>
> On Sun, Apr 18, 2010 at 11:52 PM, Alyed  wrote:
>
>> I guess what you meant, is you don't have a physical card to provide the
>> timing needed by Meetme. Then, if you are looking for dahdi to use kernel
>> timer, then you need not to upgrade Aterisk but Dahdi to 2.3.0
>>
>> Alyed
>>
>>
>> 2010/4/18 Thomas Perron 
>>
>> I read that I need to run 1.6.2.6 (at least 1.6) to use the MeetMe
>>> application since I don't have a zdummy timing driver.
>>> Anyway, I want to upgrade my machine to 1.6.2.6.
>>> Does anyone have the exact steps?
>>> I see a lot of references on the web but any other links from our
>>> community may be preferred!
>>> Thank you
>>> Tom
>>>
>>> --
>>>
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>>>
>>
>>
>>
>>
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Re: [asterisk-users] X-lite direct sip call - Is it possible?

[Alyed]

Sorry for my mislead should have said "I've never been able to with xlite"
it's just with Sjphone it's straight forward.

Alyed

2010/4/19 bruce bruce 

> That is not correct. It's possible by adding a display name and adding the
> IP address of the pbx you are calling as the host ip. Then uncheck the
> register button and place calls in format: ex...@ip
>
> For adding characters and @ sign, push "space bar" and then type whatever
> you wish.
>
> -Bruce
>
> On Mon, Apr 19, 2010 at 12:08 AM, Alyed  wrote:
>
>> You can't do that with Xlite, try Sjphone instead.
>>
>> Alyed
>>
>>
>> 2010/4/17 bruce bruce 
>>
>>> Hi Guys,
>>>
>>> Wondering if anyone has tried to make a direct SIP peer to peer call
>>> using x-lite without any registrations of any sort. I can't seem to find the
>>> setting.
>>>
>>> Thanks,
>>> bruce
>>>
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>>>
>>
>>
>>
>>
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>
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Re: [asterisk-users] X-lite direct sip call - Is it possible?

[Alyed]

You can't do that with Xlite, try Sjphone instead.

Alyed


2010/4/17 bruce bruce 

> Hi Guys,
>
> Wondering if anyone has tried to make a direct SIP peer to peer call using
> x-lite without any registrations of any sort. I can't seem to find the
> setting.
>
> Thanks,
> bruce
>
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> _
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Re: [asterisk-users] meetme / upgrade to 1.6.2.6

[Alyed]

I guess what you meant, is you don't have a physical card to provide the
timing needed by Meetme. Then, if you are looking for dahdi to use kernel
timer, then you need not to upgrade Aterisk but Dahdi to 2.3.0

Alyed


2010/4/18 Thomas Perron 

> I read that I need to run 1.6.2.6 (at least 1.6) to use the MeetMe
> application since I don't have a zdummy timing driver.
> Anyway, I want to upgrade my machine to 1.6.2.6.
> Does anyone have the exact steps?
> I see a lot of references on the web but any other links from our
> community may be preferred!
> Thank you
> Tom
>
> --
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Re: [asterisk-users] All incoming calls landing in [customers] context

[Alyed]

Have a look at:
http://www.voip-info.org/wiki/view/Asterisk+IAX+authentication

It's about IAX but guess will give you some good hints on how to solve your
problem.

Alyed


2010/4/13 Mike Diehl 

> Hi all,
>
> I'm trying to tighten things up a bit and I seem be be running into
> something
> that doesn't make sense to me.
>
> I've got 2 contexts, one for customers, and one for guests, that I include
> into [customers] and [default], in extensions.conf, as below:
>
> =
> [default]
> include = dial_GUEST
>
> [customers]
> include = parkedcalls
> include = dial
> =
>
> The contexts, dial, and dial_GUEST essentially handle all call routing,
> with
> the idea that guests (anonymous internet callers) can't get out to the
> pstn.
>
> The problem is that ALL incoming calls are landing in [customers] even if
> the
> caller is an unregistered SIP client.
>
> As soon as a call comes in, I see it jump immediately to x...@customers:1
> and
> this happends with registered or unregistered clients.
>
> What am I doing wrong?
>
> --
>
> Take care and have fun,
> Mike Diehl.
>
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Re: [asterisk-users] Being attacked by an Amazon EC2 ...

[Alyed]

Think we need some solution WITHIN the Asterisk core. Roderick A. suggested
something that looks nice using iptables, some others have pointed out using
RBL or fail2ban, but the best would be to have some generic solution not
dependant on third party programs.

I'm not aware of the asterisk.dev list but maybe someone can tell if they
can help us here?

Alyed


2010/4/13 Randy R 

> On Mon, Apr 12, 2010 at 7:17 PM, Darrick Hartman
>  wrote:
> > That only addresses EC2 (and assumes that Amazon has any interest in
> > protecting their reputation).  What about attacks that come from other
> > locations?  Granted it's pretty easy to buy time on an EC2 server so
> > this may be the primary source for a period of time.
>
> With the growth of the cloud offerings, this problem will likely grow,
> so  yes, a generic solution is needed. What I want to see though, and
> no provder has done much if anything about it, is REPORTING and
> INVESTIGATION. It is easy to use a script to report and submit, we can
> all do that, even I could (if I had a box running and needed to). The
> hard part is them having their tech/sys people actually look at the
> network and see, "Oh, ya, there's some shit happening that on that
> instance..."
>
> If Amazon's form submit didn't even work, that's a really bad
> reflection on their brand in a lot of ways, including tech competence.
> If that is know to geeks like us, it won't hurt them which is why,
> like a broken record, I keep saying: put your Amazon experience out to
> the public. When it starts being mentioned in Wired, "Storm Cloud" or
> something, THEN Amazon will have to do something.
>
> I do not believe Amazon is taking reasonable measures now in doing
> their job, and that they should be working towards that goal,
> reasonable measures as opposed to NO measures.
>
> /r
>
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Re: [asterisk-users] Remote registering fails

[Alyed]

>The context that I'm using for the local extensions is not [general].

Sorry quite didn't get what you mean. Nevertheless I I think it is a matter
of NAT/firewall management.

Alyed


2010/4/11 Daniel Bareiro 

> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hi, Alyed.
>
> On Sun, 11 Apr 2010, Alyed wrote:
>
> >> Daniel, you are having a problem often seen in pre 1.4.14 versions.
> >>
> >> Before this release srvlookup=no was the default for sip.conf and
> >> guess the same for iax.conf . So if you are working with a previous
> >> release just add this parameter .. but change it to
> >>
> >> serverlookup=yes
> >>
> >> under your iax.conf [general] section.
>
> > Sorry, the parameter should be.
> >
> > srvlookup=yes
>
> I'm using Asterisk 1.4.24.1. Anyway, I was seeing the file sip.conf and
> yes I have srvlookup=yes in [general]. In iax.conf it is not defined
> explicitly, so I suppose that it will be taking the value by default.
>
> The context that I'm using for the local extensions is not [general].
> Can it have to do?
>
> Thanks for your reply.
>
> Regards,
> Daniel
>
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v1.4.9 (GNU/Linux)
>
> iEYEARECAAYFAkvBw+sACgkQZpa/GxTmHTcdFQCfWiXsyRQ85s1fy9Ygb+IhlGGy
> 8kgAniMCjFLfZoyrEKKxao4FcRLsXTil
> =ltqS
> -END PGP SIGNATURE-
>
>
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Re: [asterisk-users] Remote registering fails

[Alyed]

Sorry, the parameter should be.

srvlookup=yes

Alyed

2010/4/10 Alyed 

> Daniel, you are having a problem often seen in pre 1.4.14 versions.
>
> Before this release srvlookup=no was the default for sip.conf and guess
> the same for iax.conf . So if you are working with a previous release just
> add this parameter .. but change it to
>
> serverlookup=yes
>
> under your iax.conf [general] section.
>
> Alyed
>
>
>
> 2010/4/10 Daniel Bareiro 
>
> -BEGIN PGP SIGNED MESSAGE-
>> Hash: SHA1
>>
>> Hi all!
>>
>> I'm trying to test with a friend who has an Asterisk in his office with
>> the Asterisk which I have in my house. Then I have an extension that he
>> is trying to register remotely.
>>
>> Trying with the Twinkle client, I see that it is registered:
>>
>> -
>> ---
>> 400/400190.0.163.57 D   N  5060 OK (35 ms)
>> -
>> ---
>>
>> but to the few seconds I obtain the following thing in Asterisk CLI:
>>
>> -
>> ---
>> 400/400190.0.163.57 D   N  5060
>> UNREACHABLE
>> -
>> ---
>>
>> And Twinkle gives an error "408 request timeout". And when he tries to
>> make the register through his Asterisk instead of use Twinkle, after a
>> little while he obtains errors of this type:
>>
>> -
>> ---
>> [Apr 10 19:07:18] NOTICE[16848]: chan_sip.c:7618 sip_reg_timeout:--
>> Registration for '4...@myremotehome.com' timed out, trying again
>> (Attempt #138)
>> -
>> ---
>>
>> This is the configuration that I'm using for the extension:
>>
>> -
>> ---
>> [400]
>> username=400
>> type=friend
>> secret=passwd
>> qualify=yes
>> callerid="Daniel" <400>
>> host=dynamic
>> nat=no
>> context=from-internal
>> mailbox=...@voicemail
>> canreinvite=no
>> -
>> ---
>>
>> I tried with both "nat=yes" ---as it is possible to be observed above---
>> and "nat=no", and we always obtain the same behavior. My Asterisk server
>> is installed in the same firewall with GNU/Linux.
>>
>> I don't believe that it is a problem with the ports since the client
>> registers itself at some time. Whatever happens, I'm allowing
>> connections for the remote IP to the 5060 tcp/UDP port and 1:2
>> UDP in the firewall. The router that it is ahead has these ports
>> redirected to the firewall.
>>
>> Also I'm using externhost, externip and localnet in
>> /etc/asterisk/sip.conf
>>
>>
>> Which can be the problem?
>>
>> Thanks in advance for your reply.
>>
>> Regards,
>> Daniel
>>
>> -BEGIN PGP SIGNATURE-
>> Version: GnuPG v1.4.9 (GNU/Linux)
>>
>> iEYEARECAAYFAkvBAFQACgkQZpa/GxTmHTe0mgCcCmDNhkMm3DMc/Ckd7AAzZneF
>> 4ngAn0SL/IC58kNDktcRsxJOaKPoAuCL
>> =Ve4J
>> -END PGP SIGNATURE-
>>
>>
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Re: [asterisk-users] Remote registering fails

[Alyed]

Daniel, you are having a problem often seen in pre 1.4.14 versions.

Before this release srvlookup=no was the default for sip.conf and guess the
same for iax.conf . So if you are working with a previous release just add
this parameter .. but change it to

serverlookup=yes

under your iax.conf [general] section.

Alyed



2010/4/10 Daniel Bareiro 

> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hi all!
>
> I'm trying to test with a friend who has an Asterisk in his office with
> the Asterisk which I have in my house. Then I have an extension that he
> is trying to register remotely.
>
> Trying with the Twinkle client, I see that it is registered:
>
> -
> ---
> 400/400190.0.163.57 D   N  5060 OK (35 ms)
> -
> ---
>
> but to the few seconds I obtain the following thing in Asterisk CLI:
>
> -
> ---
> 400/400190.0.163.57 D   N  5060 UNREACHABLE
> -
> ---
>
> And Twinkle gives an error "408 request timeout". And when he tries to
> make the register through his Asterisk instead of use Twinkle, after a
> little while he obtains errors of this type:
>
> -
> ---
> [Apr 10 19:07:18] NOTICE[16848]: chan_sip.c:7618 sip_reg_timeout:--
> Registration for '4...@myremotehome.com' timed out, trying again
> (Attempt #138)
> -
> ---
>
> This is the configuration that I'm using for the extension:
>
> -
> ---
> [400]
> username=400
> type=friend
> secret=passwd
> qualify=yes
> callerid="Daniel" <400>
> host=dynamic
> nat=no
> context=from-internal
> mailbox=...@voicemail
> canreinvite=no
> -
> ---
>
> I tried with both "nat=yes" ---as it is possible to be observed above---
> and "nat=no", and we always obtain the same behavior. My Asterisk server
> is installed in the same firewall with GNU/Linux.
>
> I don't believe that it is a problem with the ports since the client
> registers itself at some time. Whatever happens, I'm allowing
> connections for the remote IP to the 5060 tcp/UDP port and 1:2
> UDP in the firewall. The router that it is ahead has these ports
> redirected to the firewall.
>
> Also I'm using externhost, externip and localnet in
> /etc/asterisk/sip.conf
>
>
> Which can be the problem?
>
> Thanks in advance for your reply.
>
> Regards,
> Daniel
>
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v1.4.9 (GNU/Linux)
>
> iEYEARECAAYFAkvBAFQACgkQZpa/GxTmHTe0mgCcCmDNhkMm3DMc/Ckd7AAzZneF
> 4ngAn0SL/IC58kNDktcRsxJOaKPoAuCL
> =Ve4J
> -END PGP SIGNATURE-
>
>
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Re: [asterisk-users] Callerid over IAX Trunks

[Alyed]

Don't have a system to test this right now, but read somewhere this was a 2
steps solution:

1) Leave the callerid in your tunk definition blank (in your example the 999
username)

2) Use brakets around the callerid definition of your peers: callerid= <200>
(extension 200 in your example)

Let us know if it worked.

Alyed


2010/4/9 Ye Liu 

> Hello everyone,
>
> I'm fairly new to asterisk and this list. Currently I'm working on IAX
> trunks to send/receive calls between 2 asterisk boxes with asterisk
> 1.6.1.1+asterisk gui 2.0. After some work in the gui, two boxes can
> send/receive calls to/from the other just fine, the only problem I
> have is the caller id.
>
> Here is my setup:
>
> 1. on both boxes, I added an IAX user in the gui, say the extension
> and password are 999
> 2. I then created IAX trunks for each box using 999 as username and
> password, hostname/IP was set to be other box's IP
> 3. when done, from the system status panel, I saw the trunks
> successfully registered to the other box
> 4. then I added "Outgoing Call Rules" to each box:
>for box1, _2XX --> to_box2_trunk
>for box2, _1XX --> to_box1_trunk
>
> This setup works ok, the only problem is caller id, i.e. when
> extension(200) from box2 calls to extension(100) from box1, the call
> can be made but the caller id displayed on 100 is 999 not 200.
>
> I have been on this problem for some time already, could anyone here
> give me a bit help please?
> --
> Ye Liu (AKA @jaux)
>
> http://jaux.net
>
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Re: [asterisk-users] dnd not working correctly

[Alyed]

yes, that's the default location unless there's any change in
/etc/asterisk/asterisk.conf

I think it should be there, cause when it is not asterisk complains with a
message letting you know it wasn't able to find it.

>I dont have a working server to look at so i didn't know if i was even
looking in the right place.
Pls look for them in the server you are actually having the problems with
cause I can't remember that sound file being on the official's asterisk
release.

Alyed


2010/3/30 Ott Rose 

>  where are those sound files kept? i looked last night in
> /var/lib/asterisk/sounds and i didn't see anything named do-not-disturb.
>
> if its supposed to be in there then thats a problem. I dont have a working
> server to look at so i didn't know if i was even looking in the right place.
>
> --
> Date: Mon, 29 Mar 2010 23:58:43 -0600
> From: al...@vivoxie.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] dnd not working correctly
>
> I'm not an Amportal expert so all I can say from:
>
> > -- Executing [...@from-internal:8] Playback("SIP/117-01f6",
> "do-not-disturb&activated") in new stack
> > -- Executing [...@from-internal:9] Macro("SIP/117-01f6",
> "hangupcall,") in new stack
>
> is that Asterisk is playing the "do-not-disturb&activated" file (apparently
> without errors) and then the next instruction is to hangup the call, hence
> Asterisk hangs it up.
>
> Just to be sure play this sound file independently.
>
> Sorry but other than this there's little I can do, maybe someone else has
> experience with this.
>
> Alyed
>
>
> 2010/3/29 Ott Rose 
>
>
> i posted this on the freepbx site. here is the response
>
>
> "from the trace, everything is working. Check your asterisk log for file
> errors playing back the audio, could be your sound files are not installed
> or messed up."
>
>
>
> so i checked /etc/log/asterisk/full
>
> and in vi full i did /error   and  /117 (my ext) and /activate didn't
> really find anything
>
> i didn't see anything but i might be over looking it. I did grep error full
> and it returned some errors but not related to dnd as far as i can tell. is
> there some place else to look, a better way to search that file, or can
> someone tell me what i am looking for?
>
>
>
>
> ------
> Date: Fri, 26 Mar 2010 18:34:46 -0600
> From: al...@vivoxie.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] dnd not working correctly
>
> Seems like an Amportal configration problem not and Asterisk issue. Maybe
> you should try in one of the FreePBX users list.
>
> Alyed
>
>
>
> 2010/3/26 Ott Rose 
>
>  i have posted this question couple of times and never really got any hits
> i wasn't able to provide any debug info
>
> Connected to Asterisk 1.6.0.21 currently running on phoneserver (pid =
> 3309)
> Verbosity is at least 4
>   == Using SIP RTP TOS bits 184
>   == Using SIP RTP CoS mark 5
>   == Using SIP VRTP TOS bits 136
>   == Using SIP VRTP CoS mark 6
>   == Extension Changed 117[ext-local] new state InUse for Notify User 102
>   == Extension Changed 117[ext-local] new state InUse for Notify User 103
>   == Extension Changed 117[ext-local] new state InUse for Notify User 114
> -- Executing [...@from-internal:1] Answer("SIP/117-01f6", "") in
> new stack
> -- Executing [...@from-internal:2] Wait("SIP/117-01f6", "1") in
> new stack
> -- Executing [...@from-internal:3] Macro("SIP/117-01f6",
> "user-callerid,") in new stack
> -- Executing [...@macro-user-callerid:1] Set("SIP/117-01f6",
> "AMPUSER=117") in new stack
> -- Executing [...@macro-user-callerid:2] GotoIf("SIP/117-01f6",
> "0?report") in new stack
> -- Executing [...@macro-user-callerid:3] ExecIf("SIP/117-01f6",
> "1?Set(REALCALLERIDNUM=117)") in new stack
> -- Executing [...@macro-user-callerid:4] Set("SIP/117-01f6",
> "AMPUSER=117") in new stack
> -- Executing [...@macro-user-callerid:5] Set("SIP/117-01f6",
> "AMPUSERCIDNAME=My Name") in new stack
> -- Executing [...@macro-user-callerid:6] GotoIf("SIP/117-01f6",
> "0?report") in new stack
> -- Executing [...@macro-user-callerid:7] Set("SIP/117-01f6",
> "AMPUSERCID=117") in new stack
> -- Executing [...@macro-user-callerid:8] Set("SIP/117-01f6

Re: [asterisk-users] dnd not working correctly

[Alyed]

I'm not an Amportal expert so all I can say from:

> -- Executing [...@from-internal:8] Playback("SIP/117-01f6",
"do-not-disturb&activated") in new stack
> -- Executing [...@from-internal:9] Macro("SIP/117-01f6",
"hangupcall,") in new stack

is that Asterisk is playing the "do-not-disturb&activated" file (apparently
without errors) and then the next instruction is to hangup the call, hence
Asterisk hangs it up.

Just to be sure play this sound file independently.

Sorry but other than this there's little I can do, maybe someone else has
experience with this.

Alyed


2010/3/29 Ott Rose 

>
> i posted this on the freepbx site. here is the response
>
>
> "from the trace, everything is working. Check your asterisk log for file
> errors playing back the audio, could be your sound files are not installed
> or messed up."
>
>
>
> so i checked /etc/log/asterisk/full
>
> and in vi full i did /error   and  /117 (my ext) and /activate didn't
> really find anything
>
> i didn't see anything but i might be over looking it. I did grep error full
> and it returned some errors but not related to dnd as far as i can tell. is
> there some place else to look, a better way to search that file, or can
> someone tell me what i am looking for?
>
>
>
>
> --
> Date: Fri, 26 Mar 2010 18:34:46 -0600
> From: al...@vivoxie.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] dnd not working correctly
>
> Seems like an Amportal configration problem not and Asterisk issue. Maybe
> you should try in one of the FreePBX users list.
>
> Alyed
>
>
>
> 2010/3/26 Ott Rose 
>
>  i have posted this question couple of times and never really got any hits
> i wasn't able to provide any debug info
>
> Connected to Asterisk 1.6.0.21 currently running on phoneserver (pid =
> 3309)
> Verbosity is at least 4
>   == Using SIP RTP TOS bits 184
>   == Using SIP RTP CoS mark 5
>   == Using SIP VRTP TOS bits 136
>   == Using SIP VRTP CoS mark 6
>   == Extension Changed 117[ext-local] new state InUse for Notify User 102
>   == Extension Changed 117[ext-local] new state InUse for Notify User 103
>   == Extension Changed 117[ext-local] new state InUse for Notify User 114
> -- Executing [...@from-internal:1] Answer("SIP/117-01f6", "") in
> new stack
> -- Executing [...@from-internal:2] Wait("SIP/117-01f6", "1") in
> new stack
> -- Executing [...@from-internal:3] Macro("SIP/117-01f6",
> "user-callerid,") in new stack
> -- Executing [...@macro-user-callerid:1] Set("SIP/117-01f6",
> "AMPUSER=117") in new stack
> -- Executing [...@macro-user-callerid:2] GotoIf("SIP/117-01f6",
> "0?report") in new stack
> -- Executing [...@macro-user-callerid:3] ExecIf("SIP/117-01f6",
> "1?Set(REALCALLERIDNUM=117)") in new stack
> -- Executing [...@macro-user-callerid:4] Set("SIP/117-01f6",
> "AMPUSER=117") in new stack
> -- Executing [...@macro-user-callerid:5] Set("SIP/117-01f6",
> "AMPUSERCIDNAME=My Name") in new stack
> -- Executing [...@macro-user-callerid:6] GotoIf("SIP/117-01f6",
> "0?report") in new stack
> -- Executing [...@macro-user-callerid:7] Set("SIP/117-01f6",
> "AMPUSERCID=117") in new stack
> -- Executing [...@macro-user-callerid:8] Set("SIP/117-01f6",
> "CALLERID(all)="My Name" <117>") in new stack
> -- Executing [...@macro-user-callerid:9] GotoIf("SIP/117-01f6",
> "0?continue") in new stack
> -- Executing [...@macro-user-callerid:10] Set("SIP/117-01f6",
> "__TTL=64") in new stack
> -- Executing [...@macro-user-callerid:11] GotoIf("SIP/117-01f6",
> "1?continue") in new stack
> -- Goto (macro-user-callerid,s,18)
> -- Executing [...@macro-user-callerid:18] NoOp("SIP/117-01f6",
> "Using CallerID "My Name" <117>") in new stack
> -- Executing [...@from-internal:4] GotoIf("SIP/117-01f6",
> "1?activate:deactivate") in new stack
> -- Goto (from-internal,*76,5)
> -- Executing [...@from-internal:5] Set("SIP/117-01f6",
> "DB(DND/117)=YES") in new stack
> -- Executing [...@from-internal:6] Set("SIP/117-01f6",
> "STATE=BUSY") in new stack
> -- Executing [...@from-internal:7] Gosub("SIP/117-01f6",
> "

Re: [asterisk-users] Updating Asterisk and its use with MySQL

[Alyed]

Yes I'm talking about Asterisk Now's GUI and yes, you can just install this
component.
google for Asterisk Gui 2.0 and you'll find plenty of info.

Regarding the DB I can't help you here, maybe someone else can.

Alyed


2010/3/28 Daniel Bareiro 

> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> On Sun, 28 Mar 2010, Alyed wrote:
>
> >> My idea is to continue making the configurations by hand at the
> >> moment, that it is the way that I used until now, to familiarize to
> >> me with the handling of Asterisk at lower level, without using a
> >> graphical interface, and in a later stage of the tests to take these
> >> configurations through something like FreePBX. What think of this
> >> form to think?
>
> > I would suggest trying Digium's GUI first and then FreePBX since the
> > first one I find it more readable. You'll find out eventually that
> > there's no easy way to migrate from pure command line to a GUI, but
> > you'll learn a lot in the meantime.
>
> I didn't know that there was Digium's GUI. It is FLOSS? I was looking
> for in the site of Digium in the download section, but the unique thing
> that I saw that it speaks of a GUI is AsteriskNow, that in fact it is a
> complete distribution of GNU/Linux. You talked about to the GUI provided
> by AsteriskNow? Because if is this case, I don't believe that it is very
> practical. When I spoke of GUI was referring to a separated component to
> install over which already one had running.
>
> As far as the use of Asterisk with a DBMS (MySQL, for example), do you
> know some document or reference where indicate the steps to follow to
> migrate from config files?
>
> Thanks for your reply.
>
> Regards,
> Daniel
>
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v1.4.9 (GNU/Linux)
>
> iEYEARECAAYFAkuvu8oACgkQZpa/GxTmHTdFVACePM0WaIfeHQmM+w8cpLuGGt/5
> XSAAoI+YrC+9Y91ElRhFBrxAG6XVxEyh
> =e4iN
> -END PGP SIGNATURE-
>
>
> --
> _
> -- 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] Updating Asterisk and its use with MySQL

[Alyed]

>My idea is to continue making the configurations by hand at the moment,
>that it is the way that I used until now, to familiarize to me with the
>handling of Asterisk at lower level, without using a graphical
>interface, and in a later stage of the tests to take these
>configurations through something like FreePBX. What think of this form
>to think?

I would suggest trying Digium's GUI first and then FreePBX since the first
one I find it more readable. You'll find out eventually that there's no easy
way to migrate from pure command line to a GUI, but you'll learn a lot in
the meantime.

Have Fun!

Alyed



2010/3/28 Daniel Bareiro 

> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> - -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hi, Jim.
>
> On Sun, 28 Mar 2010, Jim Dickenson wrote:
>
> >>> Make sure not to do "make samples" or you will overwrite your .conf
> >>> file. This is the important one to watch out for. You can save off
> >>> your .conf files and then restore them or compare your files with
> >>> the new ones to see if there are any important new settings.
>
> >> I had thought that "make config" was what I would have to avoid.
> >> Which is the difference? does "make config" create the init scripts
> >> and "make samples" the example configuration files?
>
> > Yes, "make config" installs /etc/init.d/asterisk on Linux systems and
> > does the appropriate chkconfig steps so will start on boot while "make
> > samples" installs the .conf files in, by default, /etc/asterisk.
>
> Perfect.
>
> >> Do these two "makes" have the same behavior for Asterisk and DAHDI? I
> >> have understood that "make config" in DAHDI Tools is the one that
> >> creates both the configuration files and init scripts.
>
> > There is no "make config" for dahdi. I think /etc/dahdi files do not
> > get overwritten if they are there already.
>
> Hmmm... nevertheless I have documented this procedure in my Dokuwiki of
> the time that I made the installation and compilation:
>
> # tar xvzf dahdi-linux-2.1.0.4.tar.gz
> # tar xvzf dahdi-tools-2.1.0.2.tar.gz
>
> ~/Asterisk/dahdi-linux-2.1.0.4# make
> ~/Asterisk/dahdi-linux-2.1.0.4# make install
>
> ~/Asterisk/dahdi-tools-2.1.0.2# ./configure
> ~/Asterisk/dahdi-tools-2.1.0.2# make menuselect   # In order to select a
> customized configuration
> ~/Asterisk/dahdi-tools-2.1.0.2# make
> ~/Asterisk/dahdi-tools-2.1.0.2# make install
> ~/Asterisk/dahdi-tools-2.1.0.2# make config   # In order to install
> scripts and config files
>
> >> When I compiled the version that I'm using at the moment of DAHDI
> >> Linux only I used "make" and "make install" without using "make
> >> samples" or "make config". Are also generated configuration files
> >> with DAHDI Linux?
>
> > I think if you are installing dahdi complete from source you do "make
> > all" and "make install" and "make config"
>
> Thanks. I will consider it if I install this package of DAHDI.
>
>
> Thanks for your reply.
>
> Regards,
> Daniel
>
> - -BEGIN PGP SIGNATURE-
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>
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> =Q6s+
> - -END PGP SIGNATURE-
>
> -BEGIN PGP SIGNATURE-
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> iEYEARECAAYFAkuvqh8ACgkQZpa/GxTmHTfoygCfZtRoPj8ieJjWVtsIqPFIk5Q/
> 4QQAnjWRKkOJls9dFVwVM0IQORkmDIPd
> =YxoR
> -END PGP SIGNATURE-
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Cisco 7960 become UNREACHABLE behind pix firewall

[Alyed]

From: http://www.voip-info.org/wiki/view/Asterisk+sip+qualify
"If you turn on *qualify* in the configuration of a SIP device in
sip.conf<http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf>,
Asterisk will send a SIP
OPTIONS<http://www.voip-info.org/wiki/view/SIP+method+options>command
regularly to check that the device is still online. If the device
does not answer within the configured (or default) period (in ms) Asterisk
considers the device off-line for future calls. This status can be checked
by the SIPPEER 
function<http://www.voip-info.org/wiki/view/Asterisk+func+sippeer>,
and inversely this function will only provide status information for peers
which have *qualify=yes*."
My guess is that your Nat/firewall is closing the connection after some time
the phone is idle, so this way Asterisk will make sure to always have
communication going trhough that connection so your NAT/firewall won't just
close it.

try playing with qualifyfreq as well.

Let us know if it helped.

Alyed



2010/3/27 James Lamanna 

> Hi,
> I have about 10 Cisco 7960s behind a PIX 506E (IOS v6.3) firewall.
> After some period of time, asterisk says that some of them are
> unreachable, and the phones lose their registration.
> The only way to make the phones recover is to clear the NAT
> translation tables for the phones on the PIX (clear xlate...)
> Does anyone know how to fix this? As you can imagine, it is quite
> annoying. And it does not happen to all the phones either.
>
> sip fixup is enabled on the PIX
>
> phone config parts:
>
> nat_enable : 1
> nat_received_processing : 0
> nat_address: [public ip of PIX]
>
> Thank you.
>
> -- James
> (Please CC me on all replies)
>
> --
> _
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Re: [asterisk-users] need help on setup rtp directly between 2 sip clients

[Alyed]

>so the asterisk is in middle in all version, right? thank you for your
explanation
is the one whom everyone goes and says "hey I'm 101 and live downstairs can
I play with you guys?"

>my goal is asterisk is on internet - WAN IP address and the softphones are
in NAT but the xlite supports the ICE function that is >why i ask the media
should be go directly between softphones and no need go through the asterisk


I guess you still don't fully get it :)

The scenario you mention is similar (at least for the "direct call" thingy)
ICE doesn't mean you don't need to know where the callee is it just means it
will play a little with the SDP part of the SIP. Have a look at
http://www.voiptraversal.com/ice_methodology.htm to better understand what's
ICE about.

Alyed


2010/3/26 haloha 

> Hi Alyed
>
> so the asterisk is in middle in all version, right? thank you for your
> explanation
> all devices i mean are asterisk + softphones
> my goal is asterisk is on internet - WAN IP address and the softphones are
> in NAT but the xlite supports the ICE function that is why i ask the media
> should be go directly between softphones and no need go through the asterisk
>
>
> will check the SJphone feature, thank you for your suggestion
>
>
> Thank you
>
> On Sat, Mar 27, 2010 at 9:04 AM, Alyed  wrote:
>
>> If your sofphones are registering to the asterisk, then asterisk needs to
>> be in the middle, otherwise there's no way your 101 sofpthone user can
>> actually know where (by where I mean which IP) is the 102 softphone user.
>>
>> UNLESS (yes, there's a big unless) you dial from 101 DIRECTLY to 102. How?
>> well dialing directly to 102's IP.That's where Xlite doesn't work, but
>> SJphone does.
>>
>> SJphone supports the advanced SIP URI syntax which for a user is:
>> sip:usern...@the.user.ip
>>
>> Nevertheless.. if you are inside a LAN, why wouldn't you want those
>> calls to go through asterisk??? If you have collision problems I suggest you
>> fix them instead of asking everyone to call using SIP uri.
>>
>> Alyed
>>
>>
>> 2010/3/26 haloha 
>>
>>> Hi Alyed
>>>
>>>
>>> xilte softphone work perfectly on other sip server(opensips server)
>>>
>>> Don't remember the exact syntax but guess it's something like
>>> sip:usern...@the.phones.ip:
>>>>
>>>> 5060
>>>
>>>
>>> >>>you mean i config the extension.conf look like exten =>
>>> 1000,1,Dial(SIP/1...@ip address:5060), is it right?
>>>
>>> the problem i got here is the asterisk server to stay middle of  media
>>> first, then redirect the media later, how to fix it,asterisk no need stay in
>>> middle of media because all devices are in the same LAN
>>>
>>> is there another hint
>>>
>>> Thank you
>>>
>>>
>>> On Fri, Mar 26, 2010 at 11:56 PM, Alyed  wrote:
>>>
>>>> I guess to do what you want you need to dial directly between the
>>>> phones. Can't do it with xlite but you can with SJphones
>>>>
>>>> Don't remember the exact syntax but guess it's something like
>>>> sip:usern...@the.phones.ip:5060
>>>>
>>>> Alyed
>>>>
>>>>
>>>>
>>>
>>> --
>>> _
>>> -- 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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>>
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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Re: [asterisk-users] need help on setup rtp directly between 2 sip clients

[Alyed]

If your sofphones are registering to the asterisk, then asterisk needs to be
in the middle, otherwise there's no way your 101 sofpthone user can actually
know where (by where I mean which IP) is the 102 softphone user.

UNLESS (yes, there's a big unless) you dial from 101 DIRECTLY to 102. How?
well dialing directly to 102's IP.That's where Xlite doesn't work, but
SJphone does.

SJphone supports the advanced SIP URI syntax which for a user is:
sip:usern...@the.user.ip

Nevertheless.. if you are inside a LAN, why wouldn't you want those
calls to go through asterisk??? If you have collision problems I suggest you
fix them instead of asking everyone to call using SIP uri.

Alyed


2010/3/26 haloha 

> Hi Alyed
>
> xilte softphone work perfectly on other sip server(opensips server)
>
> Don't remember the exact syntax but guess it's something like
> sip:usern...@the.phones.ip:
>>
>> 5060
>
>
> >>>you mean i config the extension.conf look like exten =>
> 1000,1,Dial(SIP/1...@ip address:5060), is it right?
>
> the problem i got here is the asterisk server to stay middle of  media
> first, then redirect the media later, how to fix it,asterisk no need stay in
> middle of media because all devices are in the same LAN
>
> is there another hint
>
> Thank you
>
>
> On Fri, Mar 26, 2010 at 11:56 PM, Alyed  wrote:
>
>> I guess to do what you want you need to dial directly between the phones.
>> Can't do it with xlite but you can with SJphones
>>
>> Don't remember the exact syntax but guess it's something like
>> sip:usern...@the.phones.ip:5060
>>
>> Alyed
>>
>>
>>
>
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Re: [asterisk-users] What does this error message mean

[Alyed]

I've seen this before and I know the reason but not really the solution:

You have used this same username/password combination for another SIP
client, or maybe the same one but with different IP. Even when that one is
offline from some time on, Asterisk doesn't renew it's internal database, so
still thinks it might be somewhere there.

I guess this has to do with the SIP expiry as long as I can remember it's
defaulted to 3600 secs (1 hour). Think you should be able to overcome this
situation either by configuring the qualify option in sip.conf for that peer
or rebooting your Asterisk server (just restarting the service might not
work).

Alyed



2010/3/26 Ira 

> I get this when my brother in law tries to call in from his box to mine.
>
> WARNING[4855]: chan_sip.c:12675 check_auth: username mismatch, have
> <100>, digest has 
>
> or after changing the register line:
>
> WARNING[4855]: chan_sip.c:12675 check_auth: username mismatch, have
> <100>, digest has <199>
>
> I have done everything I can think of and still failure.
>
> Currently the register line says:
>
> register => name:pass@ 0.0.0.0/199
>
> and there is both a context [199] and [100] in extensions.conf with a
> goto to where I want and thse two lines in the declared context in
> sip.conf:
>
> exten => 100,1,goto(,,)
> exten => 199,1,goto(,,)
>
> I'm sure I'll feel quite silly when the error of my ways is pointed
> out, but for the moment I'm stumped.
>
> Ira
>
>
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Re: [asterisk-users] dnd not working correctly

[Alyed]

Seems like an Amportal configration problem not and Asterisk issue. Maybe
you should try in one of the FreePBX users list.

Alyed



2010/3/26 Ott Rose 

>  i have posted this question couple of times and never really got any hits
> i wasn't able to provide any debug info
>
> Connected to Asterisk 1.6.0.21 currently running on phoneserver (pid =
> 3309)
> Verbosity is at least 4
>   == Using SIP RTP TOS bits 184
>   == Using SIP RTP CoS mark 5
>   == Using SIP VRTP TOS bits 136
>   == Using SIP VRTP CoS mark 6
>   == Extension Changed 117[ext-local] new state InUse for Notify User 102
>   == Extension Changed 117[ext-local] new state InUse for Notify User 103
>   == Extension Changed 117[ext-local] new state InUse for Notify User 114
> -- Executing [...@from-internal:1] Answer("SIP/117-01f6", "") in
> new stack
> -- Executing [...@from-internal:2] Wait("SIP/117-01f6", "1") in
> new stack
> -- Executing [...@from-internal:3] Macro("SIP/117-01f6",
> "user-callerid,") in new stack
> -- Executing [...@macro-user-callerid:1] Set("SIP/117-01f6",
> "AMPUSER=117") in new stack
> -- Executing [...@macro-user-callerid:2] GotoIf("SIP/117-01f6",
> "0?report") in new stack
> -- Executing [...@macro-user-callerid:3] ExecIf("SIP/117-01f6",
> "1?Set(REALCALLERIDNUM=117)") in new stack
> -- Executing [...@macro-user-callerid:4] Set("SIP/117-01f6",
> "AMPUSER=117") in new stack
> -- Executing [...@macro-user-callerid:5] Set("SIP/117-01f6",
> "AMPUSERCIDNAME=My Name") in new stack
> -- Executing [...@macro-user-callerid:6] GotoIf("SIP/117-01f6",
> "0?report") in new stack
> -- Executing [...@macro-user-callerid:7] Set("SIP/117-01f6",
> "AMPUSERCID=117") in new stack
> -- Executing [...@macro-user-callerid:8] Set("SIP/117-01f6",
> "CALLERID(all)="My Name" <117>") in new stack
> -- Executing [...@macro-user-callerid:9] GotoIf("SIP/117-01f6",
> "0?continue") in new stack
> -- Executing [...@macro-user-callerid:10] Set("SIP/117-01f6",
> "__TTL=64") in new stack
> -- Executing [...@macro-user-callerid:11] GotoIf("SIP/117-01f6",
> "1?continue") in new stack
> -- Goto (macro-user-callerid,s,18)
> -- Executing [...@macro-user-callerid:18] NoOp("SIP/117-01f6",
> "Using CallerID "My Name" <117>") in new stack
> -- Executing [...@from-internal:4] GotoIf("SIP/117-01f6",
> "1?activate:deactivate") in new stack
> -- Goto (from-internal,*76,5)
> -- Executing [...@from-internal:5] Set("SIP/117-01f6",
> "DB(DND/117)=YES") in new stack
> -- Executing [...@from-internal:6] Set("SIP/117-01f6",
> "STATE=BUSY") in new stack
> -- Executing [...@from-internal:7] Gosub("SIP/117-01f6",
> "app-dnd-toggle,sstate,1") in new stack
> -- Executing [sst...@app-dnd-toggle:1] Set("SIP/117-01f6",
> "DEVICE_STATE(Custom:DND117)=BUSY") in new stack
> -- Executing [sst...@app-dnd-toggle:2] Set("SIP/117-01f6",
> "DEVICES=117") in new stack
> -- Executing [sst...@app-dnd-toggle:3] GotoIf("SIP/117-01f6",
> "0?return") in new stack
>   == Extension Changed 117[ext-local] new state Busy for Notify User 102
> -- Executing [sst...@app-dnd-toggle:4] Set("SIP/117-01f6",
> "LOOPCNT=1") in new stack
> -- Executing [sst...@app-dnd-toggle:5] Set("SIP/117-01f6",
> "ITER=1") in new stack
> -- Executing [sst...@app-dnd-toggle:6] Set("SIP/117-01f6",
> "DEVICE_STATE(Custom:DEVDND117)=BUSY") in new stack
>   == Extension Changed 117[ext-local] new state Busy for Notify User 103
>   == Extension Changed 117[ext-local] new state Busy for Notify User 114
> -- Executing [sst...@app-dnd-toggle:7] Set("SIP/117-01f6",
> "ITER=2") in new stack
> -- Executing [sst...@app-dnd-toggle:8] GotoIf("SIP/117-01f6",
> "0?begin") in new stack
> -- Executing [sst...@app-dnd-toggle:9] Return("SIP/117-01f6", "")
> in new stack
> -- Executing [...@from-internal:8] Playback("SIP/117-01f6",
> "do-not-disturb&activated") in new stack
> -- Executing [...@from-internal:9] Macro("SIP/117-01f6",
> "hangupc

Re: [asterisk-users] Re :Re: Sip module and dns (Alyed)

[Alyed]

Just for the sake of this thread I'll paste part of the last post regarding
this issue in the asterisk bug tracker.

kpfleming on 2005-03-10 post: "Essentially, what we are saying is that if
you are going to use DNS to resolve critical information in your Asterisk
configuration, you need to do everything possible to ensure that the DNS
lookups will not block for long periods of time. "

Alyed


2010/3/26 Luis Silva 

> >Just to check, have you set up
> >srvlookup=yes
> >
> >under the general context in your sip.conf?
> >
> >Alyed
>
> No, but I put it now but the result is the same. And googleing further
> https://issues.asterisk.org/view.php?id=3723, it seems that is an old
> issue...
> Don't know for witch version is, 1.2?... But is what is happening to me.
> I'm
> putting bind in the asterisk server and make some tests.
> The sip modules blocks in dns queries but if doesn't block in sip
> registrations timeout that works for me...
>
> Regards
> Luis Silva
>
> >2010/3/26 Luis Silva 
> >
> >> Hi again,
> >>
> >> In other asterisk it happened the same... No internet, no justvoip
> >> resolution, no sip...
> >> Remove the trunk, sip up... I'm going to test using bind with a "local"
> >> zone.
> >> More ideas/suggestions?
> >>
> >> Regards
> >> Luis Silva
> >>
> >>
> >> >Hi ,
> >> >
> >> >I had some problems in the past with sip trunks, asterisk-users Digest,
> >> > Vol
> >> >68, Issue 4, message 6, and  had a reply (message 9) saying that It
> could
> >> be
> >> >a dns issue.
> >> >
> >> >Well today I had a problem again with sip module and it really seams a
> dns
> >> >issue.
> >> >
> >> >I have an asterisk, version 1.4.26.1, that has 4 bri access and two sip
> >> >trunks. I'm having internet access problems and when this happens and
> if
> >> one
> >> >of the trunks tries to reregister its panic time!!! All the sip peers
> goes
> >> >unreachable, trunks and phones,  and the sip module "freezes", sip
> reload
> >> >takes many many time to act.  My solution is to remove the sip trunks
> from
> >> >the configuration.
> >> >
> >> >But why this happens? Why If there is no dns resolution of the trunks
> sip
> >> >module "freezes" ? This is more strange because if by some reason the
> >> >internet is down but still exists dns cache all is ok. (of course sip
> >> trunks
> >> >unreachable)
> >>
> >> >This is supposed to be like this? There is no dns tunning for
> >> asterisk+sip?
> >>
> >> >To avoid this I'm starting to think putting bind in the asterisk server
> >> and
> >> >publishing there the zones of the sip trunks. (Or instead of names
> start
> >> >using the ip's)
> >>
> >> >Any comments?
> >>
> >> >Regards,
> >> >Luis Silva
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] need help on setup rtp directly between 2 sip clients

[Alyed]

I guess to do what you want you need to dial directly between the phones.
Can't do it with xlite but you can with SJphones

Don't remember the exact syntax but guess it's something like
sip:usern...@the.phones.ip:5060

Alyed



2010/3/26 haloha 

> Hi all
>
> my asterisk server, 2 sip client softphones are the same LAN
>
> asterisk ip address : 192.168.1.5
> sip client 1 : 192.168.1.4
> sip client 2 : 192.168.1.2
>
> asterisk starts ok with sip
>
> setup the sip.conf
> [test]
> type=friend
> username=test
> secret=1000
> host=dynamic
> context=cucku
> directmedia=yes
> directrtpsetup=yes
>
> [1000]
> type=friend
> username=1000
> secret=1000
> host=dynamic
> context=cucku
> directmedia=yes
> directrtpsetup=yes
>
> when make call between 2 sip clients and see the debug in asterisk console
> the problem is asterisk setup the inital call for media = asterisk IP
> address, when all things done, asterisk does re-invite to setup the rtp
> directly between 2 sip clients
>
> is there any way to setup rtp directly between 2 sip clients, no need to go
> through asterisk server
>
> here is my debug log:
> <--- SIP read from UDP://192.168.1.4:18341 --->
> INVITE sip:1...@192.168.1.5  SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.4:18341
> ;branch=z9hG4bK-d8754z-da695e1f167deb68-1---d8754z-;rport
> Max-Forwards: 70
> Contact: 
> To: "1000">
> From: "Do Nguyen Ha"
> >;tag=f543a140
> Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg.
> CSeq: 2 INVITE
> Content-Type: application/sdp
> User-Agent: X-Lite release 1104o stamp 56125
> Content-Length: 261
> v=0
> o=- 8 2 IN IP4 192.168.1.4
> s=CounterPath X-Lite 3.0
> c=IN IP4 192.168.1.4
> t=0 0
> m=audio 50420 RTP/AVP 107 0 8 101
>
> <--- Transmitting (no NAT) to 192.168.1.4:18341 --->
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 192.168.1.4:18341
> ;branch=z9hG4bK-d8754z-da695e1f167deb68-1---d8754z-;received=192.168.1.4;rport=18341
> From: "Do Nguyen Ha"
> >;tag=f543a140
> To: "1000">
> Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg.
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX 1.6.0.26
> Supported: replaces, timer
> Contact: >
> Content-Length: 0
>
> Reliably Transmitting (no NAT) to 192.168.1.2:34312:
> INVITE sip:1...@192.168.1.2:34312;rinstance=862211afcf483176 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK2dfa6cce;rport
> Max-Forwards: 70
> From: "Do Nguyen Ha" 
> >;tag=as2886cf30
> To: 
> Contact: >
> Call-ID: 5fcaa0dc5dd8cac86f01164b2ea6d...@192.168.1.5
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX 1.6.0.26
> Date: Thu, 25 Mar 2010 12:15:05 GMT
> Supported: replaces, timer
> Content-Type: application/sdp
> Content-Length: 309
> v=0
> o=root 1983608375 1983608375 IN IP4 192.168.1.5
> s=Asterisk PBX 1.6.0.26
> c=IN IP4 192.168.1.5
> t=0 0
> m=audio 17580 RTP/AVP 0 3 8 101
>
> <--- SIP read from UDP://192.168.1.2:34312 --->
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK2dfa6cce;rport=5060
> Contact: 
> To: ;tag=403c255c
> From: "Do Nguyen Ha"
> >;tag=as2886cf30
> Call-ID: 5fcaa0dc5dd8cac86f01164b2ea6d...@192.168.1.5
> CSeq: 102 INVITE
> User-Agent: X-Lite release 1104o stamp 56125
> Content-Length: 0
>
>
> <--- Transmitting (no NAT) to 192.168.1.4:18341 --->
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 192.168.1.4:18341
> ;branch=z9hG4bK-d8754z-da695e1f167deb68-1---d8754z-;received=192.168.1.4;rport=18341
> From: "Do Nguyen Ha"
> >;tag=f543a140
> To: "1000">;tag=as0307d0b3
> Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg.
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX 1.6.0.26
> Supported: replaces, timer
> ontact: >
> Content-Length: 0
>
>
> <--- SIP read from UDP://192.168.1.2:34312 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK2dfa6cce;rport=5060
> Contact: 
> To: ;tag=403c255c
> From: "Do Nguyen Ha"
> >;tag=as2886cf30
> Call-ID: 5fcaa0dc5dd8cac86f01164b2ea6d...@192.168.1.5
> CSeq: 102 INVITE
> Content-Type: application/sdp
> User-Agent: X-Lite release 1104o stamp 56125
> Content-Length: 183
> v=0
> o=- 8 2 IN IP4 192.168.1.2
> s=CounterPath X-Lite 3.0
> c=IN IP4 192.168.1.2
> t=0 0
> m=audio 53062 RTP/AVP 0 8 101
>
> <->
> ACK sip:1...@192.168.1.2:34312;rinstance=862211afcf483176 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK2852e1cc;rport
> Max-Forwards: 70
> From: "Do Nguyen Ha" 
> >;tag=as2886cf30
> To: ;tag=403c255c
> Contact: >
> Call-ID: 5fcaa0d

Re: [asterisk-users] Sip module and dns

[Alyed]

Just to check, have you set up
srvlookup=yes

under the general context in your sip.conf?

Alyed


2010/3/26 Luis Silva 

> Hi again,
>
> In other asterisk it happened the same... No internet, no justvoip
> resolution, no sip...
> Remove the trunk, sip up... I'm going to test using bind with a "local"
> zone.
> More ideas/suggestions?
>
> Regards
> Luis Silva
>
>
> >Hi ,
> >
> >I had some problems in the past with sip trunks, asterisk-users Digest,
> Vol
> >68, Issue 4, message 6, and  had a reply (message 9) saying that It could
> be
> >a dns issue.
> >
> >Well today I had a problem again with sip module and it really seams a dns
> >issue.
> >
> >I have an asterisk, version 1.4.26.1, that has 4 bri access and two sip
> >trunks. I'm having internet access problems and when this happens and if
> one
> >of the trunks tries to reregister its panic time!!! All the sip peers goes
> >unreachable, trunks and phones,  and the sip module "freezes", sip reload
> >takes many many time to act.  My solution is to remove the sip trunks from
> >the configuration.
> >
> >But why this happens? Why If there is no dns resolution of the trunks sip
> >module "freezes" ? This is more strange because if by some reason the
> >internet is down but still exists dns cache all is ok. (of course sip
> trunks
> >unreachable)
>
> >This is supposed to be like this? There is no dns tunning for
> asterisk+sip?
>
> >To avoid this I'm starting to think putting bind in the asterisk server
> and
> >publishing there the zones of the sip trunks. (Or instead of names start
> >using the ip's)
>
> >Any comments?
>
> >Regards,
> >Luis Silva
>
>
> --
> _
> -- 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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Re: [asterisk-users] "Failed to play transfer sound! " during attended transfer

[Alyed]

so doesn't looks like overload

Could it be a problem with the firmware of your softphones? Have you been
using some new phones lately? someone else in a different thread pointed on
attended transfer bugs with SNOM phones.

> We are eagerly waiting for your solution.
Hope we can help but don't so much pressure on me or the listers :)

Alyed



2010/3/26 kamrun nahar bina 

> Dear sir,
>
> Thanks for your reply.
>
> our memory size is 4GB.
> concurrent calls no : 30.
> Our memory condition is below :
>
> Cpu(s):  0.3%us,  0.7%sy,  0.0%ni, 98.5%id,  0.0%wa,  0.1%hi,  0.3%si,
> 0.0%st
> Mem:   4147888k total,  3986540k used,   161348k free,76852k buffers
> Swap:  2031608k total,   56k used,  2031552k free,  3170396k cached
>
>   PID USER  PR  NI  VIRT  RES  SHR S %CPU %MEMTIME+  COMMAND
> 23160 root  15   0  440m 415m 5688 S  4.3 10.3 398:13.93 asterisk
>
> Our disk space condition is below:
> FilesystemSize  Used Avail Use% Mounted on
> /dev/mapper/VolGroup00-LogVol00
>   901G  245G  610G  29% /
> /dev/sda1  99M   18M   77M  19% /boot
> tmpfs 2.0G 0  2.0G   0% /dev/shm
>
>
> We are eagerly waiting for your solution.
>
> Thanks in advance.
>
> Nahar
>
>
>
> On Fri, Mar 26, 2010 at 2:32 PM, Alyed  wrote:
>
>> If you didn't have this problem before I'll check up for any changes
>> lately (i suppose you have done so, but ask this just to be safe)
>> I see you have lots of agents and also lots of hard disk space, so I guess
>> disk space is not an issue. Please check it anyway.
>>
>> how many concurrent calls you have? 2 GB in RAM seems little against 600
>> registered agents.
>>
>> Alyed
>>
>>
>> 2010/3/25 kamrun nahar bina 
>>
>>> Dear sir,
>>>
>>> We have been using asterisk for 4 years. Now we have got problems which
>>> occurs during the attended transfer.
>>> But we are not always getting this problem. Sometimes it happens. But now
>>> we cannot understand why this is happening?
>>>
>>> problem is:"Failed to play transfer sound! "
>>>
>>> The log of asterisk is as like as followings:
>>>
>>> [Mar 25 17:58:40] DEBUG[23168] chan_sip.c: Invalid SIP message -
>>> rejected , no callid, len 366
>>>
>>>
>>>
>>> [Mar 25 17:58:40] DEBUG[13617] audiohook.c: Write factory 0x1bfa000c was
>>> pretty quick last time, waiting for them.
>>> [Mar 25 17:58:40] DEBUG[13579] audiohook.c: Read factory 0x1bd61fe0 was
>>> pretty quick last time, waiting for them.
>>>
>>>
>>>
>>> [Mar 25 17:58:40] DEBUG[23168] chan_sip.c: Setting SIP_ALREADYGONE on
>>> dialog 5bd1acee539e699b4f5e79c94a348...@113.34.235.8
>>>
>>> [Mar 25 17:58:40] DEBUG[23168] chan_sip.c: Received bye, issuing owner
>>> hangup
>>> [Mar 25 17:58:40] DEBUG[13617] audiohook.c: Write factory 0x1bfa000c was
>>> pretty quick last time, waiting for them.
>>> [Mar 25 17:58:40] WARNING[13591] res_features.c: Failed to play transfer
>>>
>>>
>>>
>>> sound!
>>>
>>> Our system is as like as:
>>> The number of User agent is: 1650
>>> The number of Actual registered user agent is: 600
>>>
>>> Our System configuration is :
>>> IBM X3550
>>> CPU: Intel(R) Xeon(R) CPU X5460 @ 3.16GHz
>>>
>>>
>>>
>>> Memory: 2GB
>>> HDD: 3.5 SATA 1TB x 2
>>> version of asterisk: 1.4.23.1
>>>
>>> Asterisk and the User-Agent is connected through the Internet.
>>>
>>>
>>>
>>> ..And Is there any solution to solve this problem? We have investigated 
>>> in several places but we cannot find out the reason?
>>> We need this solution very urgently. We are eagerly waiting for reply.
>>>
>>> Thanks in advance
>>>
>>>
>>>
>>> Nahar
>>>
>>>
>>> --
>>>
>>> _
>>> -- 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] send a call from A to B use sip trunk prablem

[Alyed]

it doesn't seems to be a problem of communication between A and B

>-- Executing [...@macro-dialout-trunk:19]
Dial("SIP/192.168.0.151-088e7938",
"ZAP/g0/15921256331|300|M(setmusic^none)Tt") in new stack
> == Everyone is busy/congested at this time (1:0/0/1)

That's says it's more a problem with your Zap channels than your SIP
connection.

First try playing a sound in B when receiving the call, that way you can be
sure the connection is ok. If that one works then move to PSTN.

Alyed


2010/3/25 Aaron chen 

> i have a prablom here,
>
> i want to send a call from A to B use sip trunk ,
>
> the call can sended B,but not work to PSTN.
>
> the message from B server. help pls,what's rong?
>
>
>
>>
>> <--- SIP read from 192.168.0.176:5060 --->
>> INVITE sip:15921256...@192.168.0.151 SIP/2.0
>> Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK51a51b96;rport
>> From: "50005" 
>> >;tag=as72a55960
>> To: >
>> Contact: >
>> Call-ID: 28272ebb12ee6e4c1f06fca651456...@192.168.0.151
>> CSeq: 102 INVITE
>> User-Agent: Asterisk PBX
>> Max-Forwards: 70
>> Date: Fri, 26 Mar 2010 02:12:07 GMT
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
>> Supported: replaces
>> Content-Type: application/sdp
>> Content-Length: 380
>> v=0
>> o=root 15081 15081 IN IP4 192.168.0.176
>> s=session
>> c=IN IP4 192.168.0.176
>> t=0 0
>> m=audio 12726 RTP/AVP 0 18 8 3 4 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:18 G729/8000
>> a=fmtp:18 annexb=no
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:3 GSM/8000
>> a=rtpmap:4 G723/8000
>> a=fmtp:4 annexa=no
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>> a=ptime:20
>> a=sendrecv
>> <->
>> --- (14 headers 18 lines) ---
>> Sending to 192.168.0.176 : 5060 (NAT)
>> Using INVITE request as basis request -
>> 28272ebb12ee6e4c1f06fca651456...@192.168.0.151
>> Found peer 's1'
>> Found RTP audio format 0
>> Found RTP audio format 18
>> Found RTP audio format 8
>> Found RTP audio format 3
>> Found RTP audio format 4
>> Found RTP audio format 101
>> Peer audio RTP is at port 192.168.0.176:12726
>> Found audio description format PCMU for ID 0
>> Found audio description format G729 for ID 18
>> Found audio description format PCMA for ID 8
>> Found audio description format GSM for ID 3
>> Found audio description format G723 for ID 4
>> Found audio description format telephone-event for ID 101
>> Capabilities: us - 0x10f (g723|gsm|ulaw|alaw|g729), peer - audio=0x10f
>> (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10f
>> (g723|gsm|ulaw|alaw|g729)
>> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
>> (telephone-event), combined - 0x1 (telephone-event)
>> Peer audio RTP is at port 192.168.0.176:12726
>> Looking for 15921256331 in from-internal (domain 192.168.0.151)
>> list_route: hop: >
>> gd-branch*CLI>
>> <--- Transmitting (NAT) to 192.168.0.176:5060 --->
>> SIP/2.0 100 Trying
>> Via: SIP/2.0/UDP 192.168.0.176:5060
>> ;branch=z9hG4bK51a51b96;received=192.168.0.176;rport=5060
>> From: "50005" 
>> >;tag=as72a55960
>> To: >
>> Call-ID: 28272ebb12ee6e4c1f06fca651456...@192.168.0.151
>> CSeq: 102 INVITE
>> User-Agent: Asterisk PBX
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>> Supported: replaces
>> Contact: 
>> >
>> Content-Length: 0
>>
>> <>
>> -- Executing [15921256...@from-internal:1]
>> Set("SIP/192.168.0.151-088e7938", "MOHCLASS=none") in new stack
>> -- Executing [15921256...@from-internal:2]
>> Macro("SIP/192.168.0.151-088e7938", "user-callerid|SKIPTTL|") in new stack
>> -- Executing [...@macro-user-callerid:1]
>> Set("SIP/192.168.0.151-088e7938", "AMPUSER=50005") in new stack
>> -- Executing [...@macro-user-callerid:2]
>> GotoIf("SIP/192.168.0.151-088e7938", "0?report") in new stack
>> -- Executing [...@macro-user-callerid:3]
>> ExecIf("SIP/192.168.0.151-088e7938", "1|Set|REALCALLERIDNUM=50005") in new
>> stack
>> -- Executing [...@macro-user-callerid:4]
>> Set("SIP/192.168.0.151-088e7938", "AMPUSER=") in new stack
>> -- Executing [...@macro-user-callerid:5]
>> Set("SIP/192.168.0.151-088e7938", "AMPUSE

Re: [asterisk-users] "Failed to play transfer sound! " during attended transfer

[Alyed]

If you didn't have this problem before I'll check up for any changes lately
(i suppose you have done so, but ask this just to be safe)
I see you have lots of agents and also lots of hard disk space, so I guess
disk space is not an issue. Please check it anyway.

how many concurrent calls you have? 2 GB in RAM seems little against 600
registered agents.

Alyed


2010/3/25 kamrun nahar bina 

> Dear sir,
>
> We have been using asterisk for 4 years. Now we have got problems which
> occurs during the attended transfer.
> But we are not always getting this problem. Sometimes it happens. But now
> we cannot understand why this is happening?
>
> problem is:"Failed to play transfer sound! "
>
> The log of asterisk is as like as followings:
>
> [Mar 25 17:58:40] DEBUG[23168] chan_sip.c: Invalid SIP message -
> rejected , no callid, len 366
>
> [Mar 25 17:58:40] DEBUG[13617] audiohook.c: Write factory 0x1bfa000c was
> pretty quick last time, waiting for them.
> [Mar 25 17:58:40] DEBUG[13579] audiohook.c: Read factory 0x1bd61fe0 was
> pretty quick last time, waiting for them.
>
> [Mar 25 17:58:40] DEBUG[23168] chan_sip.c: Setting SIP_ALREADYGONE on
> dialog 5bd1acee539e699b4f5e79c94a348...@113.34.235.8
>
> [Mar 25 17:58:40] DEBUG[23168] chan_sip.c: Received bye, issuing owner
> hangup
> [Mar 25 17:58:40] DEBUG[13617] audiohook.c: Write factory 0x1bfa000c was
> pretty quick last time, waiting for them.
> [Mar 25 17:58:40] WARNING[13591] res_features.c: Failed to play transfer
>
> sound!
>
> Our system is as like as:
> The number of User agent is: 1650
> The number of Actual registered user agent is: 600
>
> Our System configuration is :
> IBM X3550
> CPU: Intel(R) Xeon(R) CPU X5460 @ 3.16GHz
>
> Memory: 2GB
> HDD: 3.5 SATA 1TB x 2
> version of asterisk: 1.4.23.1
>
> Asterisk and the User-Agent is connected through the Internet.
>
> ..And Is there any solution to solve this problem? We have investigated 
> in several places but we cannot find out the reason?
> We need this solution very urgently. We are eagerly waiting for reply.
>
> Thanks in advance
>
> Nahar
>
>
> --
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Re: [asterisk-users] software version

[Alyed]

Don't be so hard in him/her we all make mistakes, let's just learn from them
and move on.

Alyed


2010/3/24 Ott Rose 

>  thanks for hijacking my thread.
>
> i have an idea don't help him/her so that people will help me!
>
>
> now i am going to re-post this.
>
> > Date: Wed, 24 Mar 2010 09:08:02 -0700
> > From: asterisk@sedwards.com
> > To: asterisk-users@lists.digium.com
> > Subject: Re: [asterisk-users] software version
> >
> > On Wed, 24 Mar 2010, Amine Mrichcha wrote:
> >
> > > I do have asterisk installed for a call center and I would like to know
>
> > > if it is possible to create a scipt and execute it from a PC connected
> > > to the Network without accessing the server. This script should restart
>
> > > asterisk and another service related to aheeva.
> > >
> > > The problem now is that each time I have to access using PUTY to the
> > > server to start and run services manually.
> > >
> > > Service asterisk restart
> > >
> > > Any help would be appreciated, sorry if it is a newbie question.
> >
> > On Wed, 24 Mar 2010, Steve Edwards wrote:
> >
> > > 1) Not a -dev question. Try -users.
> > >
> > > 2) Choose a better subject. Are you asking how to start Asterisk on
> > > reboot or how to restart Asterisk because it is crashing? (If it is
> > > crashing, you should fix the problem, not apply a band-aid.)
> > >
> > > 3) Be more specific. By "access" do you mean "login" via ssh?
> >
> > 4) Don't hijack existing threads.
> >
> > --
> > Thanks in advance,
> > -
> > Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
> > Newline Fax: +1-760-731-3000
> >
> > --
> > _
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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] software version (lets try it again)

[Alyed]

If it is only a version move, say from asterisk 1.6.1 to 1.6.1.X it's
generally ok, but be careful since there are some changes that might hit you
from 1.6.0 to 1.6.2 need to have a read into the change log before changing
versions, same for the other packages you mention.

Alyed

2010/3/24 Ott Rose 

>  what is the general view about the versions of the packages that are used
> with asterisk.
>
> lame
> asterisk
> asterisk-addons
> dahdi
> libpri
>
>
>
> i like to say on a version and not upgrade due to my experience with Linux
> and upgrading screwing up things. When it comes to Asterisk i have only one
> server build under my belt and I have had issue along the way.
>
>
> What do most people do with the software versions? In general if i have a
> stable system and there is no reason like security or new needed features i
> don't upgrade. Should i do new builds with the same config (old packages)
> that i know works?
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Re: [asterisk-users] Restarting Asterisk using a script - Thanks to all -

[Alyed]

Guess it's not a matter of asterisk as it is of Linux scripting.
first check: http://linuxproblem.org/art_9.html
then try something like:
http://forums.digitalpoint.com/showthread.php?t=70926

but as Steve said, why you need to restart the asterisk service in the first
place

Fix what's wrong don't just use the band-aid


2010/3/24 Steve Edwards 

> On Wed, 24 Mar 2010, Amine Mrichcha wrote:
>
> > I do have asterisk installed for a call center and I would like to know
> > if it is possible to create a scipt and execute it from a PC connected
> > to the Network without accessing the server. This script should restart
> > asterisk and another service related to aheeva.
> >
> > The problem now is that each time I have to access using PUTY to the
> > server to start and run services manually.
> >
> > Service asterisk restart
> >
> > Any help would be appreciated, sorry if it is a newbie question.
>
> Why do you need to restart Asterisk? Is it crashing (in which case you
> should fix the underlying cause, not just put a band-aid over it) or is it
> not starting when the system is started?
>
> If you are not going to access the server, what event is going to trigger
> the execution of your script?
>
> --
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline  Fax: +1-760-731-3000
>
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Mobile phone shut down, but Queue() Ring as usual

[Alyed]

Try the same as in

http://lists.digium.com/pipermail/asterisk-users/2010-March/246316.html

just make sure to add this in the [channels] context ;)

Hope it helps.

Alyed


2010/3/23 Zhang Shukun 

> hi, all
>
> i use Queue() to call a Mobile phone, there is only one mobile phone
> in the queue. even if the mobile phone shut down, Queue() is ring in
> the cli verbose
>
> as mobile phone is normally working. what i want to see is if the
> mobile phone is shut down.
>
> queue() will end immediately to tell on one in the queue.
>
> is there any method to do this ?
>
> --
> Best regards,
> Sucan
>
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Re: [asterisk-users] Voicemail, Asterisk and Grandstream BT200

[Alyed]

you are right, under [channels] is where it's supposed to be my mistake, i
guess i was thinking in sip.conf  :)

>However, the following doubt arises to me: it would also have had this
>problem for some originating call from a telephone that is not a cell
>phone?

yes, and this can be a really serious problem if you don't fix it. So don't
forget to include this parameters from now on. I have played with them and
found setting busycount=5 is not very efficent, so leave it to 3 or 4 at
most.

Good to hear your problem is solved.

Alyed


2010/3/22 Daniel Bareiro 

> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hi, Alyed.
>
> On Mon, 22 Mar 2010, Alyed wrote:
>
> >> I was with the following situation: if I call from a cell phone, my
> >> Asterisk take the call, it presents to the caller the possibility to
> >> dialing an extension number and, in case of not doing it, it
> >> transfers this call to a specific extension.
> >>
> >> Then, if in this extension nobody takes the call, the service of
> >> voicemail is triggered so that the caller leaves its message from the
> >> cell phone. But if it hangs after to let the message without have
> >> pressed previously the pound key, the channel is taken and no longer
> >> any other call enters the PBX from the PSTN. This does not happen if
> >> the caller presses the pound key after to have left his message.
> >>
> >> As I have a box at which the cable arrives from the PSTN in which
> >> there are two ports of derivation and in one of them it leaves the
> >> cable for the Asterisk PBX (connected only then), after to have
> >> detected this problem I tried connecting in the other port an analog
> >> telephone and, indeed, it did not have tone as if never it had been
> >> hung. In addition this was confirmed because the MWI light never
> >> blinked on the telephone.
> >>
> >> After restarting the Asterisk server, yes the MWI light blinks and in
> >> addition I could corob the time in which the channel was "taken"
> >> seeing that the message lasted more than nine minutes.
> >>
> >> To what this problem can be due? It has to do the call is made
> >> specifically from cell phone through the PSTN (because if I leave a
> >> message hanging directly without pressing the pound key from an local
> >> extension, this does not happen)? There is some form to avoid it?
>
> > Make sure you have
> > busydetect=yes
> > busycount=3
> >
> > somewhere below your [general] context in chan_dahdi.conf (or
> > zapata.conf depending on your asterisk version) and restart the the
> > service.
> >
> > This should be enoough to do the magic.
>
> It didn't have configured these two parameters so I added now them but
> in the [channels] context since I don't have a [general] context (It
> does not sound to me that in the file by default generated by Asterisk
> there would not be it either, although I can be mistaken).
>
> Beyond that, with these two parameters, I no longer have the problem
> mentioned before. Thanks!
>
> However, the following doubt arises to me: it would also have had this
> problem for some originating call from a telephone that is not a cell
> phone?
>
> Thanks for your reply.
>
> Regards,
> Daniel
>
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v1.4.9 (GNU/Linux)
>
> iEYEARECAAYFAkunNjQACgkQZpa/GxTmHTfAbACfT8PVkcp/xESdqsiczg3YY/Dd
> FGcAn1TdOqiZaKAjLg4h3SDt/34A4bKX
> =37qZ
> -END PGP SIGNATURE-
>
>
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> _
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Re: [asterisk-users] Voicemail, Asterisk and Grandstream BT200

[Alyed]

Make sure you have
busydetect=yes
busycount=3

somewhere below your [general] context in chan_dahdi.conf (or zapata.conf
depending on your asterisk version) and restart the the service.

This should be enoough to do the magic.

Alyed


2010/3/21 Daniel Bareiro 

> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hi, Gordon.
>
> On Sun, 21 Mar 2010, Gordon Henderson wrote:
>
> >> I'm testing with a Grandstream BT200 telephone and, according to I
> >> read, it has a LED that blinks if for that extension messages were
> >> left.
> >>
> >> In "Voice Mail UserID", under the "ACCOUNT" tab, I put *100 that is
> >> the extension in which my Asterisk answer the voicemail service and
> >> if then I press MESSAGE button, the telephone communicates with
> >> Asterisk and, after to introduce the password, it indicates to me
> >> that I have messages. But the luminous indicator does not work.
> >>
> >> It is necessary to configure something special for this? It can be
> >> that it doesn't work because there is to introduce one password
> >> previously?
>
> > There's another setting in the phone you need to set "SUBSCRIBE for
> > MWI".
>
> Yes. I was needing to indicate the use of MWI of the side of the
> configuration of the telephone. I selected the "SUBSCRIBES for MWI"
> checkbox.
>
> > And make-sure the mailbox number is listed in the sip.conf entry for
> > that phone.
>
> According to which I was reading, the MWI notifications become by the
> option "mailbox=" in the configuration of the extension. For this
> extension, the 104, had "mailbox=104" but still with MWI enabled option,
> it was not working. After to think enough on this subject, I have
> noticed that instead of 104 I had to put 1...@voicemail since "voicemail"
> it was context that I'm using in voicemail.conf.
>
> With this already was working.
>
> However, beyond this, I was with the following situation: if I call from
> a cell phone, my Asterisk take the call, it presents to the caller the
> possibility to dialing an extension number and, in case of not doing it,
> it transfers this call to a specific extension.
>
> Then, if in this extension nobody takes the call, the service of
> voicemail is triggered so that the caller leaves its message from the
> cell phone. But if it hangs after to let the message without have
> pressed previously the pound key, the channel is taken and no longer any
> other call enters the PBX from the PSTN. This does not happen if the
> caller presses the pound key after to have left his message.
>
> As I have a box at which the cable arrives from the PSTN in which there
> are two ports of derivation and in one of them it leaves the cable for
> the Asterisk PBX (connected only then), after to have detected this
> problem I tried connecting in the other port an analog telephone and,
> indeed, it did not have tone as if never it had been hung. In addition
> this was confirmed because the MWI light never blinked on the telephone.
>
> After restarting the Asterisk server, yes the MWI light blinks and in
> addition I could corob the time in which the channel was "taken" seeing
> that the message lasted more than nine minutes.
>
> To what this problem can be due? It has to do the call is made
> specifically from cell phone through the PSTN (because if I leave a
> message hanging directly without pressing the pound key from an local
> extension, this does not happen)? There is some form to avoid it?
>
> Thanks for your reply!
>
> Regards,
> Daniel
>
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v1.4.9 (GNU/Linux)
>
> iEYEARECAAYFAkums0oACgkQZpa/GxTmHTcGpQCghJvfphxc5ZzZhouryA+OlwGm
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> =Hskd
> -END PGP SIGNATURE-
>
>
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re: [asterisk-users] Asterisk outbound calling does not wait for answer before playback

[Alyed Tzompa]

Had the same issue time ago, but Eric shed good light on it, 
have a look at:

http://lists.digium.com/pipermail/asterisk-users/2006-November/172079.html

Summary: sorry, no nice work around.

Alyed  


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Hello Asteriskers, :-)

We're trying to set up an outbound notification calling for system 
alerts with Asterisk 1.4.0.  We generate a call file in 
/var/spool/asterisk/outgoing and the outbound call is originated through 
Zap/1 (Sangoma A200D to a Canadian POTS line).  The problem is that 
Asterisk does not wait for the other side to answer before it starts 
playing the message.  So the person called answers the phone after the 
second or third ring and only hears the tail end of the message and the 
"goodbye".

Ideally, we want to deliver the message immediately after the person 
answers, or if an answering machine picks up, right after the "beep".

Any suggestions?

(1) The call file generator script (works ok):
#!/bin/sh

TMPFILE=`mktemp /tmp/tmp.XXX` || exit 1
echo "TMPFILE = $TMPFILE"

cat < $TMPFILE
Channel: Zap/g1/phone_number_here
Callerid: SYSTEM
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: dialout
Extension: s
Priority: 1
EOT

mv -v $TMPFILE /var/spool/asterisk/outgoing

(2) The dialout context in extensions.conf (problem - starts playback 
before call is answered)
[dialout]
exten => s,1,NoOp(Dialout)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=8)
exten => s,n,Set(MACHINE=0)
exten => s,n,Answer
exten => s,n,BackgroundDetect(silence/5,1000,50)
exten => s,n,NoOp(Ans Machine detected)
exten => s,n,Set(MACHINE=1)
exten => s,n,BackgroundDetect(silence/30,1000,50,30050)
exten => s,n,NoOp(Ans Machine Message Too Long)
exten => s,n,Hangup

exten => talk,1,GotoIf($[${MACHINE}=1]?machine:human)
exten => talk,2(machine),Goto(dialout-machine,s,1)
exten => talk,3(human),Goto(dialout-human,s,1)

[dialout-machine]
exten => s,1,NoOp(Dialout to Ans Machine)
exten => s,n,Playback(/tmp/asterisk-recording)
exten => s,n,Wait(1)
; we'd like to do something to wait for the "beep" here...
exten => s,n,Playback(vm-goodbye)
exten => s,n,Hangup

[dialout-human]
exten => s,1,NoOp(Dialout to Human)
exten => s,n,Playback(/tmp/asterisk-recording)
exten => s,n,Wait(1)
exten => s,n,Playback(vm-goodbye)
exten => s,n,Hangup

(3) *CLI>
-- Attempting call on Zap/1/1234567 for [EMAIL PROTECTED]:1 (Retry 1)
   > Channel Zap/1-1 was answered.
-- Executing [EMAIL PROTECTED]:1] NoOp("Zap/1-1", "Dialout") in new stack
-- Executing [EMAIL PROTECTED]:2] Set("Zap/1-1", "TIMEOUT(digit)=5") in new 
stack
-- Digit timeout set to 5
-- Executing [EMAIL PROTECTED]:3] Set("Zap/1-1", "TIMEOUT(response)=8") in 
new stack
-- Response timeout set to 8
-- Executing [EMAIL PROTECTED]:4] Set("Zap/1-1", "MACHINE=0") in new stack
-- Executing [EMAIL PROTECTED]:5] Answer("Zap/1-1", "") in new stack
(Problem: Asterisk does not wait until the call is answered on the far end!)
-- Executing [EMAIL PROTECTED]:6] BackgroundDetect("Zap/1-1", 
"silence/5|1000|50") in new stack
-- Playing 'silence/5' (language 'en')
-- Executing [EMAIL PROTECTED]:1] GotoIf("Zap/1-1", "0?machine:human") 
in new stack
-- Goto (dialout,talk,3)
-- Executing [EMAIL PROTECTED]:3] Goto("Zap/1-1", "dialout-human|s|1") 
in new stack
-- Goto (dialout-human,s,1)
-- Executing [EMAIL PROTECTED]:1] NoOp("Zap/1-1", "Dialout to Human") 
in new stack
-- Executing [EMAIL PROTECTED]:2] 
Playback("Zap/1-1","/tmp/asterisk-recording") in new stack
-- Playing '/tmp/asterisk-recording' (language 'en')
-- Executing [EMAIL PROTECTED]:3] Wait("Zap/1-1", "1") in new stack
-- Executing [EMAIL PROTECTED]:4] Playback("Zap/1-1", "vm-goodbye") 
in new stack
-- Playing 'vm-goodbye' (language 'en')
-- Executing [EMAIL PROTECTED]:5] Hangup("Zap/1-1", "") in new stack
  == Spawn extension (dialout-human, s, 5) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
[Feb  8 13:29:37] NOTICE[32512]: pbx_spool.c:351 attempt_thread: Call 
completed to Zap/1/1234567

Thanks for any ideas on this!

Alvin

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re: [asterisk-users] Automatic Dial, Play message

[Alyed Tzompa]

I've made a very simple one time ago I
could share with you, it's made on bash, takes as input a CSV file,
places the calls using the /var/spool/asterisk/outbound directory, and
restricts the number of calls to a given number at a time (say 10)

I can share it with you only if you promise NOT to use it for
Telemarketing, otherwise might the mighty spirits of (place the name of
some super natural power here) cast the most terrible spell on you
forever and ever.

so, if you still want it just contact me directly :)

Alyed  


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Does anyone have some method, or AGI scripts that will automatically
call a list of numbers from a database and play a pre-recorded
message?

Just for example, you have a database of

FirstName, LastName, PhoneNumber
Jon, Beck, 9194713175

So it would pull each record with phone number, dial the number, when
answered play a pre-recorded message.

It could be used to notify parents at a school that a after school
game is canceled.

I appreciate any direction you can point me in.

--
***
Forrest Beck
IAXTEL: 17002871718
[EMAIL PROTECTED]

-- 
***
Forrest Beck
IAXTEL: 17002871718
[EMAIL PROTECTED]
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re: [asterisk-users] error when compiling zaptel-1.4

[Alyed Tzompa]

The error lies here:

>make[2]: Entering directory `/usr/src/kernel-headers-2.4.27-3-386'

>make: *** arch/i386/boot: No such file or directory.  Stop.

do you have the kernel-headers installed? (e.g.  
glibc-kernheaders-2.4-9.1.87.i386.rpm for Fedora) 

Alyed 


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when i compile zaptel
make linux26
make install
i got these errors:

make[1]: Leaving directory `/usr/src/zaptel-1.4/wct4xxp'
make -C datamods clean
make[1]: Entering directory `/usr/src/zaptel-1.4/datamods'
make -C /lib/modules/2.4.27-3-386/build
SUBDIRS=/usr/src/zaptel-1.4/datamods clean
make[2]: Entering directory `/usr/src/kernel-headers-2.4.27-3-386'
make: *** arch/i386/boot: No such file or directory.  Stop.
make: Entering an unknown directorymake: Leaving an unknown
directorymake[2]: *** [archclean] Error 2
make[2]: Leaving directory `/usr/src/kernel-headers-2.4.27-3-386'
make[1]: *** [clean] Error 2
make[1]: Leaving directory `/usr/src/zaptel-1.4/datamods'
make: *** [clean] Error 2

any idea
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Re: [asterisk-users] Sangoma A101 with Unicall

[Alyed Tzompa]

Carlos: Had you tried re-compiling wanpipe?

Had a similar problem, and eventhough I'm pretty sure I compiled
wanpipe and it did the zaptel patch succesfully, once I finished with
the Unicall installation, somehow the patch was not wotking correctly.
So after a couple of days of looking around I decided to re-compile
wanpipe (run  ./Setup install where you untared the
wanpipe-X.X.tar.gz) and afterwards it worked fine.

if it doesn't try contacting me directly, maybe I can have an eye on your 
system and help you out.

Alyed  


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On Thu, 21 Dec 2006 08:33:27 -0500, Doug Lytle wrote
> Carlos Chavez wrote:
> > CAS signalling on span 1 conflicts with HDLC with FCS check on channel
> >
> 
> My guess is not to use HDLC, as the error says above, that it 
> conflicts with CAS.
> 

 I wish it were that easy and obvious.  I only found one setting on the
configuration to disable HDLC and it is disabled.

--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001

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[asterisk-users] mfcr/R2

[Alyed Tzompa]

Hello!

I'm tryuing to bring up an R2 connection but eventhough I've followed
the guidelines in: http://zarzamora.com.mx/asterisk/17 something seems
to be missing.

When an incomming call is generated I get:

Nov 24 06:01:17 WARNING[-197416016]: chan_unicall.c:612 unicall_report:
MFC/R2 UniCall/24  <- 0001 
[1/  
1/Idle 
/Idle]

Nov 24 06:01:17 WARNING[-197416016]: chan_unicall.c:612 unicall_report: MFC/R2 
UniCall/24 Detected

Nov 24 06:01:17 WARNING[-197416016]: chan_unicall.c:612 unicall_report: MFC/R2 
UniCall/24 Making a new call with CRN 32771

Nov 24 06:01:17 WARNING[-197416016]: chan_unicall.c:612 unicall_report:
MFC/R2 UniCall/24 1101  -> 
[2/  
2/Idle 
/Idle]

Nov 24 06:01:17 WARNING[-197416016]: chan_unicall.c:2672 handle_uc_event: 
Unicall/24 event Detected

and that's it, afterwards just a busy tone and the telco guy says the channel 
turns "sealed".

When I try an outbound call I get:

-- Attempting call on Unicall/g2/12345678 for [EMAIL PROTECTED]:1 (Retry 1)

Nov 24 05:53:52 WARNING[-286483536]: chan_unicall.c:612 unicall_report: MFC/R2 
UniCall/24 Call control(1)

Nov 24 05:53:52 WARNING[-286483536]: chan_unicall.c:612 unicall_report: MFC/R2 
UniCall/24 Make call

Nov 24 05:53:52 WARNING[-286483536]: chan_unicall.c:612 unicall_report: MFC/R2 
UniCall/24 Making a new call with CRN 32769

Nov 24 05:53:52 WARNING[-286483536]: chan_unicall.c:612 unicall_report:
MFC/R2 UniCall/24 0001  -> 
[1/  
1/Idle 
/Idle]

Nov 24 05:53:52 WARNING[-286483536]: chan_unicall.c:2672 handle_uc_event: 
Unicall/24 event Dialing

Nov 24 05:53:54 WARNING[-286483536]: chan_unicall.c:612 unicall_report: MFC/R2 
UniCall/24 seize_ack_wait_expired

Nov 24 05:53:54 WARNING[-286483536]: chan_unicall.c:612 unicall_report:
MFC/R2 UniCall/24 R2 prot. err. [1/ 
40/Seize
/Idle] cause 32776 - Seize ack timed out

Nov 24 05:53:54 WARNING[-286483536]: chan_unicall.c:612 unicall_report:
MFC/R2 UniCall/24 1001  -> 
[1/  
1/Idle 
/Idle]

Nov 24 05:53:54 WARNING[-286483536]: chan_unicall.c:2672 handle_uc_event: 
Unicall/24 event Protocol failure

-- Unicall/24 protocol error. Cause 32776

Nov 24 05:53:54 WARNING[-286483536]: chan_unicall.c:612 unicall_report: MFC/R2 
UniCall/24 Channel echo cancel

Nov 24 05:53:54 WARNING[-286483536]: chan_unicall.c:612 unicall_report: MFC/R2 
UniCall/24 Channel gains

Nov 24 05:53:54 WARNING[-286483536]: chan_unicall.c:612 unicall_report: MFC/R2 
UniCall/24 Channel switching

-- Hungup 'UniCall/24-1'

Nov 24 05:53:54 NOTICE[-286483536]: pbx_spool.c:234 attempt_thread: Call failed 
to go through, reason 1

Something else funny is happening: as soon as the telco makes a reset
of the trunk they say they start receving data as if the PBX would be
generating calls, then all channels go "sealed" except for 1.

Anyone having an idea on how to solve this?

Alyed


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re: [asterisk-users] Asterisk 1.2.13 can't load module app_curl.so

[Alyed Tzompa]

First look if you have the libidn.so.11 library. if you don't 
then install it, otherwise you can simply copy-paste it into the /usr/lib 
folder where Asterisk is looking for it or make a symbolic link to it.

Alyed 



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When i start asterisk : 
asterisk -vvvc

[app_curl.so]Nov 16 15:23:57 
WARNING[4469]: loader.c:325 __load_resource: /usr/lib/libidn.so.11: undefined 
symbol: stringprep_rfc3454_A_1
Nov 16 15:23:57 WARNING[4469]: loader.c:554 load_modules: Loading module 
app_curl.so failed!

Do you have any 
insght ?

My setup :

Fedora Core 6
Asterisk 1.2.13


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[asterisk-users] Harris picking up before extension

[Alyed Tzompa]

		Hi there!
I'm setting up a connection between Asterisk ver. 1.2.13 and a Harris
20-20 PBX. More less everything went fine, but the problem I have now
is when dialing to the Harris PBX, it seems to pick up my call as
soon as it reaches it.
For example if from the Asterisk outgoing folder I dial an extension,
say 100, and play a prompt as soon as it is picked up, the promt is
beign played as soon as it reaches the Harris, eventhough the given
extension can still be ringing. If I let the extension ring for a while
and then pick up I only hear the prompt in the middle (or as far as it
went
till I picked up).
Have tried kewlstart, loopstart, groundstart and even the
answeronpolarityswitch configs in zapata.conf but can't find the
solution.
Any one having solved this problem?Alyed
		

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[asterisk-users] Harris 20-20

[Alyed Tzompa]

		Hi there!
I'm setting up a connection between Asterisk ver. 1.2.13 and a Harris
20-20 PBX. More less everything went fine, but the problem I have now
is that when dialing to the Harris PBX it seems to pick up my call as
soon as it reaches it.
For example if from the Asterisk outgoing folder I dial an extension,
say 100, and play it a prompt as soon as it is picked up, the promt is
beign played as soon as it reaches the Harris, eventhough the given
extension can still be ringing. If I let it ring and then pick up this
extension I only hear the prompt in the middle (or as far as it went
till I picked up).
Have tried kewlstart, loopstart, groundstart and even the
answeronpolarityswitch configs in zapata.conf but can't find the
solution.
Any one having solved this problem?Alyed
		

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re: [asterisk-users] connect Sipura with Asterisk - both behind NAT

[Alyed Tzompa]

		If your Asterisk is behind a NAT you can use externip=x.x.x.x (sip.conf)
If your Sipura is behind a NAT you can use nat= yes (sip.conf)
		Btu I'm really afraid that unless you use a SIP
proxy (e.g. Portaone) you won't be able to succesfully connect both
elements if they both are behind NATs. It is just because of how SIP and NAT work together.
		
		
Alyed 
		
		
		
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		Does anybody have a good link how to connect Sipura with Asteriks, bothbehind NAT?  I'm using FWD but their connection is like a weather(especially IAX), I need something more reliable. I was thinking of using stun and/or proxy but can not find any good linkexplaining how to setup Linux server-- #Joseph___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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re: [asterisk-users] Good phones for outside of the office?

[Alyed Tzompa]

		Isn't your problem more about NAT traversal rather than the phones themselves?if so better use some iax softphone, have a look at: http://www.voip-info.org/wiki-VOIP+Phonesof course you can use SIP based hard/soft phones but using iax based ones is cheaper and faster.Alyed
		
		

Return-Path: <[EMAIL PROTECTED]> Mon Oct 30 12:30:21 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by maila11.webcontrolcenter.com with SMTP;Mon, 30 Oct 2006 12:30:21 -0700
		I am looking for phones that work well (or at all) when outside of the network and behind a router, such as at someone's home or in a hotel. My Polycom IP601s do not seem to be up to the task, so I am hoping that there is a good alternative for my outside sales people to use to talk to my asterisk server.Thanks in advance,Warren___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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re: [asterisk-users] Live creation of trunk groups

[Alyed Tzompa]

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

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		Hi,Is there a way to create trunk groups while asterisk is running.For exemple let's say that zapata.conf defines g0 as channels 1-23I would like (while asterisk is running) define g1 as 1-10 and g1 as 10-23Any hints appreciated.Andre Courchesne___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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re: [asterisk-users] Re: ECHO Cancellation in SIP Calls

[Alyed Tzompa]

		Echo is generated by the analog end to where you place the call, not the IP side of it.
As far as I know the echo cancelation in the Asterisk can only be tweaked in the zapata.conf (since IP calls don't generate it)
I'm afraid there is little you can do to here.Alyed 
		
		
		
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		On 2006-10-26 03:18:20 -0700, Stefan Agethen  said:> Hi,> > i am from Germany, so excuse my School English.> > I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update > of Asterisk 2 wooks ago, Echos accure in my SIP Calls.> > I use SNOM 360, sometimes there is no echo (for example if i call > myself via SIP->Asterisk->SIPProvider->TELEKOM->ISDN)> but if i call other people there occures Echo many times. The Routing > is always the same :> SIP (SNOM) -> Asterisk -> VoIPProvider -> ISDN/POTS> > Can i control the cancellation with the zapata.conf ?> > I have tried this with "echocancel=..." and so on, no luck :(> > I would be glad to get some help, the Docs of Asterisk dont explain how > to cancel Echos in ! SIP !Are you hearing the echo, or is the far end party?___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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re: [asterisk-users] Re: duplicate "ghost" calls with long duration

[Alyed Tzompa]

		you can also try using
busydetect=yes
busycount=4
in your zapata.conf
Hopefuly you won't start getting sudden hang ups, due to false positives and it will be helpful enough.
Alyed 
		
		
		
Return-Path: <[EMAIL PROTECTED]> Tue Oct 17 14:30:11 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by maila11.webcontrolcenter.com with SMTP;   Tue, 17 Oct 2006 14:30:11 -0700Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1])	by lists.digium.com (Postfix) with ESMTP id C3E341FCA42;
		
		On 2006-10-17 09:00:51 -0700, Bjoern Metzdorf  said:>> I run into that from time to time for this business account we have >> where channels were staying open for a long time so I made a script run >> from cron to hang up any extension over X amount of time:>> >> /usr/sbin/asterisk -rx "show channels concise" |awk -F : '($11 > 5400) >> {print "/usr/sbin/asterisk -rx \"soft hangup " $1 "\""} '|sh>> >> This looks at any calls over 90 minutes then hangs it up. You can >> modify it for your issue say something like:>> >> /usr/sbin/asterisk -rx "show channels concise" |awk -F : >> '/YOUR_X_SIPURA_NUMBER/'|awk -F : '($11 > 5400) {print >> "/usr/sbin/asterisk -rx \"soft hangup " $1 "\""} '|sh>> >> Not practical though for saving money... If someone is on for say 1 >> minute and there is an issue with the channel not hanging up, 5399 >> minutes would still be billed.> > What version are you using?> > I never had these issues with asterisk 1.0.x in 15 months. That leads > me to a problematic 1.2.x or to faulty bristuff-patches.> > I will upgrade asterisk asap to latest 1.2.x and add an absolute > timeout to those destinations.> > But: Are we the only ones experiencing this?That really doesn't sound at all the same to me as what he is describing?  In his case (ie not hung up calls) if you are using SIP handsets, then the rtptimeout setting can cut the calls off when there is no audio data flowing.Good Luck,Marty___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] Cisco CCM - Asterisk

[Alyed Tzompa]

		Hi!
You were right! changing the Calling Search Space for the SIP trunk worked!.
Many thanks for your help Eric. Now I'm fine tuning the application, but the communication with Cisco works pretty fione so far.Alyed 
		
		
		
Return-Path: <[EMAIL PROTECTED]> Wed Oct 11 10:57:00 2006Received: from rwcrmhc15.comcast.net [204.127.192.85] by maila11.webcontrolcenter.com with SMTP;   Wed, 11 Oct 2006 10:57:00 -0700
		
		
		I
think SIP is working on your CM becuase it sends a 404 message, I think
your problem is the CM cant find the endpoint in its database.I
am running CM 4.1(3)sr1, and under where you enter the device name
there is a section called, "Call Routing Information". There you should
see a field called "Calling Search Space".You should select the
Calling Search space that the destination endpoints are in. Or if you
want to create a new calling search space, and then give them access to
each other you can do that.In the attachment, I circled the calling search space field I see on my Add NEW SIP TRUNK PAGE.Hope this helps. -- Original message ------From: "Alyed Tzompa" > > 		Many thanks for your interest :)> > In the Cisco the only thing I've donde so far is enable a SIP trunk and pointing > it towards the IP the Asterisk has.> > For this I mean:> > Device --> Trunk --> Add a new Trunk> > Seleccted for Trunk Type "SIP trunk"> > Selected for Device Protocol "SIP"> > Entered for  > 		Device Name "asterisk"> > 		Entered for  > 		Description "asterisk"> > Selected the appropriate device for Device Pool> > Entered for Destination Address "192.168.1.20" (Asterisk's IP)> > Selected for Outgoing Transport Type "UDP"> > Entered the appropriate routing patterns> > clicked on "Insert"> > in sip.conf I'm using the same config shown in th VoIP-info wiki:> > [cisco]> > type=friend> > context=cisco-test> > host=192.168.1.100> > disallow=all> > allow=ulaw> > allow=alaw> > nat=no> > canreinvite=yes> > qualify=yes > > Regards,> > Alyed  > > > Return-Path:  Tue Oct 10 16:24:58 2006> Received: from rwcrmhc13.comcast.net [216.148.227.153] by > maila11.webcontrolcenter.com with SMTP;>Tue, 10 Oct 2006 16:24:58 -0700> > I can check for you tomorrow and send you screen shots.> > How do you have the asterisk configured on the CM. Please send me some > configuration information.> > -Eric>  -- Original message --> From: "Alyed Tzompa" > > > > 		Do you know where in the Cisco menu can I> > check out this "calling search spaces" you point out? I don't have> > access to it right now so want to know if this is a configurable menu> > in the web interface or will need to ask for expert (and expensive)> > Cisco assistance.> > > > Alyed  > > > > > > Return-Path:  Tue Oct 10 11:04:09 2006> > Received: from rwcrmhc11.comcast.net [216.148.227.151] by > > maila11.webcontrolcenter.com with SMTP;> >Tue, 10 Oct 2006 11:04:09 -0700> > > > Looks> > like the CallManager is unable to find the endpoint in its database.> > Make sure asterisk trunk on the Call manager is in the same "calling> > Search Space" as the phones are in, or make sure there is access> > between the "calling search spaces"> > > > -Eric> > > >  -- Original message --> > From: "Alyed Tzompa" > > > > > > 		Hi!> > > > > > I'm trying to communicate a Cisco CCM 4.0 with Asterisk 1.2.11, I 've > followed > > > the info in > > > > > > > > > http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Integrat> > > ion > > > > > > but still not able to make Asterisk communicate with Cisco. I keep on > > receiving > > > ---  > > > 		SIP/2.0 400 Bad Request - 'Malformed/Missing URL' > > > 		--- and ---   > > > > > > SIP/2.0 404 Not Found  ---  > > > 		messages everytime I send a call. Had play a lot with the way > > > SIP messages are sent to the Cisco, but always been unseccessful. > > > > > > I'm begining to think this is more of a Cisco config problem than> > > Asterisk, has anyone had a similar problem??? I'm not a Cisco expert so> > > dun't know if I need to "enable" SIP messageing/reception in the Cisco.> > > > > > Regards,> > > > > > Alyed  > > > > > > > > > > > > > > > > > > 
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Re: [asterisk-users] Cisco CCM - Asterisk

[Alyed Tzompa]

		What I want is to transfer some calls to a Cisco extension, so think I don't need to do the upgrade to CM5.I'm I right?AlyedOn Tue, Oct 10, 2006 at 01:21:28PM -0500, Lacy Moore - Aspendora wrote:> I'm begining to think this is more of a Cisco config problem than Asterisk,> >has anyone had a similar problem??? I'm not a Cisco expert so dun't know if> >I need to "enable" SIP messageing/reception in the Cisco.>>> Yeah, pretty sure to enable it you're going to have to upgrade to CCM 5.  I> could be wrong, that has been known to happen, once.SIP trunks are supported in CM4.2 and CME 3.4 and 4.0See the data sheet for CM4.2 here:http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_data_sheet0900aecd8042402c.htmlYou're probably thinking of SIP endpoint functionality, which was addedin CM5.--Phil
		
		

		

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Re: [asterisk-users] Cisco CCM - Asterisk

[Alyed Tzompa]

		Do you know where in the Cisco menu can I
check out this "calling search spaces" you point out? I don't have
access to it right now so want to know if this is a configurable menu
in the web interface or will need to ask for expert (and expensive)
Cisco assistance.Alyed 
		
		
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		Looks
like the CallManager is unable to find the endpoint in its database.
Make sure asterisk trunk on the Call manager is in the same "calling
Search Space" as the phones are in, or make sure there is access
between the "calling search spaces"-Eric -- Original message --From: "Alyed Tzompa" > > 		Hi!> > I'm trying to communicate a Cisco CCM 4.0 with Asterisk 1.2.11, I 've followed > the info in > > http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Integrat> ion > > but still not able to make Asterisk communicate with Cisco. I keep on receiving > ---  > 		SIP/2.0 400 Bad Request - 'Malformed/Missing URL' > 		--- and ---   > > SIP/2.0 404 Not Found  ---  > 		messages everytime I send a call. Had play a lot with the way > SIP messages are sent to the Cisco, but always been unseccessful. > > I'm begining to think this is more of a Cisco config problem than> Asterisk, has anyone had a similar problem??? I'm not a Cisco expert so> dun't know if I need to "enable" SIP messageing/reception in the Cisco.> > Regards,> > Alyed  > > > 
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[asterisk-users] Cisco CCM - Asterisk

[Alyed Tzompa]

		Hi!
I'm trying to communicate a Cisco CCM 4.0 with Asterisk 1.2.11, I 've followed the info in 
http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Integration 
but still not able to make Asterisk communicate with Cisco. I keep on receiving --- 
		SIP/2.0 400 Bad Request - 'Malformed/Missing URL'
		--- and ---  
		
SIP/2.0 404 Not Found --- 
		messages everytime I send a call. Had play a lot with the way SIP messages are sent to the Cisco, but always been unseccessful. 
I'm begining to think this is more of a Cisco config problem than
Asterisk, has anyone had a similar problem??? I'm not a Cisco expert so
dun't know if I need to "enable" SIP messageing/reception in the Cisco.
Regards,
		
		
		
Alyed 
		

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Re: [asterisk-users] Spurious hangups on zaptel interface

[Alyed Tzompa]

		I'm curious... why will this work??
busydetect will just cut the line if there are 4 tones (les or more
depending the busycount param), and call progress will in fact try not
to cut the call due to false hangups.Alyed
		
		
		
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		Barry D. Hassler wrote:> We seem to be getting unexpected hangups on our * system, very> consistent when calling particular numbers that we can associate with a> clients phone system. These hangups generally occur when our call is> transferred within their system (to voicemail usually). > > I'm suspecting their may be some sort of "flash" (for lack of a better> term) on the called side, but I can't verify this. > > the situation does appear to be consistent and reproducible, but only> with specific phone systems that our calls go through.> > Has anyone else experienced this, or have any potential resolutions?> I've researched this quite a bit, but not turning up anything> particularly relevant.> > I am using asterisk 1.2.9.1Remove busydetect=yes and callprogress=yes from your /etc/asterisk/zapata.conf___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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re: [asterisk-users] Spurious hangups on zaptel interface

[Alyed Tzompa]

		I'm experiencing the same problems, but
unfortunatelly haven't been able to associate them with any number
since they appear to be random. But maybe we can do a little research
about it, and hopefully find teh solution for both:
are your PSTN lines POTS or E1/T1? can you make a couple of calls and
post here the logs? would be nice if you can enable the full Asterisk
log for a single call and post that one.Alyed 
		
		
		
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We seem to be getting unexpected hangups on our * system, very
consistent when calling particular numbers that we can associate with a
clients phone system. These hangups generally occur when our call is
transferred within their system (to voicemail usually). 
I'm suspecting their may be some sort of "flash" (for lack of a better term) on the called side, but I can't verify this. 
the situation does appear to be consistent and reproducible, but only with specific phone systems that our calls go through.
Has anyone else experienced this, or have any potential resolutions?
I've researched this quite a bit, but not turning up anything
particularly relevant.
I am using asterisk 1.2.9.1Barry D. HasslerPresidentHCST
2332 Grange Hall Road
Beavercreek, Ohio 45431-2345
http://www.hcst.net/

 
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FWD: 3934279000 (655480) 
HCST*Net Support Issues: please email [EMAIL PROTECTED]Billing Issues: Please email [EMAIL PROTECTED]
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Re: [asterisk-users] Are you using app_meetme or app_conference

[Alyed Tzompa]

		Be careful when using heavily ChanSpy. We
did couple of weeks ago and the result was having Asterisk crashing
almost once every day. How heavy? around 4 people using it 8 hours a
day, each one using ChanSpy every 3-5 mins.
we were not able to find the exact reason, so just stop using it.Alyed 
		
		
		
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		On 15:27, Wed 27 Sep 06, Erick Perez wrote:> Hi, for call centers with voip phones and calls coming in via SIP and> Zap, what app_ are people using to do:We use SIP and IAX2 and SCCP (chan_sccp). Zap is notpossible for us because we want to run it on OpenBSD and thezaptel is not ported to it yet.> -conference> -listening to conversation of agentsChanSpy works for this.> > Is app_meetme or app_conference?We use app_conference. We have to since there's no timerzaptel stuff for OpenBSD.> > Does app_meetme still suffers from the need to transcode to slin?I have no idea.-- Michiel van Baak[EMAIL PROTECTED]http://michiel.vanbaak.euGnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD"Why is it drug addicts and computer afficionados are both called users?"___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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[asterisk-users] H323 IP phones

[Alyed Tzompa]

		Hi guys!Can someone give advice on nice H323 IP phones brands?? I'm looking for some H323 IP phones for a customer. Diving in theinternet found the Uniden - TVUNIDEN_UIP300, but haven't ever heard about them. Can someone give feedback experince about it??, configease, sound quality, visual appearance, end-user feedback, any infowill be appreciated.thnx!Alyed

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[asterisk-users] fw: Uniden - TVUNIDEN_UIP300

[Alyed Tzompa]

		Hi guys!
		Can someone give advice on nice H323 IP phones brands?
		?? 
I'm looking for some H323 IP phones for a customer. Diving in the
internet found the Uniden - TVUNIDEN_UIP300, but haven't ever heard 
about them. Can someone give feedback experince about it??, config
ease, sound quality, visual appearance, end-user feedback, any info
will be appreciated.
thnx!
		
Alyed
		

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Re: [asterisk-users] g729 and polycoms problem

[Alyed Tzompa]

		Sorry but I've ran out of ideas...Anyone else out there with a successful Polycom g729 pass through-only experience?Alyed
		
		
		

Return-Path: <[EMAIL PROTECTED]> Thu Sep 21 11:27:21 2006Received: from nz-out-0102.google.com [64.233.162.206] by maila11.webcontrolcenter.com with SMTP;Thu, 21 Sep 2006 11:27:21 -0700Received: by nz-out-0102.google.com with SMTP id z6so390195nzd
		didn't work :(Regards,SantiagoOn 9/20/06, Alyed Tzompa <[EMAIL PROTECTED]>wrote:> Not an expert at reading Polycom config files, but guess g729 and ulaw are> both preference 1 isn't it?>> hey... you have in your sip.conf configuration "canreinvite=no"... think> this may be a problem: since Asterisk will always stay in the path of the> RTPs, I think it might need to have the proper transcoder, as it does not,> then the error arises... at least that's what I think :)>> set "canreinvite=yes" (or just comment it since that's the default) on both> parties and try again.>> Let me know if it works.>> Alyed>> > Return-Path: <[EMAIL PROTECTED]>Wed> Sep 20 12:38:41 2006> Received: from digium-69-16-138-164.phx1.puregig.net> [69.16.138.164] by maila11.webcontrolcenter.com with SMTP;> Wed, 20 Sep 2006 12:38:41 -0700> Received: from digium-69-16-138-164.phx1.puregig.net> (localhost [127.0.0.1])>> Still having the same problem. i modified the sip.cfg in order to make> g729 the first choice:>>>> voice.codecPref.G711A="3" voice.codecPref.G729AB="1"> voice.codecPref.IP_4000.G711Mu="1" voice.codecPref.IP_4000.G711A="2"> voice.codecPref.IP_4000.G729AB=""/>>>> Cheers,> Santiago>> On 9/19/06, Alyed Tzompa wrote:> > Make sure the codec used by the Polycom will be only g729 via the phone's> > web interface, as far as I remember Polycom will try always to use ulaw or> > alaw first unless it is configured to use only or as first choice the g729> > codec.> >> > Alyed> >> > > > Return-Path: Tue>> > Sep 19 14:47:54 2006> > Received: from digium-69-16-138-164.phx1.puregig.net> > [69.16.138.164] by maila11.webcontrolcenter.com with SMTP;> > Tue, 19 Sep 2006 14:47:54 -0700> > Received: from digium-69-16-138-164.phx1.puregig.net> > (localhost [127.0.0.1])> > by lists.digium.com (Postfix) with ESMTP id AB0F03C1F4;> >> > Hi, I'm experiencing some problems with polycom phones, asterisk and g729> > codec.> >> > As I understand, between polycom and polycom i can use g729 without> > license at all as long as I'm using codec_g729.so module (i'm using> > the Open Source Implementation (> >> http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/> > )> > because it's pure pass-thru and there's no transcoding).> >> > My sip.conf has the following options:> >> > [general]> > disallow=all> > allow=g729> > allow=ulaw> >> >> > [voipuser]> > type=friend> > username=user> > host=dynamic> > callerid=user <202>> > [EMAIL PROTECTED]> > secret=gbvVf423> > canreinvite=no> > insecure=yes> > disallow=all> > allow=g729> >> >> > so i force the voipuser to use g729 as main codec. The problem comes> > when i try to connect to other polycom phone with the same config as> > voipuser. The CLI shows the following:> >> > Sep 19 18:37:38 NOTICE[8226]: chan_sip.c:3691 process_sdp: No compatible> > codecs!> >> > show modules doesnt show codec_g729.so but if i try to load it i get this:> >> > Unable to load module codec_g729.so> > Sep 19 18:39:16 WARNING[10688]: loader.c:305 __load_resource: Module> > 'codec_g729.so' already exists> >> >> > Anyone had this issue?> >> > If you need more information, feel fre to ask for it :)> >> >> > Thanks a lot!> >> > Santiago> > ___> > --Bandwidth and Colocation provided by Easynews.com --> >> > asterisk-users mailing list> > To UNSUBSCRIBE or update options visit:> >> http://lists.digium.com/mailman/listinfo/asterisk-users> >> >> > ___> > --Bandwidth and Colocation provided by Easynews.com --> >> > asterisk-users mailing list> > To UNSUBSCRIBE or update options visit:> >> >> http://lists.digium.com/mailman/listinfo/asterisk-users> >> >> >> ___> --Bandwidth and Colocation provided by Easynews.com -->> asterisk-users mailing list> To UNSUBSCRIBE or update options visit:> http://lists.digium.com/mailman/listinfo/asterisk-users>>> ___> --Bandwidth and Colocation provided by Easynews.com -->> 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] Configuring Codecs

[Alyed Tzompa]

		We can help guy, but pls send detailed info regarding a specific problem. The more info you provide, the best we can help.
As for g729 codec instructions look at
http://www.digium.com/en/supportcenter/documentation/viewdocs/G729 
there you'll find detailed and updated information about it.
Alyed 
		
		
		
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		Good Day,I
am new to Asterisk and I need help in configuring codecs in Asterisk. I
also will need to buy the license for G 729 codec and need to put that
key in Asterisk. Can someone please provide the step by step
instructions for the codec configurations?Also I am having
problems forwarding calls to landline or cellular numbers. As soon as
the preson on the forwarded end picks up the call drops - what can
cause this and what is the solution?ThanksWyatt___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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re: [asterisk-users] Cisco 7970 behind NAT

[Alyed Tzompa]

		Since the phone is the one behind a NAT,
and the registration is done only with SIP packages, setting or not the
"nat" is not an issue (ONLY for registration purposes). You can see
this since Asterisk is receiving the registration. Why is it denying
it?... wel,  that's something that will most likely has to do with
the registrationn parameters (user-passwd), but certainly not with the
network configuration.Alyed
		
		
		
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		Does anyone have this working? I have a Cisco 7970 with the 8-0-2-SR1S firmware loaded on it. I can get the phone to register with * just fine when I place my asterisk server on the same subnet and do no NAT. When I give my asterisk server a static public IP and put the phone behind a NAT to connect to the server registration fails. I turn on sip debugging and see that the phone is trying to register but it gets 401 Unauthorized. The same phone config is being used with only modifications to the IPs of the proxy and some NAT settings. I've adjusted NAT settings in two places (phone config and sip.conf).Example:sip.confchange "nat=never" to "nat=yes"Phone config:change0to1Does anyone have a similar setup with a 7970 behind NAT to an asterisk server that is not behind NAT? Any help or thoughts would be greatly appreciated.Jeremiah___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] g729 and polycoms problem

[Alyed Tzompa]

		Not an expert at reading Polycom config files, but guess g729 and ulaw are both preference 1 isn't it?
hey... you have in your sip.conf configuration "canreinvite=no"...
think this may be a problem: since Asterisk will always stay in the
path of the RTPs, I think it might need to have the proper transcoder,
as it does not, then the error arises... at least that's what I think :)
set "canreinvite=yes" (or just comment it since that's the default) on both parties and try again.
Let me know if it works.
Alyed 
		
		
		
Return-Path: <[EMAIL PROTECTED]> Wed Sep 20 12:38:41 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by maila11.webcontrolcenter.com with SMTP;   Wed, 20 Sep 2006 12:38:41 -0700Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1])
		Still having the same problem. i modified the sip.cfg in order to makeg729 the first choice:voice.codecPref.G711A="3" voice.codecPref.G729AB="1"voice.codecPref.IP_4000.G711Mu="1" voice.codecPref.IP_4000.G711A="2"voice.codecPref.IP_4000.G729AB=""/>Cheers,SantiagoOn 9/19/06, Alyed Tzompa  wrote:>  Make sure the codec used by the Polycom will be only g729 via the phone's> web interface, as far as I remember Polycom will try always to use ulaw or> alaw first unless it is configured to use only or as first choice the g729> codec.>> Alyed>>  > Return-Path:  Tue> Sep 19 14:47:54 2006> Received: from digium-69-16-138-164.phx1.puregig.net> [69.16.138.164] by maila11.webcontrolcenter.com with SMTP;>  Tue, 19 Sep 2006 14:47:54 -0700> Received: from digium-69-16-138-164.phx1.puregig.net> (localhost [127.0.0.1])>  by lists.digium.com (Postfix) with ESMTP id AB0F03C1F4;>> Hi, I'm experiencing some problems with polycom phones, asterisk and g729> codec.>> As I understand, between polycom and polycom i can use g729 without> license at all as long as I'm using codec_g729.so module (i'm using> the Open Source Implementation (> http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/> )> because it's pure pass-thru and there's no transcoding).>> My sip.conf has the following options:>> [general]> disallow=all> allow=g729> allow=ulaw>>> [voipuser]> type=friend> username=user> host=dynamic> callerid=user <202>> [EMAIL PROTECTED]> secret=gbvVf423> canreinvite=no> insecure=yes> disallow=all> allow=g729>>> so i force the voipuser to use g729 as main codec. The problem comes> when i try to connect to other polycom phone with the same config as> voipuser. The CLI shows the following:>> Sep 19 18:37:38 NOTICE[8226]: chan_sip.c:3691 process_sdp: No compatible> codecs!>> show modules doesnt show codec_g729.so but if i try to load it i get this:>> Unable to load module codec_g729.so> Sep 19 18:39:16 WARNING[10688]: loader.c:305 __load_resource: Module> 'codec_g729.so' already exists>>> Anyone had this issue?>> If you need more information, feel fre to ask for it :)>>> Thanks a lot!>> Santiago> ___> --Bandwidth and Colocation provided by Easynews.com -->> asterisk-users mailing list> To UNSUBSCRIBE or update options visit:> http://lists.digium.com/mailman/listinfo/asterisk-users>>> ___> --Bandwidth and Colocation provided by Easynews.com -->> asterisk-users mailing list> To UNSUBSCRIBE or update options visit:>> http://lists.digium.com/mailman/listinfo/asterisk-users>>>___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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re: [asterisk-users] g729 and polycoms problem

[Alyed Tzompa]

		Make sure the codec used by the Polycom
will be only g729 via the phone's web interface, as far as I remember
Polycom will try always to use ulaw or alaw first unless it is
configured to use only or as first choice the g729 codec.Alyed 
		
		
		
Return-Path: <[EMAIL PROTECTED]> Tue Sep 19 14:47:54 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by maila11.webcontrolcenter.com with SMTP;   Tue, 19 Sep 2006 14:47:54 -0700Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1])	by lists.digium.com (Postfix) with ESMTP id AB0F03C1F4;
		Hi, I'm experiencing some problems with polycom phones, asterisk and g729 codec.As I understand, between polycom and polycom i can use g729 withoutlicense at all as long as I'm using codec_g729.so module (i'm usingthe Open Source Implementation (http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ )because it's pure pass-thru and there's no transcoding).My sip.conf has the following options:[general]disallow=allallow=g729allow=ulaw[voipuser]type=friendusername=userhost=dynamiccallerid=user <202>[EMAIL PROTECTED]secret=gbvVf423canreinvite=noinsecure=yesdisallow=allallow=g729so i force the voipuser to use g729 as main codec. The problem comeswhen i try to connect to other polycom phone with the same config asvoipuser. The CLI shows the following:Sep 19 18:37:38 NOTICE[8226]: chan_sip.c:3691 process_sdp: No compatible codecs!show modules doesnt show codec_g729.so but if i try to load it i get this:Unable to load module codec_g729.soSep 19 18:39:16 WARNING[10688]: loader.c:305 __load_resource: Module'codec_g729.so' already existsAnyone had this issue?If you need more information, feel fre to ask for it :)Thanks a lot!Santiago___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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[asterisk-users] channel.c: Avoided initial deadlock for '0x8de2dc0', 10 retries!

[Alyed Tzompa]

		Hi there!
I'm having lots of problems with an Asterisk used by a customer. 
Got hundreds (yes hundreds, about 3-4 per minute) of this messages
every hour:
WARNING[12685] channel.c: Avoided initial deadlock for '0x96dee78', 10 retries!
The hex number changes with every message. The warning number does so as well but less often, thrice or twice a day. 
This Asterisk cuts offs calls once or twice, every day and is makes the
customer run into serious problems since it is a call center.
Asteriski version is: 1.2.10 with zaptel 1.2.7 and libpri 1.2.3
2 PRI ISDN E1's and about 50 concurrent calls constantly
Any ideas?
Many thanx!Alyed
		

		

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Re: [asterisk-users] nat and qualify questions

[Alyed Tzompa]

		

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

		> qualify=xxx|no|yes


>where XXX is the number of milliseconds used. If yes the default timeout is used, 2 seconds.

>If you turn on qualify in the configuration of a SIP device in sip.conf, Asterisk will send a SIP OPTIONS
command regularly to check that the device
>is still online. If the
device does not answer within the configured (or default) period (in
ms) Asterisk considers the device off-line for future calls.

		

What happens if you use nat=yes is that Asterisk will consider the IP
for communicating with the SIP user agent (UA) as the IP from where the
SIP invite comes from instead of taking the one included in the SDP
message. Hence if you are using phones inside a LAN this 2 addresses
will be the same, but if your SIP UA is outside they will not.
Having all your phones set with nat=yes and qualify =yes, will not
affect the behaviour of your phones if your network is not really full,
but will be a bad and dirty way to do it :)Alyed
		
		
		
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		As
far as i know qualify=yes will increase you network traffic, this will
make asterisk to communicate with all sip friends every X seconds, not
sure the default value.On 8/1/06, 
BerkHolz, Steven <[EMAIL PROTECTED]> wrote:Are there any 
problems with always having nat=yes and qualify=yes? We just opened up 
our server to be accessible to SIP from the internet. (used to require 
VPN) I had to set the SIP 
setting for my test softphone to nat=yes and qualify=yes.This makes 
sense. Some of these phone 
will never leave our building.Some of these phone 
will come and go. (laptops) Is the any negatives 
to just have all phones set to nat=yes and qualify=yes?If not, why is it 
not the default?  Thank You,Steven 
BerkHolz- MCSA 
- MCSE -Manager of Information SystemsTESCO Group 
CompaniesFax. 248-836-5101www.TESCOGroup.comBoard member 
ofwww.glimasoutheast.org
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re: [asterisk-users] SIP and NAT

[Alyed Tzompa]

		Could you please explain what the network configuration you want to try? it would be really helpful.
you can be as simple as:  SIPphone--> internet --> NAT--> asterisk
or whatever your particular scenario is.Alyed 
		
		
		
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		Hello all. I'm having a little problem here with NAT, and I already read a lot of documentation on web, but I still cant understand how to get asterisk and "external (on internet)" sip clients connected.Could anybody give me a tip ?ThanksLincoln___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] G729 Softphone

[Alyed Tzompa]

		As far as I there is no free softphone that can handle G729 codec. So you will need a licenced one.
Have a call center working with eyebeam from counterpath (previously
known as Xten) for about a year with no problems. Don't know if it
supports 
the URL option, but I'm pretty sure it will. Anyway you can ask this directly to their tech support. Alyed
		
		
		
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		Daniel Salama a écrit :> Looking for a SIP or IAX softphone for a call center application that > can do G729 codec. Any recommendations? Ideally it would do screen pops, > meaning that it will understand the URL option of the Dial command.Of course I'm a little biased, but I think MozPhone is well suited to call center application: it does natively support URL option of Dial or Queue command. It does not support G729 though, but speex will give you nice quality / low bandwith.Have a look at http://moziax.mozdev.org/ and please send feedback / comment / questions to MozPhone's mailing list at: http://moziax.mozdev.org/list.htmlThanks,Jean-Denis___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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re: [asterisk-users] So many configuration files!

[Alyed Tzompa]

		first: download the latest version 1.2.5 had some bugs and is already several months old.Depending on how you want your asterisk to behave will be the amount of files you'll need to mess with. Let's say you want a very basic installation with some SIP phones (hard or soft), then you'll have to deal with sip.conf and extensions.conf only so everything else is just vanity :DUsually you would like to have some voicemail, conference rooms, music on hold, pick up your neighbours extension, dial another asterisk or an IAX softphone, and PSTN access, then change some configs in voicemail.conf, meetme.conf, musiconhold.conf, features.conf, iax.conf and zapata.conf respectively.Want more action?Manage asterisk from an external application and mess with manager.conf,  change the way logs are being saved and CRMs with logger.conf and  the cdr_ *.conf files,  try some text to speach (TTS) with festival.confFeel like you are in the right track?try dealing with any ".c" file, recompile asterisk and make it behave just the way you always dream of (btw if it works you might want to share your new feature with all of us :) )Alyed 
		
		
Return-Path: <[EMAIL PROTECTED]> Tue Jul 11 11:51:59 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP;   Tue, 11 Jul 2006 11:51:59 -0700Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1])
		
		I'm working with Asterisk 1.2.5 to get a working system.There are 50 Asterisk configuration files in /etc/asterisk.Are they _all_ called by Asterisk or are some only used in a #include?Is there any way to get a list of which ones Asterisk uses by default?There is only a single #include file and it doesn't even exist.I have only messed with 4 files so far.Are there any more I should be editing?Which ones could be safely ignored?So far the system is just SIP with Zaptel to be added next.The 4 files I have changed are:sip.confextensions.confextensions_additional.confvoicemail.confMy list of files in /etc/asterisk - sorted most recent last:[EMAIL PROTECTED] asterisk # ls -1trzapata.confvpb.conftelcordia-1.adsiskinny.confsip_notify.confrtp.confrpt.confres_odbc.confqueues.confprivacy.confphone.confoss.confosp.confmusiconhold.confmodules.confmodem.confmisdn.confmgcp.confmeetme.confmanager.conflogger.confindications.confiaxprov.confiax.conffestival.conffeatures.confextensions.aelextconfig.confenum.confdundi.confdnsmgr.confcodecs.confcdr_tds.confcdr_pgsql.confcdr_odbc.confcdr_manager.confcdr_custom.confcdr.confasterisk.confasterisk.adsialsa.confalarmreceiver.confagents.confadtranvofr.confadsi.confsip.confextensions.confextensions_additional.confvoicemail.conf-- Larry Alkoff N2LA - Austin TXUsing Thunderbird on Slackware Linux___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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re: [asterisk-users] setting of volume

[Alyed Tzompa]

		>like if some one speaks loud, he would get a low volume.I'm sorry, but this goes far beyond Asterisk (at least for the moment) :)Anyway you can still play with rxgain and txgain in zapata.conf, but this will increase/reduce the overall volume gains and can also affect echo perception.Alyed 
		
		
		

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Is possible to set an default volume on a PBX, so that all lines have the same volume, like if some one speaks loud, he would get a low volume.
		


		


		 

		


		 

		
//Michael 
		

		
		

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re: [asterisk-users] trouble with * and # infront of a phonenumber

[Alyed Tzompa]

		As of Asterisk 1.0.X a "#" was recognized as a pattern not as a digit, hence in order to use it at the begining of an extension you should use "_" before it. I guess this is still valid in 1.2.X versions.i.e: use _#31#0046011 in your extensions.confAlyed 
		
		
		

Return-Path: <[EMAIL PROTECTED]> Sat Jul 08 12:32:46 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP;Sat, 8 Jul 2006 12:32:46 -0700
		
		
		

		
I cant make call when using #31#0046011
		


		


		 

		
The call just disapper  and noting shows on CLI 
		


		


		 

		


		 

		
Using asterisk 1.2.9.1
		


		
I am using cubix IAX2 softphone 
		


		


		 

		
And I had the same problem when I used xlite sip softphone 
		


		


		 

		
My voip-provider is Rixtelecom 
		


		


		 

		
//Michael Nielsen 
		


		


		 

		


		 
		
		

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RE: Re: [asterisk-users] Help with MusicOnHold!!!

[Alyed Tzompa]

		2 things might worth having a look:
a) set up in your zapata.conf:
    musiconhold=default
b) You say the asterisk version is 1.1, but 1.1 is developement
version, maybe was just a typo, but you should be using either a 1.0.X
or 1.2.X versionAlyed 
		
		
		
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		Hi,
 
Yes we have that installed.  Music on hold works fine when called in extensions.conf 
e.g. 
exten => 5000,2,MusicOnHold()
 
However when I put someone on hold the music does not play
I am using the Polycom Soundstation IP301 and X-lite phones.
 
Thanks
 
Julian
> Date: Fri, 7 Jul 2006 09:35:56 +0800> From: [EMAIL PROTECTED]> To: asterisk-users@lists.digium.com> Subject: Re: [asterisk-users] Help with MusicOnHold> >
Do you have the "mpg123" utility in your system? By default, Asterisk > uses "mpg123" to play the mp3 files for music on hold.> > -kokmeng.> > Julian Varanini wrote:> > > Hi,> >  >
> I am running asterisk 1.1.  When a client is placed on hold from the >
> x-lite or polycom phone, no hold music is heard.  I have > > musicclass=default set up in sip.conf and default exists in >
> musiconhold.conf.  Has anyone had a similar experience? Any help would > > be appreciated.> >  > > Thanks> >  > > Julian> >> >> >> >> >> >> >> >___> >--Bandwidth and Colocation provided by Easynews.com --> >> >asterisk-users mailing list> >To UNSUBSCRIBE or update options visit:> >   http://lists.digium.com/mailman/listinfo/asterisk-users> >  > >> > > ___> --Bandwidth and Colocation provided by Easynews.com --> > asterisk-users mailing list> To UNSUBSCRIBE or update options visit:>    http://lists.digium.com/mailman/listinfo/asterisk-users
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[asterisk-users] E1 additional calling party number

[Alyed Tzompa]

		Hi there!
I'm setting up an E1 with a new Telco and they are asking me to add the
extension number (CallerID)into an "Additional calling party number". Guess it
refeers to  a part of the E1 trace they are getting. I've been
playing around with the callerid and in zapata.conf and sip.conf but
have not been able to send them the data the way they are asking.
Anyone there has an idea on what config shall I try??
So far with my configs I'm able to send them the data in the "Display" part of the trace.
My configurations look like this:
sip.conf
[general]
trustrpid =yes
restrictcid=yes / no
[1007]
host=dynamic
type=friend
secret=1007
callerid=1007
zapata.conf
usercallerid=yes
hidecallerid=no
usecallingpres=yesAlyed
		

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re: [asterisk-users] Incoming Call matching to peer

[Alyed Tzompa]

		You have a little confusion:
friend => can GENERATE and RECEIVE calls
peer => can only GENERATE calls
user => can only RECEIVE callsAlyed 
		
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		I have this in sip.conf:[ata1]username = ata1accountcode = ata1qualify = yessecret = footype = friendhost = dynamicfromdomain = ipt.gumby.comcontext = fromata_startqualify = yesWhen
a call comes in from this device, if I have type=peer, Asterisk doesn't
match it, but it does if type=friend. Why not? I thought peers made and
received calls, and friends only made calls. If that's the case, then
it should match against type=peer, and certainly not against
type=friend.Doug.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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[Asterisk-Users] additional calling party number

[Alyed Tzompa]

		Hi there!
I'm setting up an E1 with a new Telco and they are asking me to add the
extension number into an "Additional calling party number". Guess it
refeers to  a part of the E1 trace they are getting. I've been
playing around with the callerid and in zapata.conf and sip.conf but
have not been able to send them the data the way they are asking.
Anyone there has an idea on what config shall I try??
So far with my config I'm able to send them the data in the "Display" part of the trace.
sip.conf
[general]
trustrpid =yes
restrictcid=yes / no
[1007]
host=dynamic
type=friend
secret=1007
callerid=1007
zapata.conf
usercallerid=yes
hidecallerid=no
usecallingpres=yesAlyed

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re: [Asterisk-Users] Re: no audio

[Alyed Tzompa]

		That was a bug fixed in Asterisk version 1.2.3  recently version 1.2.6 was released, so don't worry you can try the latest one without timing fears :DAlyed 
		
		

Return-Path: <[EMAIL PROTECTED]> Sat Apr 01 15:42:39 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP;Sat, 1 Apr 2006 15:42:39 -0700
		Wierd timing - I'm struggling with exactly the same issue. My problemwas with ZAP - ZAP. The phones ring, but no audio. Turns out there'sa bug with the version I'm running. It has to do w/ the system date. When I changed my system date to 1-Jan-06, everything worked!! Here'swhat I found from another posting:>this is a new bug which is submitted: http://bugs.digium.com/view.php?id=6349>change your system date to an older value. everything will work again.I'm hoping the bug is fixed in a more recent release build, but Ihaven't tried yet.Yours,HughOn 4/1/06, Luis herrera <[EMAIL PROTECTED]>wrote:> Hi. I have a [EMAIL PROTECTED] setup at my home. My problems is with> phones outside my network. I call the extensions> without a problem, it rings but when they answer I> can't hear them and they can hear me.> I set up in the SIP.CONF> nat=yes>> I'm I missing any other setting or do I need a special> switch that support asterisk.> Thank you for your help.>>> __> Do You Yahoo!?> Tired of spam? Yahoo! Mail has the best spam protection around> http://mail.yahoo.com> ___> --Bandwidth and Colocation provided by Easynews.com -->> Asterisk-Users mailing list> To UNSUBSCRIBE or update options visit:> http://lists.digium.com/mailman/listinfo/asterisk-users>___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] G729 codec problems

[Alyed Tzompa]

		I used g729 couple of times in the past and got the warning messages ONLY when I was trying to use more channels than the total amount of licenses I'd got.If you are sure you are using only one device that needs the license, I would suggest to check out how it is communicating with Asterisk. Also, if you have enough time try using the g729 with another soft / IP phone and see if you get the same result.Alyed ---I am not. I have one license and use i channel.It seems to detect the fact there are no more channels left and keepswarning me about it in case I want to use more.It is fine, but the warning is constant. All you see on Asteriskconsole is running warning message.RudolfOn 4/2/06, Kevin P. Fleming <[EMAIL PROTECTED]>wrote:> RumaTech wrote:>> > And it keeps running like that. Call usually come through OK. If i try> > to use "show g729" command, it shows that all codecs are in use. Well,> > this is fine, I am using one, but I do not want to see those warnings.> > Once is quite enough. Those continuos warnings make it impossible to se> > any other asterisk output. How does one turns them off?>> You can't make them stop except by not trying to use more channels than> you have licenses for.> ___> --Bandwidth and Colocation provided by Easynews.com -->> Asterisk-Users mailing list> To UNSUBSCRIBE or update options visit:> http://lists.digium.com/mailman/listinfo/asterisk-users>___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] Problem: ringtones stop unexpectedly

[Alyed Tzompa]

		Have you tryed phoning a fixed line instead of a cell phone?is this giving the same result?I assume your outgoing call to a the cellphone goes through a Zap channel. Try another one (e.g. Zap channel 2), and let us know the result.Alyed 
		
		

Return-Path: <[EMAIL PROTECTED]> Sat Apr 01 18:47:36 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP;Sat, 1 Apr 2006 18:47:36 -0700
		
		I should've mentioned that before. I've tried doing that and it has noeffect. I've tried both upper and lower-case 'r's.I've also tried a workaround that I thought would work, but it doesn't:Answering the call and then using the playtones(ringing) command beforeconnecting to my cellphone. -Original Message-Date: Sat, 1 Apr 2006 19:59:46 +0100From: "Julian J. M." <[EMAIL PROTECTED]>Subject: Re: [Asterisk-Users] Problem: ringtones stop unexpectedlywhen multiple channels are dialedTo: "Asterisk Users Mailing List - Non-Commercial Discussion"Message-ID:<[EMAIL PROTECTED]>Content-Type: text/plain; charset=ISO-8859-1Try adding 'r' to the dial options. According to "show application dial":r - Indicate ringing to the calling party. Pass no audio to thecallingparty until the called channel has answered.exten => 3058472194,1,Dial(SIP/1035&SIP/[EMAIL PROTECTED],50, r)Julian.On 4/1/06, Carlos A. Alfaro <[EMAIL PROTECTED]>wrote: Hello Everyone. I usually find my own solutions for problems but thistime,> after several months, I've given up. My asterisk is set up so that incoming calls from my voip provider ring on> both my sip extension and my cellphone at the same time. When the system> receives an incoming call, ringtones indicating that the call is being> connected play normally for the first 5 seconds to the caller, but they> suddenly stop as the call to my cellphone starts to make progress. This> causes some people to hang up, despite the fact that the call is stillbeing> connected. Callers who stay on the line are able to talk to me on either> the sip extension or the cellphone once I pick up either one. I have tried a lot of workarounds like including a priority to answer the> incoming call, invoke the playtones command before the dial command, but> this doesn't seem to work either. Can anyone replicate the problem? HaveI> ran into a bug? I have pasted as much info as I deemed relevant; pleaselet> me know if I'm missing something. Thanks.--___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-usersEnd of Asterisk-Users Digest, Vol 21, Issue 2*___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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re: [Asterisk-Users] Asterisk Between PBX and FXS

[Alyed Tzompa]

		As I understand this, it's a problem of redirecting the call to the same FXS channel.
To replicate this behaviour in the Asterisk you could try the following in the extensions.conf:
(suppose your FXS channel is group 1 in zapata.conf)
exten => 100,1,Dial(Zap/g1/${EXTEN},20)
exten => 100,1,Hangup
		

exten => 200,1,Dial(Zap/g1/${EXTEN},20)
exten => 200,1,Hangup
		

Then you'll end up with 2 extensions using the same FXS channel (of course not at the same time).
Hope this is what you are looking for.
Alyed
		
		
		
Return-Path: <[EMAIL PROTECTED]> Wed Mar 29 15:42:30 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP;   Wed, 29 Mar 2006 15:42:30 -0700Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1])	by lists.digium.com (Postfix) with ESMTP id 347B3C322;	Wed, 29 Mar 2006 15:38:29 -0700 (MST)
		Hi guys,I'm setting up asterisk to run with another pbx server. This pbx serversupport a feature that allows 2 extensions connect to the same FXS. No I putasterisk in the middle.Asterisk receives the call and dial to a SIP/peer.How the pbx installed support 2 extensions to one fxs... How can I figure outin asterisk which extension was dialed before the call came to asterisk?Does asterisk receive this information in some variable?${BRIDGEPEER}${CALLERID(dndi)}${BLINDTRANSFER}${BLINDTRANSFER}I tried the above variables without success.Thanks in advance.Fernando Lujan___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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RE: [Asterisk-Users] Dumb question - reaching the PSTN

[Alyed Tzompa]

		I may add a very nice configuration:
-  Use two (or more) Asterisks to create your own VoIP network
Very useful if you have broadband and several facilities spread out in distant geographical locations.Alyed 
		
		
		
Return-Path: <[EMAIL PROTECTED]> Wed Mar 29 16:32:16 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP;   Wed, 29 Mar 2006 16:32:16 -0700
		
		
		Not dumb at all.  Seeing that most end users in the world are still onthe PSTN, you're going to want to attach to it at some point.Since this is Asterisk, you have options.  Lots of them.  And, usingcareful dialplan-ning you can mix and match to your heart's content.Here are some examples:1) Use a Digium or a non-digium card to connect to POTS, ISDN, PRI, etc.2) Use a "in and out" VoIP provider such as telasip, vonage, etc3) Use an "out only" VoIP provider such as voipjet (generally lowerper-minute rates)4) Use an "in only" VoIP provider (examples fail me...)5) Use an FXO card (as in #1) but connect it to another device ratherthan the telco.  E.g. a cell phone6) Use a full-blown GSM gateway7) EtcUsing the above options, I personally have configured Asterisk to:1) Interface to my T12) Use voipjet to place outbound calls3) When dialing a company cell phone, send it out a cellular deviceuntil they're all used up, then use the T1Bob McDowell-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED] On Behalf Of CharlesMarcusSent: Wednesday, March 29, 2006 4:38 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Dumb question - reaching the PSTNHi everyone,I am fairly new to the idea of VoIP, although I've been reading about itoff and on for the last few years. Now it is starting to look matureenough to consider implementing it, but there is one thing that Ihaven't been able to get a clear answer on...With Vonage, you are using the Vonage network - it is theirresponsibility to route your call to the endpoint, which is more thanlikely on the old fashined PSTN.If I install Asterisk, how do my calls actually get completed? How dothey get 'bridged' over to the PSTN?I attended a Seminar today hosted by Dynasis, and one of the issues wasVoIP. ShoreTel was there, and the said I had to have phone lines,whether they were POTS lines, chennels from a T-1, whatever, we stillhad to have phone lines.Now I'm confused.If I implement an Asterisk based system (yes, I'd be paying a consultantto help), will I still have to maintain phone lines and pay full pricefor Long Distance?Simple pointers to White Papers on this issue will be sufficient.Many thanks,-- Best regards,Charles___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-usersEMAIL PRIVELEGED & CONFIDENTIAL CLIENT COMMUNICATION   *** PRIVILEGED AND CONFIDENTIAL CLIENT COMMUNICATION ***This
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re: [Asterisk-Users] Asterisk and LCR

[Alyed Tzompa]

		I use Portaone's PortaSIP for everything related to LCRAlyed
		
		
		
Return-Path: <[EMAIL PROTECTED]> Wed Mar 29 16:48:54 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP;   Wed, 29 Mar 2006 16:48:54 -0700Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1])	by lists.digium.com (Postfix) with ESMTP id 2D302C3B8;	Wed, 29 Mar 2006 16:36:31 -0700 (MST)
		
		I'm wondering how you guys handle least cost routing within Asterisk.Basically, we have a few providers with, obviously, different rates  per route. Additionally, we have a number of clients who have DIDs  assigned to them (either pointing to a single SIP peer or to more  complex dialing rules, such as, ring multiple SIP peers, or ring SIP  peers sequentially, or overflow to external forwarded number).What we are trying to do is to ease the pain of Asterisk dialing  through the least cost route regardless of where the destination is.  If the call is to go through a provider, well, then dial the least  cost provider. If someone is dialing a number which happens to be one  of the DIDs assigned in our system, it shouldn't have to go out  through any provider but just be routed properly in the dialing plan.Also, if a client has defined a rule to forward calls to another  number, how can I let the dial plan to dial out that number while  taking advantage of the LCR in place?We looked at lcdc script but not quite sure is what we need or maybe  don't understand it completely to implement it correctly.Thanks,Waldo___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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re: [Asterisk-Users] User Extension Custom Voicemail

[Alyed Tzompa]

		if you want to use the defaults unavailable and bussy just put it in /var/spool/asterisk/voicemail/default/100/unavail.wav
The "100" extension will automatically be created after you leave your
first voicemail message, change "unavail.wav" for "busy.wav" to use
them as
Voicemail(u100) and 
		Voicemail(b100) respectivelly in your extensions.conf
		
		
Alyed 
		
		
		
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		I created extension 100 in my asterisk system for myself.  I then wentahead and created a WAV file to use for my voicemail message.  Where wouldI go about putting this WAV file so that I can use it?Thank you,Jyran GluckyAdvisory ProgrammerBlueWare, Inc.Strategic HealthWare Solutions3060 W. 13th StreetCadillac, MI 49601Phone:  (231) 779-0224 ext. 111Fax: 231-779-1002Skype: Jyran GluckyAIM: JyranGluckymailto:[EMAIL PROTECTED]http://www.blueware.netDID YOU KNOW?BlueWare is the Grand Prize Winner of the 2005 IBM Beacon Award BEST DB2(Document Management) Application Worldwide.BlueWare Market Share for Hospital Document Management Systems is in 25states in the US.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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re: [Asterisk-Users] [OT] Polycom provisioning

[Alyed Tzompa]

		Polycom's can work in one of two ways:
a) using self configuration
b) downloading it from a ftp server
To make your Polycoms work with Asterisk you actually don't need the
phone to download any configuration, with the one embeded is ok. In any
case, when turned on, the phone searches for the ftp server, when it
cannot find it, it just continues loading with its self pre-configured
parameters.
Once this process is finished you should keep on with your own local network and sip configurationsAlyed
		
		
		
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		Does anyone have the polycom soundpoint ip's successfully remotely provisioning?  I've got the phone pulling default configs, and it's downloading phone specific information, but it's not actually using that information.  Any help would be appreciated :)-- Aaron DanielComputer Systems TechnicianSam Houston State University[EMAIL PROTECTED](936) 294-4198___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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re: [Asterisk-Users] kernel recompilation on a asterisk server

[Alyed Tzompa]

		Think a zaptel recompile is just what you need.Alyed 
		
		
		
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		i've got a asterisk server running on slackware 10.2 sice 3 weeks, iwill upgrade the kernel to 2.6.x from 2.4.x , i will upgrade to thelast stable release of kernel...i've got asterisk 1.2.4, zaptel 1.2.4 and libpri 1.2.2, correctlycompiled and configured, obviusly if i try to load the new kernel ican't load wtcxx modules, should i recompile zaptel 1.2.4 with make &&make install or there is a specific procedure to follow?thanks___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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RE: [Asterisk-Users] hardware and network requirements

[Alyed Tzompa]

Have a customer running some 25-28 concurrents calls (with about 35 agents logged in)without problems with a P4 2.X Ghz,  1GB RAM, I'm doing no transcoding btw.Alyed  Return-Path: <[EMAIL PROTECTED]> Sat Feb 04 16:59:29 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP;Sat, 4 Feb 2006 16:59:29 -0700> i'm planning to migrate a callcenter to asterisk and VOIP, > the call center can have up to 25 cuncurrents agents logged in.> Can a normal server with> 1 GB ram> 100 GB HDD> Pentium 4 3.6 Ghz CPU> Ethernet 10/100/1000One of our clients has a similar sized setup running on an Athlon64 2800+(2.2Ghz I think), 1GB RAM, 2x80GB HDDs in RAID1.You don't say how the calls are coming in, but I'd try and keep transcodingto a minimum. if they're coming from a PRI (i.e. alaw or ulaw) and you wantto keep them that way down to the users, 25 concurrent calls @ 80kbps-ish isonly 2mbps, so even a 100mbps LAN is fine for the task.Personally, I build our asterisk boxes rather than buying off-the-shelfservers, but I doubt it makes much difference one way or t'other. Go withwhichever approach you feel most comfortable.Regards,Chris-- C.M. Bagnall, Director, Minotaur I.T. LimitedThis email is made from 100% recycled electrons___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] nwebmail

[Alyed Tzompa]

>Also get the book (again I dont >have the URL if some one does please post it).  http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 Alyed   If you are new I would reccomend using [EMAIL PROTECTED]http://asteriskathome.soundforge.net . It is a greatresource for beginers. Also get the book (again I donthave the URL if some one does please post it).AsteriskRegards,Dovid--- yrving rivas  wrote:> Hello!> > I am new to Asterisk, AMP, Linux...did I say> all?..> I just installed Asterisk, and for my needs it is> working great.> In my AMP I see the nwebmail but I can´t get into> it.> When I place my login and password, comes with the> following message:> "An internal error has occured.> Please contact your system administrator.> If you are the system administrator, check the log> files".> > The log files don´t help me very much.> > Can someone tell me how to use the nwebmail?, how> to get in for first time?> > Regards!> > Yrving> > > > -> Do You Yahoo!? La mejor conexión a Internet y 2GB> extra a tu correo por $100 al mes.> http://net.yahoo.com.mx >___> --Bandwidth and Colocation provided by Easynews.com> --> > Asterisk-Users mailing list> To UNSUBSCRIBE or update options visit:> >http://lists.digium.com/mailman/listinfo/asterisk-users> __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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re: [Asterisk-Users] No D-channels available! Using Primary on channel 16 anyway!

[Alyed Tzompa]

I had a very similar problem some months ago, was using a Sangoma A101 card though. The problem was something related to the card's memory and was able to solve it by updating the driver. It was caused due to I was using a brand new card with a not so updated driver (I was using one that I thought was "stable") So my advice here is to check the driver version you are using if not the very last one, then update it. Try looking at the /var/log/messages file for any extra info, you might find something interesting. Alyed Return-Path: <[EMAIL PROTECTED]> Thu Jan 12 06:16:56 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP; Thu, 12 Jan 2006 06:16:56 -0700Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1]) by lists.digium.com (Postfix) with ESMTP id 6834A4066; Thu, 12 Jan 2006 06:15:45 -0700 (MST)Received: from psmtp.com (exprod5mx148.postini.com [64.18.0.180]) by lists.digium.com (Postfix) with SMTP id ACB784060 for ; Hi!I have a E1 PRI connected to my TE400P card on span 1, and two channelbanks on span 3 and 4 and  * 1.2.Every few hours I get this message and asterisk dies just after that:Warning: No D-channels available! Using Primary on channel  16 anyway!When this happens restarting zaptel and asterisk services, generally puts the system back onlinemy zaptel.con reads:span=1,1,0,ccs,hdb3 #span=2,0,0,cas,hdb3 span=3,2,0,esf,b8zs   #<-- This because we have two American CBsspan=4,3,0,esf,b8zsbchan=1-15dchan=16bchan=17-31fxoks=63-86fxoks=87-110loadzone = usIdeas anybody? Please?  Things done:* zttool/ztcfg* Trying R2 instead of PRI (R2 is the south americanstatdar, which wont even start)*Added crc4 to span1, with ugly sound consequences-- Paavum Regina, Per Secula et Secularum!!
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RE: [Asterisk-Users] read .what else to do ?

[Alyed Tzompa]

Sorry, I don't know how to forward a range of ports. To forward a single port, use something like: ip nat inside source static udp 192.168.1.2 5060 x.x.x.x 5060 extendable where x.x.x.x is your public IP.  just add the range ports tih a ":" e.g 192.168.1.2 1 : 10007   > (4)Please,I know alot of you out there have implemented AAH to work > outside your network ( Setting up your router/firewall so your remote SIP > phones can communicate with your [EMAIL PROTECTED] Server via SIP through a > NAT ).Please advise me how to make it work !!!  If what you are trying to do is a SIP --> NAT --> Internet --> Nat --> Asterisk   call  them I'm afraid you would need to use a SIP/RTP router. Alyed Return-Path: <[EMAIL PROTECTED]> Thu Jan 12 09:29:42 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP;Please note that recent IOS has SIP NAT traversal turned on by default.I believe that it only supports internal UA / external server.Since you also want the opposite, you should probably turn it off:no ip nat service sip tcp port 5060no ip nat service sip udp port 5060Some IOS versions will even crash on SIP behind NAT. Seehttp://lists.digium.com/pipermail/asterisk-users/2004-January/033718.htmlSorry, I don't know how to forward a range of ports. To forwarda single port, use something like:ip nat inside source static udp 192.168.1.2 5060 x.x.x.x 5060 extendablewhere x.x.x.x is your public IP.You can edit rtp.conf to use e.g 1-10007 (would allow 4 calls) andthen only 8 ip nat statements would be needed for RTP.You don't say what's failing. "make calls outside our LAN" sounds likeyou are trying to call using a VoIP provider that Asterisk registerswith. But "your remote SIP phones" is something different; which ofthe above are failing? Are the registrations successful? Is it justthe RTP that's not working (in which case the called phone will stillring)? If not, what error or timeout is reported?If * verbose and/or debug logs don't show precisely what is going wrong,use Ethereal (on both sides of the router if necessary) to see whatis happening.--Stewart> Hi all ,> I have tried configuring Asterisk at home to make calls outside our Lan> WITHOUT any success (Setting up your router/firewall so your remote SIP> phones can communicate with your [EMAIL PROTECTED] Server via SIP through a> NAT )> > To be precise i did the following> > (1) I Forwarded UDP Port 5060-5082 to 192.168.1.2> Forward UDP Port 1 to 2 to 192.168.1.2> > (2) I set externip = x.x.x.x (to our public WAN)> localnet =192.168.1.0 /255.255.255.0> > (3) I also set nat=yes> qualify=yes> > (4)Please,I know alot of you out there have implemented AAH to work> outside your network ( Setting up your router/firewall so your remote SIP> phones can communicate with your [EMAIL PROTECTED] Server via SIP through a> NAT ).Please advise me how to make it work !!!> > (5) I am using xten lite soft phone on my pc .> > (6) I use cisco 1700 series router ,and i have natting configured on> this router .Maybe I am using a wrong command .Please,tell me the> commands to forward the ports Port 5060-5082,1 to 2 to> 192.168.1.2 on a cisco router .> > Please reply and advice !!!> Thanks___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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re: [Asterisk-Users] Problem with an automatic responder

[Alyed Tzompa]

I would be useful if you could post your config files and the pri debug as well.  check your zapata.conf or paste it here so we can take a look.AlyedReturn-Path: <[EMAIL PROTECTED]> Thu Jan 12 10:04:28 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP; Thu, 12 Jan 2006 10:04:28 -0700Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1]) by lists.digium.com (Postfix) with ESMTP id 6DAB2C258; Thu, 12 Jan 2006 09:58:40 -0700 (MST)Received: from psmtp.com (exprod5mx128.postini.com [64.18.0.42]) by lists.digium.com (Postfix) with SMTP id 4101AC24B for ; Thu, 12 Jan 2006 09:58:34 -0700 (MST)Received: from source ([151.9.129.69]) (using TLSv1) by exprod5mx128.postini.com ([64.18.4.10]) with SMTP;  Thu, 12 Jan 2006 10:58:41 CSTReceived: from aulin (aulin.pitagora.it [193.227.67.249]) by allserv.pitagora.it (8.12.11/8.12.11) with ESMTP id k0CGwXKd006897 for ; Thu, 12 Jan 2006 17:58:33 +0100Received: from VIGGIANI ([193.227.66.207]) by MERCURIO.pitagora.it with Microsoft SMTPSVC(6.0.3790.1830); Thu, 12 Jan 2006 17:58:32 +0100X-Original-To: asterisk-users@lists.digium.comDelivered-To: asterisk-users@lists.digium.comFrom: "Mimmus" <[EMAIL PROTECTED]>To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Date: Thu, 12 Jan 2006 17:58:32 +0100Message-ID: <[EMAIL PROTECTED]>MIME-Version: 1.0Content-Type: text/plain; charset="us-ascii"Content-Transfer-Encoding: 7bitX-Mailer: Microsoft Office Outlook 11Thread-Index: AcYXmWqjqLUMCcjoQT2Cz9B43oUplQ==X-MimeOLE: Produced By Microsoft MimeOLE V6.00.2900.2180X-OriginalArrivalTime: 12 Jan 2006 16:58:32.0672 (UTC) FILETIME=[6B723600:01C61799]X-Spam-Status: No, score=-4.8 required=5.5 tests=ALL_TRUSTED,AWL,BAYES_00  autolearn=ham version=3.0.4X-Spam-Checker-Version: SpamAssassin 3.0.4 (2005-06-05) on allserv.pitagora.itX-Virus-Scanned: ClamAV version 0.88, clamav-milter version 0.87 on localhostX-Virus-Status: CleanX-pstn-levels: (S: 4.16697/99.78420 )X-pstn-settings: 1 (0.1500:0.1500) gt3 gt2 gt1X-pstn-addresses: from <[EMAIL PROTECTED]> [90/4]Subject: [Asterisk-Users] Problem with an automatic responderX-BeenThere: asterisk-users@lists.digium.comX-Mailman-Version: 2.1.5Precedence: listReply-To: Asterisk Users Mailing List - Non-Commercial Discussion List-Id: Asterisk Users Mailing List - Non-Commercial Discussion List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Sender: [EMAIL PROTECTED]Errors-To: [EMAIL PROTECTED]X-SmarterMail-Spam: SPF_NoneX-Rcpt-To: <[EMAIL PROTECTED]>Hi,I have this setup: (PSTN E1 PRI) -- Asterisk -- (crosscable) -- Alcatel PBX --- analog phonesand a few of VoIP phones directly connected to Asterisk.Calling a number (only one until now!) - an automatic responder (IVR) - fromVoIP phones works, from analog phones doesn't work: NOANSWER after a fewseconds. I'm using no 'r' in dial options (this caused a problem with an IVRsome time ago).I can post pri debug output in both cases, if needed.Thanks in advance for any help-- Mimmus___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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re: [Asterisk-Users] Outbound routing

[Alyed Tzompa]

can't you ask the users to dial a prefix? that can solve your problem. btw, which provider are you using for your calls to the USA? Alyed Return-Path: <[EMAIL PROTECTED]> Wed Jan 11 09:47:29 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP; Wed, 11 Jan 2006 09:47:29 -0700Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1]) by lists.digium.com (Postfix) with ESMTP id 91BB3C4BD; Wed, 11 Jan 2006 09:17:16 -0700 (MST)Received: from psmtp.com (exprod5mx130.postini.com [64.18.0.44]) by lists.digium.com (Postfix) with SMTP id C7900C473 for ; Wed, 11 Jan 2006 09:17:04 -0700 (MST)Received: from source ([200.93.220.22]) (using TLSv1) by exprod5mx130.postini.com ([64.18.4.10]) with SMTP;  Wed, 11 Jan 2006 08:17:09 PSTReceived: from ozzy (srv.eth0.www.manta.telconet.net [200.93.220.18]) by smtp.manta.telconet.net (8.12.11/8.12.11) with ESMTP id k0BGPPla027509 for ; Wed, 11 Jan 2006 11:25:25 -0500X-Original-To: asterisk-users@lists.digium.comDelivered-To: asterisk-users@lists.digium.comFrom: Guillermo Salas M <[EMAIL PROTECTED]>To: Asterisk Users Mailing List - Non-Commercial Discussion Content-Type: text/plainOrganization: Telconet S.A.Date: Wed, 11 Jan 2006 11:16:55 -0500Message-Id: <[EMAIL PROTECTED]>Mime-Version: 1.0X-Mailer: Evolution 2.2.3Content-Transfer-Encoding: 7bitX-Spam-Status: No, score=-4.1 required=7.5 tests=ALL_TRUSTED,AWL,BAYES_00, DNS_FROM_RFC_ABUSE autolearn=unavailable version=3.1.0X-Spam-Checker-Version: SpamAssassin 3.1.0 (2005-09-13) on  smtp.manta.telconet.netX-Virus-Scanned: ClamAV 0.87/1237/Tue Jan 10 10:53:20 2006 on smtp.manta.telconet.netX-Virus-Status: CleanX-pstn-levels: (S: 8.29151/99.85851 )X-pstn-settings: 1 (0.1500:0.1500) gt3 gt2 gt1X-pstn-addresses: from <[EMAIL PROTECTED]> [90/4]Subject: [Asterisk-Users] Outbound routingX-BeenThere: asterisk-users@lists.digium.comX-Mailman-Version: 2.1.5Precedence: listReply-To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion List-Id: Asterisk Users Mailing List - Non-Commercial Discussion List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Sender: [EMAIL PROTECTED]Errors-To: [EMAIL PROTECTED]X-SmarterMail-Spam: SPF_NoneX-Rcpt-To: <[EMAIL PROTECTED]>Hi all, I've 3 providers (A, B, and C) the A is giving me freecalls toUSA, the B is giving my freecalls to Europe, and C is to call the otredestinations. My question is, how can I configure the outboud routing toselect the right trunk for every destination?All the providers uses the dialing form 00 1 123 4567890 when 00 is thenumber dialed to call, 1 the country code, 123 the area code and 4567890the phone number.I've the following outbound routing with AMP, but the calls are beenstarted by the first provider in the trunk sequence list:Route Name : InternationalDial Patterns : 00.Trunk Sequence: A B CI want to make that the USA calls going with A, Europe calls with B andrest of the world with C.Is this possible ? Can you gime a little of help with this... Than you in advance. :)-- Guillermo Salas M.Telconet S.A. MantaCalle 15 y Av. 24 Esq.Phone : 593 5 262 8071Mobile: 593 9 985 5138SIP : [EMAIL PROTECTED]e-mail: [EMAIL PROTECTED]www : http://www.telconet.net http://www.telcocarrier.netLinux User: 255902Soporte en Linea en http://www.manta.telconet.netPlease avoid sending me Word or PowerPoint attachments.See http://www.fsf.org/philosophy/no-word-attachments.html___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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re: [Asterisk-Users] how to adjust volume

[Alyed Tzompa]

Don't know if you can actually adjust the volume in any of them, but you can try from the asterisk with rxgain / txgain in your zapata.confAlyed Return-Path: <[EMAIL PROTECTED]> Mon Jan 09 16:27:24 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP; Mon, 9 Jan 2006 16:27:24 -0700Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1]) by lists.digium.com (Postfix) with ESMTP id BAA93C839; Mon, 9 Jan 2006 16:26:29 -0700 (MST)Received: from psmtp.com (exprod5mx147.postini.com [64.18.0.179]) by lists.digium.com (Postfix) with SMTP id 08788C829 for ; Mon, 9 Jan 2006 16:26:26 -0700 (MST)Received: from source ([64.233.162.206]) by exprod5mx147.postini.com ([64.18.4.10]) with SMTP; Mon, 09 Jan 2006 15:26:29 PSTReceived: by zproxy.gmail.com with SMTP id o1so4491534nzf for ; Mon, 09 Jan 2006 15:26:29 -0800 (PST)Received: by 10.36.221.48 with SMTP id t48mr13251723nzg; Mon, 09 Jan 2006 15:26:29 -0800 (PST)Received: by 10.36.97.3 with HTTP; Mon, 9 Jan 2006 15:26:29 -0800 (PST)X-Original-To: asterisk-users@lists.digium.comDelivered-To: asterisk-users@lists.digium.comDate: Mon, 9 Jan 2006 23:26:29 +From: Asterisk guy <[EMAIL PROTECTED]>To: Asterisk Users Mailing List - Non-Commercial Discussion Precedence: listReply-To: Asterisk Users Mailing List - Non-Commercial Discussion Sender: [EMAIL PROTECTED]Errors-To: [EMAIL PROTECTED]X-SmarterMail-Spam: SPF_NoneX-Rcpt-To: <[EMAIL PROTECTED]>how to adjust voice volume for sipura 2000 and cisco ata186?___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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re: [Asterisk-Users] Problem with integrating ISDN PBX using NT mode

[Alyed Tzompa]

 == Primary D-Channel on span 4 down for TEI 65 == Primary D-Channel on span 4 down for TEI 64 == Primary D-Channel on span 4 down for TEI 66 This looks like a signaling problem, check out your configuration in zaptel.conf and the log messages you get at /var/log/messages You can also post them here if you continue experiencing problems.   Alyed Hi,I'm just in the process of replacing a crappy Siemens PBX with a new andshiny Asterisk system. To connect Legacy equipment I hooked up a smallISDN PBX (DeTeWe OpenCom 36) to one port on a Junghanns.net quadBRIcard. That port is configured for NT Point to Multipoint(Mehrgeraeteanschluss) mode. Now I can place calls from the ISDN PBX tothe other Asterisk extensions but the other way around does not work.Whenever I call from the Asterisk server to one of the extensionsconnected through the ISDN PBX that extension rings for a split secondand then the call is dropped. Here is what I get on the console: -- Executing Macro("SCCP/13-002f", "standard|Zap/g2/40") in new stack -- Executing Dial("SCCP/13-002f", "Zap/g2/40|20") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g2/40 == Primary D-Channel on span 4 up for TEI 64 == Primary D-Channel on span 4 up for TEI 66 -- Zap/10-1 is proceeding passing it to SCCP/13-002f -- Zap/10-1 is ringing -- Channel 0/1, span 4 got hangup request -- Hungup 'Zap/10-1' == No one is available to answer at this time (1:0/0/0) -- Executing Goto("SCCP/13-002f", "s-NOANSWER|1") in new stack -- Goto (macro-standard,s-NOANSWER,1) -- Executing VoiceMail("SCCP/13-002f", "u40") in new stack -- Executing Goto("SCCP/13-002f", "default|s|1") in new stack -- Goto (default,s,1) == Channel 'SCCP/13-002f' jumping out of macro 'standard' == Primary D-Channel on span 4 down for TEI 65 == Primary D-Channel on span 4 down for TEI 64 == Primary D-Channel on span 4 down for TEI 66I think I properly configured the ISDN PBX (theres not much to configurethere). Can someone help me here? What's causing the hangup request? Howcould I find out?Below is the relevant configuration.Thanks in advance,Frederik Fixzapata.conf:[channels]switchtype = euroisdnpridialplan = localprilocaldialplan = localnationalprefix = 0internationalprefix = 00;usecallingpres=yesechocancel = yesechocancelwhenbridged = yesechotraining = 100debug = 2; Festnetzanschlusssignalling = bri_cpecontext=externgroup = 1; S/T port 1channel => 1-2; S/T port 2channel => 4-5; S/T port 3channel => 7-8; Interner S0-Bussignalling = bri_net_ptmpcontext = intern-isdngroup = 2; S/T port 4channel => 10-11extensions.conf:[macro-standard]exten => s,1,Dial(${ARG1},20)exten => s,2,Goto(s-${DIALSTATUS},1)exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN})exten => s-NOANSWER,2,Goto(default,s,1)exten => s-BUSY,1,Voicemail(b${MACRO_EXTEN})exten => s-BUSY,2,Goto(default,s,1)exten => _s-.,1,Goto(s-NOANSWER,1)exten => a,1,VoicemailMain(${MACRO_EXTEN})[nebenstellen-intern]; Konferenzzimmerexten => 13,1,Macro(standard,SCCP/13); Ingridexten => 17,1,Macro(standard,${INGRID})exten => 57,1,Macro(standard_ohne_ab,Zap/g2/57); Gavigoexten => 60,1,Macro(standard,SCCP/60); Woelmexten => 66,1,Macro(standard,SCCP/66)exten => 68,1,Macro(standard_ohne_ab,Zap/g2/68); van de Beeckexten => 40,1,Macro(standard,Zap/g2/40)exten => 44,1,Macro(standard_ohne_ab,Zap/g2/44); Rohanexten => 50,1,Macro(standard,Zap/g2/50)exten => 59,1,Macro(standard_ohne_ab,Zap/g2/59); Hinterhausexten => 58,1,Macro(standard,Zap/g2/58)exten => 22,1,Macro(standard,Zap/g2/22); fuer Testzweckeexten => 61,1,Macro(standard,SIP/eyebeamtest); virtuelle Nebenstellenexten => 30,1,Macro(virtuell)exten => 35,1,Macro(virtuell)exten => 48,1,Macro(virtuell)exten => 25,1,Dial(Zap/g2/58)[intern-isdn]exten => 25,1,Dial(SCCP/13)exten => s,1,DISA(no-password|intern)[dialout]exten => _0.,1,Dial(Zap/g1/${EXTEN:1})[intern]include => dialoutinclude => nebenstellen-intern___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] Asterisk Debugging

[Alyed Tzompa]

I do agree, plus even if you don't know anything about scripting there are plenty of shell tutorials out thereAlyed No but shell scripts are pretty easy and will cleanup your file for you.On Jan 5, 2006, at 1:07 PM, Douglas Garstang wrote:> Not everyone is a C programmer extraordinairre.> -Original Message-----> From: Alyed Tzompa [mailto:[EMAIL PROTECTED]> Sent: Thursday, January 05, 2006 11:59 AM> To: Douglas Garstang; asterisk-users@lists.digium.com> Subject: RE: [Asterisk-Users] Asterisk Debugging>> Then stop looking for easy solutions and get your hands dirty > changing your c files___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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RE: [Asterisk-Users] Asterisk Debugging

[Alyed Tzompa]

Then stop looking for easy solutions and get your hands dirty changing your c files Alyed  Well, I want the output that the NoOp's generate. I want to be able to manually log lines to a file through some mechanism. I just wish I could do it without all the extra NoOp stuff at the front.   I just tried using: mylogfile => verbose   in logger.conf but all I got was the startup/shutdown asterisk messages. Besides, this isn't what I wan't. I don't want Asterisk internal generated log messages. I want my OWN log messages, that I specify.   Doug      -Original Message-From: Alyed Tzompa [mailto:[EMAIL PROTECTED]Sent: Thursday, January 05, 2006 11:18 AMTo: asterisk-users@lists.digium.comSubject: re: [Asterisk-Users] Asterisk DebuggingI don't find the console output ugly, maybe messy, but never ugly :PIf u don't like those NoOp, just take them away from ur extensions.conf. BTW, to  save the console output to a given file, just edit your logger.conf file. Say you only want the console output, then just add to your filename the verbose option . The file will be saved wherever is defined in the asterisk.conf (the default is /var/log/asterisk) after editing the file you'll need to do either an Asterisk restart or input CLI> logger rotate  at the Asterisk console.i.e. ;logger.conf[logfiles]mylogfile => verboseAlyed   I'd like to have Asterisk log useful messages during operation.Is there any way in extensions.conf that I can manually log messages to a file, say via syslog()? The console output is ugly, with all the extra "Executing NoOp("SIP/pstn.voip.com-08a28bd0"," crud at the front of each line. I'm not sure how to save console output anyway.Thanks,Doug.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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re: [Asterisk-Users] Asterisk Debugging

[Alyed Tzompa]

I don't find the console output ugly, maybe messy, but never ugly :P  If u don't like those NoOp, just take them away from ur extensions.conf. BTW, to  save the console output to a given file, just edit your logger.conf file.  Say you only want the console output, then just add to your filename the verbose option . The file will be saved wherever is defined in the asterisk.conf (the  default is /var/log/asterisk) after editing the file you'll need to do either an Asterisk restart or input CLI> logger rotate  at the Asterisk console. i.e.  ;logger.conf  [logfiles] mylogfile => verbose Alyed I'd like to have Asterisk log useful messages during operation.Is there any way in extensions.conf that I can manually log messages to a file, say via syslog()? The console output is ugly, with all the extra "Executing NoOp("SIP/pstn.voip.com-08a28bd0"," crud at the front of each line. I'm not sure how to save console output anyway.Thanks,Doug.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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