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

[Juan E. Rodríguez]

How is your system configured?
Debug output of faild faxes?

This kind of information is needed to help you!

Regards,
Juan

khalid touati wrote:
can anyone help me out in this, a big number of my faxes
are lost everyday! i would really appreciate any help on how i can
tweak asterisk (rxfax) to receive all faxes!
  
  2010/4/2 khalid touati 
  i
went ahead and i used this line:exten =>
3772,n,rxfax(${FAXFILE}|debug) as it says in the rxfax tutorial (also
because i'm not sure that FAXOPT is supported by asterisk 1.2), but no
output neither in the CLI or in a file. so is there any body who knows
about that?
i'll appreciate any help!

2010/4/2 khalid touati 


thank
you guys for responses,
Danny-, am i going to receive debug info in the CLI or in a default
file (/var/log /*)? is FAXOPT supported whitin asterisk 1.2 (sorry I
forgot to mention that i am using 1.2)? 
  
  
  2010/4/2 Tzafrir Cohen 
  
  

On Fri, Apr 02, 2010 at 10:11:59AM -0400, khalid touati
wrote:


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






They have a hard-wired log file. Make sure Asterisk can
write to it.

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



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-- 
Abdullah
  






-- 
Abdullah

  
  
  
  
-- 
Abdullah




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Re: [asterisk-users] SIP Connection Question

[Juan E. Rodríguez]

If * answers the call, it will be on the "loop" but with canreinvite or 
directrtp the media can be out of * and redirected to the final end point even 
if signaling goes through *.

For the trunk, you can have multiple simultaneous calls. I do not know about 
Mitel's licensing but with only one trunk you can have as much calls as * 
supports.

Saludos,
Juan E. Rodríguez


-Original Message-
From: "Dr. Kenneth Noisewater" 
Date: Thu, 01 Apr 2010 19:35:54 
To: ; Asterisk Users Mailing List - Non-Commercial 
Discussion
Subject: Re: [asterisk-users] SIP Connection Question

On 4/1/2010 5:09 PM, Juan E. Rodríguez wrote:
> Depends on the configuration you make. For example, if you want to route the 
> call giving the Mitel a new desrination or prefix, you can use Transfer 
> dialplan app. Transfer before answering the call will be redirected with SIP 
> 302.
>
> If the call is to be anwered on *, then canreinvite set to yes or directrtp 
> set to yes can help you.
>
>
> Saludos,
> Juan E. Rodríguez
>
>
> -Original Message-
> From: Kenneth Noisewater
> Date: Thu, 1 Apr 2010 16:50:47
> To:
> Subject: [asterisk-users] SIP Connection Question
>
>
OK, so for instance if I passed a call to Asterisk and grabbed CID info 
and did some lookups and then transferred it back to mitel to route to a 
user, then * would be out of the call path (loop, whatever). But, if I 
were to answer that call in * with an IVR to collect caller input to use 
and then transferred the call back to the Mitel to route to the 
endpoint, * would remain in the call. Is that a correct understanding?

Also one more question, and please excuse my ignorance (I'm just a 
developer with pretty limited knowledge on the telephony side of things):

When I talk about connecting the Mitel box and the Asterisk box together 
via a SIP trunk, is that trunk equal to 1 analog line, or channel or 
whatever, or can I make as many connections as I want on that trunk? 
Again, my knowledge is a bit limited, and thusfar people have been using 
a lot of terms interchangably with me to add to my confusion :). This 
only concerns me because I'm pretty sure we have to buy a license for 
each SIP trunk with Mitel.

It would be really great if I could work out a solution like this, it 
will allow me to prove Asterisk's worth to my management, and open up a 
lot of doors for us and our internal apps. The Mitel SDK is 
unfortunately rather limited, but management is not in any way 
interested in jumping ship from Mitel to Asterisk. Personally, I say 
jump, I've had great experience with Asterisk, even in fairly heavy use 
situations. Anyone have any input on selling Asterisk to higher up's? I 
know there is the whole "enterprise support" aspect, but my team manages 
the Mitel stuff as it is anyway, and I think we'd all much rather be 
dealing with Asterisk/SER as the core solution.

Thanks everyone for your input!

Kenny
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Re: [asterisk-users] SIP Connection Question

[Juan E. Rodríguez]

Depends on the configuration you make. For example, if you want to route the 
call giving the Mitel a new desrination or prefix, you can use Transfer 
dialplan app. Transfer before answering the call will be redirected with SIP 
302.

If the call is to be anwered on *, then canreinvite set to yes or directrtp set 
to yes can help you.


Saludos,
Juan E. Rodríguez


-Original Message-
From: Kenneth Noisewater 
Date: Thu, 1 Apr 2010 16:50:47 
To: 
Subject: [asterisk-users] SIP Connection Question

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Re: [asterisk-users] RPID on called party

[Juan E. Rodríguez]

Try using sendrpid=yes on sip.conf

Regards,
Juan

Ondrej Valousek wrote:
> Hello,
>
> Did anyone manage to force asterisk to put Remote-party-ID attribute on 
> the SIP outgoing call? I.e. When A calls B, I want that A gets a name of 
> B displayed on his phone.
> Note that name of A gets displayed on the B's phone fine, but this is 
> not what I want.
> This works with Cisco Call manager fine - the RPID is sent as a part of 
> the response to the SIP INVITE this way:
>
>
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 192.168.60.20:5060;branch=z9hG4bK42892c32;rport
> From: "Ondrej Valousek"   
> ;tag=as4786d518
> To:   
> ;tag=f75ff5d8-1023-4240-bc4b-d7eeb6d0d77d-42063104
> Date: Tue, 30 Mar 2010 13:53:15 GMT
> Call-ID: 465a9c200587260d164f451409489...@192.168.60.20
> CSeq: 102 INVITE
> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, 
> SUBSCRIBE, NOTIFY
> Allow-Events: presence
> *Remote-Party-ID: "Paul Ryan"  
>  ;party=called;screen=yes;privacy=off*
> Contact:   
> Content-Length: 0
>
>
> But I can not make it working with Asterisk. Does anyone have any glue 
> how to achieve this WITHOUT patching asterisk? I am happy to upgrade to 
> the latest/greatest version, I just do not want to patch.
> Many thanks,
>
> Ondrej
>
>   

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Re: [asterisk-users] Confusion on call forwarding

[Juan E. Rodríguez]

After setting promiscredir set to yes * is goign to send the call to the first 
desrination on the Contact header.

--Mensaje original--
De: Richard Kenner
Remitente: asterisk-users-boun...@lists.digium.com
Para: asterisk-users@lists.digium.com
Responder a: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] Confusion on call forwarding
Enviado: 30 Mar, 2010 14:56

> You need promiscredir set to yes on sip.conf

And then what do I do in the dialplan?  I.e., what context is the 
redirect number interpreted in?  Google searches on this issue show
inconsistent and contradictory information.

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Saludos,
Juan E. Rodríguez
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Re: [asterisk-users] Confusion on call forwarding

[Juan E. Rodríguez]

You need promiscredir set to yes on sip.conf

--Mensaje original--
De: Richard Kenner
Remitente: asterisk-users-boun...@lists.digium.com
Para: asterisk-users@lists.digium.com
Responder a: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: [asterisk-users] Confusion on call forwarding
Enviado: 30 Mar, 2010 13:38

I'm confused.  What does Asterisk do when it gets a 302 with a new number to
forward to?  Is there anything I have to do in the dialplan to make this work?
I can't find any clear documentation on this issue.

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Saludos,
Juan E. Rodríguez
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Re: [asterisk-users] a2billing wont pass the number

[Juan E. Rodríguez]

When you say 'a2billing' won't pass the number, you mean you are calling to an 
IVR or something like that.

And when did you dial you destination number twice???

Saludos,
Juan E. Rodríguez


-Original Message-
From: Nathanial Allan 
Date: Tue, 30 Mar 2010 13:08:24 
To: 
Subject: [asterisk-users] a2billing wont pass the number

I am running into an issue with A2Billing. I will explain  first of all that 
everything else works! the system is 90% complete its just this one small 
problem I am running into. 

So my problem is that when I place a call, 
1. I dial my number that I want and A2Billing gets activated 
2. it asks for my pin, upon successful entry of my pin A2Billing then
3. prompts me for my phone number then 
4. The call goes out (and actually connects for the record)

So I am entering my destination phone number twice which is not the worst thing 
that can happen, though it is a little annoying

Any light that you can shine on this problem would be greatly appreciated as I 
have been working on it for too long now and I want to get a product!


Thank You

NallaN
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Re: [asterisk-users] need help on setup rtp directly between 2 sipclients

[Juan E. Rodríguez]

Try setting canreinvite and nat to no for those extensions.

Saludos,
Juan E. Rodríguez


-Original Message-
From: Alyed 
Date: Fri, 26 Mar 2010 10:56:50 
To: Asterisk Users Mailing List - Non-Commercial 
Discussion
Subject: Re: [asterisk-users] need help on setup rtp directly between 2 sip
clients

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Re: [asterisk-users] What does this error message mean

[Juan E. Rodríguez]

Show sip.conf and extensions.conf related part.

Maybe I misread but did you mention you have a exten... Line in sip.conf???

The error is because the received user is not the same as the configured one.

--Mensaje original--
De: Ira
Remitente: asterisk-users-boun...@lists.digium.com
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Responder a: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] What does this error message mean
Enviado: 26 Mar, 2010 21:01

At 05:47 PM 3/26/2010, you wrote:
>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.

Why thanks. The sad part is it means madly flailing has exactly the 
opposite effect as you might expect.

Ira 


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Saludos,
Juan E. Rodríguez
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Re: [asterisk-users] How to get Sip response codes in Dialplan?

[Juan E. Rodríguez]

Try using DIALSTATUS.

--Mensaje original--
De: Zhang Shukun
Remitente: asterisk-users-boun...@lists.digium.com
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Responder a: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: [asterisk-users] How to get Sip response codes in Dialplan?
Enviado: 24 Mar, 2010 23:29

hi ,all

when a Dial or Queue excutes, a sip response code will return. like

== Using SIP RTP CoS mark 5
-- Got SIP response 502 "Bad Gateway" back from 211.150.119.32
-- SIP/95040-004a is circuit-busy
-- Nobody picked up in 2000 ms

My quesion is how to get the response code in the dial plan
immediatelly in order to do different thing according the returned
codes?

for example: a queue response code is "busy now" i will queue another
number immediately not let the user waiting for the timeout.

Thanks!

-- 
Best regards,
Sucan

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Saludos,
Juan E. Rodríguez
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Re: [asterisk-users] SIP Hard Phone with SMS

[Juan E. Rodríguez]

Johann:

Do you know how is the SMS sent over the IP, does it use SIP INFO
message or somthing like that?

Regards,
Juan

Johann Steinwendtner wrote:

  randulo schrieb:
  
  
2009/10/9 "Juan E. Rodríguez" :


  Does any one know about a SIP hard phone capable of sending SMS messages
(Or a SIP MESSAGE) that could be read from Asterisk dial plan??
  

The Gigaset S675IP series of DECT/SIP phone has SMS capability, but
not sure it can work with Asteris.


  
  Yes, they do. (app_sms) Make sure you have installed the latest FW.
Before, they sent the SMS out on the analog port only.

Hans

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Re: [asterisk-users] Faxing: Anyone have a compiled executable?

[Juan E. Rodríguez]

I have it running on * 1.4.19. You can get it on the internet, .so and an 
intaller that checks for dependencies.

--Mensaje original--
De: Doug
Remitente: asterisk-users-boun...@lists.digium.com
Para: asterisk-users@lists.digium.com
Responder a: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] Faxing: Anyone have a compiled executable?
Enviado: 7 Ene, 2010 01:41

At 16:49 1/5/2010, Tzafrir Cohen wrote:
 >On Tue, Jan 05, 2010 at 04:24:37PM -0600, Doug wrote:
 >> Hi,
 >>
 >> Having problems with getting either RxFax or FaxReceive
 >> to compile.  Running Asterisk 1.4 on CentOS 5.
 >
 >What version of SpanDSP do you use?

   spandsp-0.0.6pre12.tgz

and:

   libtiff-3.8.2-7.el5_3.4
   libtiff-devel-3.8.2-7.el5_3.4

Which do you recommend?


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Juan E. Rodríguez
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Re: [asterisk-users] Does A2Billing has mial list?

[Juan E. Rodríguez]

A2billing forum has a lot of information and questions are answered
very fast. Try searching on the forum before posting, cause the answer
may be there already.

forum.asterisk2billing.org/

Regards,
Juan

Bruce Nik wrote:
Hi Sucan,
  
  
  A2Billing doesn't have a mailing list but you may ask your
specific question on A2billing Forum or maybe even here. This may be of
intrest to you if you have an installation question:
  
  
  A2Billing automated install script :
  http://a2billing2asterisk.googlepages.com
  
  
  -Bruce
  
  On Tue, Dec 29, 2009 at 1:07 AM, Zhang
Shukun  wrote:
  hi,

Does A2Billing has mial list?

--
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Sucan

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Re: [asterisk-users] SIP Issue

[Juan E. Rodríguez]

You should set the ddwhome variable with the Set function or declare it
on the global context. Try the Dial command with the dial string
directly, before using the variable.

Fro debugging purposes you should set debug and verbose at least to 10
and check the logs.

Regards,
Juan

James A. Shigley wrote:
> What do you mean I should use a global function. I'm kind both well versed 
> and a newb to *
>
> James Shigley
>
>
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Juan E. 
> Rodríguez
> Sent: Monday, December 28, 2009 12:44 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] SIP Issue
>
> Is ddwhome defined in global context?? If so, then you should use global 
> function.
>
> Paste asterisk log to check.
> Saludos,
> Juan E. Rodríguez
>
>
> -Original Message-
> From: "James A. Shigley" 
> Date: Mon, 28 Dec 2009 12:11:35 
> To: Asterisk Users Mailing List - Non-Commercial 
> Discussion
> Subject: [asterisk-users] SIP Issue
>
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Re: [asterisk-users] SIP Issue

[Juan E. Rodríguez]

Is ddwhome defined in global context?? If so, then you should use global 
function.

Paste asterisk log to check.
Saludos,
Juan E. Rodríguez


-Original Message-
From: "James A. Shigley" 
Date: Mon, 28 Dec 2009 12:11:35 
To: Asterisk Users Mailing List - Non-Commercial 
Discussion
Subject: [asterisk-users] SIP Issue

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Re: [asterisk-users] pattern matching

[Juan E. Rodríguez]

You do not need to use pattern matching if you know the extension you are going 
to receive.

Check the spelling on the dialplan if it does not work. You can start at the 
duplicated comma of the 34102.

--Mensaje original--
De: Thomas Perron
Remitente: asterisk-users-boun...@lists.digium.com
Para: asterisk-users@lists.digium.com
Responder a: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: [asterisk-users] pattern matching
Enviado: 26 Dic, 2009 09:36

I want to ensure that only this extension is executed.
But, I have others that are similar.

I want:

exten => 34101,1,Answer()
exten => 34101,n,Record(34101:gsm)  ;   34101 test zip code
exten => 34101,n,Playback(34101)
exten => 34101,n,Hangup

Is this correct or do I need to have each of the four statements lead
with an underscore (_) to make an exact match?

Other code looks similar so I don't want the 102 to connect when I am
dialing 101

exten => 34102,1,Answer()
exten => 34102,,n,Record(34102:gsm)  ;   34102 test zip code
exten => 34102,n,Playback(34102)
exten => 34102,n,Hangup

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Juan E. Rodríguez
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Re: [asterisk-users] sip realtime question

[Juan E. Rodríguez]

I am not sure, but I think you will get nothing with those commands if realtime 
cathing is not set.

--Original Message--
From: Emre Kurnaz
Sender: asterisk-users-boun...@lists.digium.com
To: asterisk-users@lists.digium.com
ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] sip realtime question
Sent: Dec 11, 2009 4:02 AM

Hi everybody,

First of all i am sorry my English :)

i want to configure my asterisk server as a sip server that stores sip users in 
the mysql database connecting directly over odbc driver. My odbc configuration 
works as below

[r...@ao042 asterisk]# isql -v asterisk
+---+
| Connected!|
|   |
| sql-statement |
| help [tablename]  |
| quit  |
|   |
+---+


and i did refer to the site 
http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip . my res_odbc file as;

[r...@ao042 asterisk]# cat res_odbc.conf
[asterisk]
enabled => yes
dsn => asterisk
username => asterisk
password => ***
pre-connect => yes

[r...@ao042 asterisk]# grep -Ev '^;|^$' extconfig.conf
[settings]
sipusers => odbc,asterisk,sip_buddies
sippeers => odbc,asterisk,sip_buddies

and i created the asterisk database with sip_buddies table.

Here is my problem:

In asterisk console when i run the following command i get the answer,

ao042*CLI> realtime load sipusers name 100
   Column Name  Column Value
    
id  1
  name  100
  host  dynamic
   nat  no
  type  friend
cancallforward  yes
   canreinvite  yes
secret  Deneme01
  disallow  all
 allow  g729
 allow  ilbc
 allow  gsm
 allow  ulaw
 allow  alaw
  port  5060
regseconds  0
lastms  0
  username  100


but the following commands returns nothing

ao042*CLI> sip show users
Username   Secret   Accountcode  Def.Context  
ACL  NAT
ao042*CLI> sip show peers
Name/username  HostDyn Nat ACL Port Status 
Realtime
0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 offline]

besides it does not query anything. So what am i missing? Is there anything 
that i should mentioned in the sip.conf file?

by the way i am using RHEL 5.4 with 2.6.18-164.el5 kernel - 
asterisk16-1.6.0.17-1_centos5 rpm

Any help would be appreciated...

-- 

Emre Kurnaz
ITU/BIDB   | Istanbul Technical University / 
Information Technologies Office
Sistem Destek Grubu| System Support Team
RHCE : 805009174841679
Yar¹ Zamanl¹ Ö»renci Koordinatörü  | Part-Time Student Manager
kurn...@itu.edu.tr
+90 0212 285 3930

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Juan E. Rodríguez
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Re: [asterisk-users] Can't restart asterisk from script

[Juan E. Rodríguez]

You should replace the single quote with double quote.

--Original Message--
From: Michelle Dupuis
Sender: asterisk-users-boun...@lists.digium.com
To: 'Asterisk Users List'
ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't restart asterisk from script
Sent: Dec 9, 2009 10:59 PM

But the error message in my log shows the error to be from asterisk, so I'm
guessing I'm sending a parameter incorrectly to asterisk - which fits with
the no quote theory 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bill Kenworthy
Sent: Wednesday, December 09, 2009 9:31 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Can't restart asterisk from script

Keep in mond that cron usually has a very abbreviated environment for
security reasons - you may need to set the PATH or other environment
variables in the crontab to get it to work.

Billk


On Wed, 2009-12-09 at 20:55 -0500, Michelle Dupuis wrote:
> Interesting...I'll try that.  Thanks
> 
> 
> __
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lyle 
> Giese
> Sent: Wednesday, December 09, 2009 8:47 PM
> To: Asterisk Users List
> Subject: Re: [asterisk-users] Can't restart asterisk from script
> 
> 
> 
> Doug Lytle wrote: 
> > Warren Selby wrote:
> >   
> > > On Wed, Dec 9, 2009 at 3:08 PM, Michelle Dupuis  > > <mailto:supp...@ocg.ca>> wrote:
> > > 
> > > I'm running * 1.4 and can successfully restart asterisk from the
> > > command
> > > line with:
> > > /usr/sbin/asterisk -r -x "restart gracefully"
> > > 
> > > 
> > 
> > I have the following cron job:
> > 
> > /usr/sbin/asterisk -r -x 'restart when convenient'
> > 
> > Doug
> > 
> >   
> You probably don't need the single or double quotes at all.  I have 
> never used any quoting in crontab.
> 
> Lyle Giese
> LCR Computer Services, Inc.
> 
> 
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Saludos,
Juan E. Rodríguez
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[asterisk-users] Asterisk 302 Moved Temporarily

[Juan E. Rodríguez]

Hello,

I have an * installation that sometimes receives  a 302 "Moved
Temporarily" response to an INVITE. * sends the ACK but takes about 30
seconds to start the new INVITE to the new destination (from Contact
Header).
I have set core debugging to 20 but do not see any abnormal message.

[provider]
type=friend
host=a.reachable.host.ip
context=incoming_context
dtmfmode=rfc2833
canreinvite=yes
qualify=yes


The call is generated from a PHP AGI script with the  Dial  "RrCL"
options.


Does any one have an idea why it could be lasting about 30 seconds to
start the new call??



Regards,
Juan



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Re: [asterisk-users] Asterisk SS7 Sigtran Protocol

[Juan E. Rodríguez]

Right now, I think it does not.

Look out for it at: asterisk-...@lists.digium.com

Regards,
Juan

Khaled W Chehab wrote:

  
  

  
  
  Dears,
   
  Do Asterisk support SS7 SIGTRAN(SS7 over IP)
protocol ?
  And how to integrate 
   
   
  Regards
   
   
  Khaled 
Chehab
    
NGN Eng.
   
   
   
  
Operations Office - Lebanon
  
Office : +961 1 868686 ext 115
  
Mobile: +961 3 045212
  
E-mail:  kche...@xplorium.com
  
MSN ID :khalidche...@hotmail.com  
  
Web Site: http://www.Xplorium.com
   
   
  
  
  
  
*
No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates.
  
This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium.
  
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Re: [asterisk-users] IVR

[Juan E. Rodríguez]

As I see here, you do not have to include the big10 context inside the
default context, as you have an extension defined to reach that context
and its extention is start extension.
If the cleveland-IVR is based on the start extension too, the same
applies. 

Besides that, it would work...(maybe not the way you expect... :-) )

Regards,
Juan

Thomas Perron wrote:
Is this going to work:
  
[default]
include => stdexten
include => big10-IVR
include => cleveland-IVR
exten => _17035745353,1,Goto(big10-IVR,s,1)
exten => _15672528431,1,Goto(cleveland-IVR,s,1)
  
  
[big10-IVR]
exten => s,1,Answer()
exten => s,n,Background(dir-welcome)
;exten => s,n,WaitExten(1)
;exten => s,n,Background(astcc-please-enter-your)
  

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Re: [asterisk-users] Unable to set TOS to 184?

[Juan E. Rodríguez]

If you already mangle packets with IPTABLES, then you should comment
the line[s] tos_* on sip.conf. 

Regards,
Juan

Bart Fisher wrote:

  
  
  
  I don't understand this message:
   
  [2009-10-29 16:31:51]
WARNING[28510]: rtp.c:1997 ast_rtp_settos: Unable to set TOS to 184
   
  From what I have read the reason is
asterisk can't set TOS if not running in root.  Mine is running as
asterisk.
   
  I found one post that says to run at
boot:
   
  #!/bin/bash 
/sbin/iptables -A OUTPUT -t mangle -p udp -m udp --sport 4569 -j DSCP
--set-dscp-class ef
/sbin/iptables -A OUTPUT -t mangle -p udp -m udp --sport 1:2 -j
DSCP --set-dscp-class ef
/sbin/iptables -A OUTPUT -t mangle -p udp -m udp --sport 5060 -j DSCP
--set-dscp-class ef
   
  Does this make sense? Is this the
only method to end ths warning?
   
  Thanks, Bart
  

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Re: [asterisk-users] Dynamic DNS trunk

[Juan E. Rodríguez]

If the trunk is a dynamic IP you need the other end to register to
Asterisk, so letting Asterisk know the new IP.

Regards,
Juan

B.Masoud @ SH wrote:

  
  
  

  
  I have a trunk, and its host=dynamic dns.
  The problem is, when the IP changes the 
  Sip show peers 
  Still show the old IP of the DNS, I have to
reload and save
the configuration again so that asterisk recognize the new IP of the
DNS.
   
  Any idea how to automate such a thing? Or how
can I keep
asterisk to deal with NAMES as NAMES, and IPs as IPs.
   
  Let me know.
   
  Thanks.
  
  

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[asterisk-users] Asterisk 302 Moved Temporarily

[Juan E. Rodríguez]

Hello,

I have an * installation that sometimes receives  a 302 "Moved
Temporarily" response to an INVITE. * sends the ACK but takes about 30
seconds to start the new INVITE to the new destination (from Contact
Header).
I have set core debugging to 20 but do not see any abnormal message.

[provider]
type=friend
host=a.reachable.host.ip
context=incoming_context
dtmfmode=rfc2833
canreinvite=yes
qualify=yes


The call is generated from a PHP AGI script with the  Dial  "RrCL"
options.


Does any one have an idea why it could be lasting about 30 seconds to
start the new call??



Regards,
Juan



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Re: [asterisk-users] Best Firewall Suggestions?

[Juan E. Rodríguez]

Check out ClarkConnect and SmoothWall.

Regards,
Juan

Steve Totaro wrote:

  
  On Tue, Oct 13, 2009 at 2:41 PM, SIP 
wrote:
  

David Wathen wrote:
>
> Hi,
>
> My customer has a outdated firewall that is also presenting a NAT
> nightmare for getting the Asterisk server reachable from the
internet.
>
> What firewalls work good with VOIP? I really want to steer away
from
> any ALG supported firewall. I just want a good firewall that works
> well with Asterisk.
>
> Thanks,
>
> David Wathen
>


Depends on what level of firewall you're looking for.

For a full firewall on either a dedicated system or one of your own, I
cannot strongly enough recommend Astaro Linux firewall. Better
throughput than a pix, worlds easier to operate and configure, and
comparable in price. Very SIP/VoIP friendly. Loads of optional modules
(we use its mail filter module to filter spam/viruses for several
hundred thousand user mailboxes, for instance) to limit the cost to what
you need.

Also has a built in SIP Proxy, although I've never used it.

Excellent platform.


Of course, at home, I just use a little Linksys WRT box. It's hardly a
corporate-grade firewall, but it's quite SIP-friendly.


N.



  
  
No votes for Vyatta?  I have been seriously checking it out.
  
Thanks,
Steve T 
  
  
  
  

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Re: [asterisk-users] Billing applications

[Juan E. Rodríguez]

A2billing (Star2Billing, I think, for commercial support) is a good
choice and it's very mature software.
Astercc is very fast and has a very good callshop solution.

Regards,
Juan

voip crazy wrote:
> Hello all,
>
> I want to instal a Billing solution in the same asterisk's box. I have
> browse for ast2bill asterisk billing, astercc, and more, bu ti do not
> know which will be the best for me.
> The only things i need, are,
>   - Postpaid and prepaid applications.
>   - True CDR. Better that asterisk one, With suport for transfers
>   - I do not need support for reseller
>   - Billing for Voip, PSTN trunks
>
> I need a light app. I'm not searching a heavy app. with a lots of
> modules and applicacions. I need a ligth application for a soho and
> its needs.
>
> Any one are using a billing application which fits this needs?
> Any clue will be welcomed.
>
> Thanks in advance.
>
> VoipCrazy
>
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[asterisk-users] SIP Hard Phone with SMS

[Juan E. Rodríguez]

Does any one know about a SIP hard phone capable of sending SMS messages
(Or a SIP MESSAGE) that could be read from Asterisk dial plan??
Or at least with J2ME support, to run a little program?

Regards,
Juan

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Re: [asterisk-users] CIDlookup

[Juan E. Rodríguez]

Use CALLERID(name).

http://www.voip-info.org/wiki/view/Asterisk+func+callerid

Steve Totaro wrote:

On Thu, Jul 9, 2009 at 3:01 AM, Sriram wrote:
  

Hi List

I've a CID lookup hooked onto an inbound route (i m using trixbox) it
runs well but it returns the value as "CIDNAME"  ... if i just
want to display the CIDNAME [leaving the quotes and ] .. how can
i do it ? do i have to edit some macro in extensions.conf ?

rgds
Sriram
___



Use Cut()

http://www.voip-info.org/wiki/index.php?page=Asterisk+func+cut

  


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Re: [asterisk-users] Variable using AMI

[Juan E. Rodríguez]

Well, I do not understand very well what you are trying to do, but I'll 
give you some advice:


If you want a variable only for the AGI you call, you just have to 
declare that variable on the AGI.
If you would like to make visible that variable as long as the call is 
active and for each call, even if the name is the same, you have to set 
a channel variable with the Set(variable=value) command.
If you would like to have a variable shared between two or more 
channels, use the SHARED() funcion(Asterisk 1.6, back ported to 1.4)
If you want a variable to be accessed from all the channels, you could 
use a global variable.


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

Regards,
Juan

Carlos Ruiz Diaz wrote:

Thank you!

I did not know the existence of  DB command. The command allows me to 
store KVPs but I have to use the same variable name every time so 
every process that starts the AMI instance will override the values 
making it unusable for what I want to achieve.


It was really useful anyways. :)


2009/7/6 "Juan E. Rodríguez" <mailto:jerdg...@gmail.com>>


Maybe you could use the Asterisk Database.
In 1.4 you can do it with DBGet and DBPut:
http://www.voip-info.org/wiki/view/Asterisk+cmd+DBget
http://www.voip-info.org/wiki/view/Asterisk+cmd+DBput

In 1.6 use DB() function.

Regards,
Juan


David Backeberg wrote:

On Mon, Jul 6, 2009 at 11:37 AM, Carlos Ruiz
Diaz <mailto:carlos.ruizd...@gmail.com> wrote:
  

Hello,

if I do a variable assignation using AMI interface, that variable will be
visible only for the current AMI instance or will be readable for all AMI
instances?. I will login using the same user, concurrently. A program will
write a global variable using the same name and if asterisk don't have any
scope rules I have to find another way to do what I want.


If you want to maintain scope for a variable across multiple calls you
should maintain the value of that variable outside of asterisk and
keep setting it for each new phonecall. Global variables in asterisk
do not do what you are describing.

AMI does have something where you can name a particular AMI session,
and then communication for that session will care that name. That
should not be confused with a system-wide global variable.

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Re: [asterisk-users] Variable using AMI

[Juan E. Rodríguez]

Maybe you could use the Asterisk Database.
In 1.4 you can do it with DBGet and DBPut:
http://www.voip-info.org/wiki/view/Asterisk+cmd+DBget
http://www.voip-info.org/wiki/view/Asterisk+cmd+DBput

In 1.6 use DB() function.

Regards,
Juan

David Backeberg wrote:

On Mon, Jul 6, 2009 at 11:37 AM, Carlos Ruiz
Diaz wrote:
  

Hello,

if I do a variable assignation using AMI interface, that variable will be
visible only for the current AMI instance or will be readable for all AMI
instances?. I will login using the same user, concurrently. A program will
write a global variable using the same name and if asterisk don't have any
scope rules I have to find another way to do what I want.



If you want to maintain scope for a variable across multiple calls you
should maintain the value of that variable outside of asterisk and
keep setting it for each new phonecall. Global variables in asterisk
do not do what you are describing.

AMI does have something where you can name a particular AMI session,
and then communication for that session will care that name. That
should not be confused with a system-wide global variable.

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Re: [asterisk-users] PHP AGI Not Working and Odd Behavior

[Juan E. Rodríguez]

Try running your script with  /usr/bin/php5 script.php to test it
Or changing  #!/usr/bin/php5 -q to  #!/usr/bin/php -q


Leah Newmark wrote:

Thanks.
I didn't change anything in my dialplan. I am aware of reloading configuration 
:)

My AGIs are copied from a working asterisk install -- the shebang argument is 
how I've always done it. Either way, I have tried it without the -q as well, 
and that also didn't succeed.

I just tried your test and it worked fine to run it.

As I said, I know the server is reading the file I've been editing. I see it on 
the monitor. It's definitely opening the file to return that message...

Any other ideas?

On Wed, Jun 24, 2009 at 3:43 PM, Leah Newmarkhttp://lists.digium.com/mailman/listinfo/asterisk-users>> wrote:
  

/ Hi,


/>/
/>/ I'm running asterisk 1.4.22 on a debian server.
/>/ I have php5 installed and it works correctly command line.
/>/ When trying to run a php script via AGI, I get messages such as:
/>/ GI Tx >> I>
/>/ AGI Rx << #!/usr/bin/php5 -q
/>/ AGI Tx >> 510 Invalid or unknown command
/>/
/>/ The scripts are completely executable and owned by asterisk
/>/ -rwxr-xr-x 1 asterisk asterisk
/>/
/>/ Googling is not helping much, and php seems installed perfectly. Other
/>/ servers running the same AGIs have no such problem.
/>/
/>/ I also have noticed odd behavior. When I edit an AGI, the changes aren't
/>/ always showing up in the running of the script via asterisk.
/>/ For example, I tried editing the bash command to read #!/usr/bin/php -q,
/>/ and got the same response on my agi debug.
/>/ I know for a fact it's running the script I've edited:
/>/  Launched AGI Script /var/lib/asterisk/agi-bin/myscript.php
/>/ and it's not in some other directory.
/
Keep in mind that if you change your dialplan to call a different
script you will need to
cli> dialplan reload

Other than that, I'm not sure that it's legal to put an argument into
a shbang, as in your -q when launching php.
It's also possible you've somehow locked down php or directories way
too much. The proper test is to:
bash$ sudo -u asterisk /path/to/script.php


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Re: [asterisk-users] Asterisk SIP calls stop working having more than 300 calls (more than 600 channels)

[Juan E. Rodríguez]

I do, I am planning to have little more than 1000. Right now I had 
managed little more than 700 SIP channels + 100 IAX channels.


Do you think this can cause any problem?? --I mean, having this RTP 
ports range--



Tzafrir Cohen wrote:

On Fri, Oct 17, 2008 at 01:24:35AM -0400, Juan Rodríguez wrote:
  

Tzafrir:

Following the comments on your post, I started checking (after breaking my
head 'googling') the UDP ports in use, and found out that the script that my
Asterisk is running was using UDP connection too. This caused that ports
from 10,000 to 20,000 could not be used by Asterisk.

I change the port range from 10,000 to 40,, and now everything looks OK.



Why not change it to 9000- ?

Do you actually need more than 1000 sockets at a time?

  


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Re: [asterisk-users] 1 second delay when connecting calls

[Juan E. Rodríguez]

Neal:

Try having on sip.conf:

srvlookup=no

Regards,
Juan


[EMAIL PROTECTED] wrote:

Hello,

Thanks for your replies.

We checked our sip.conf and we have canreinvite=no already.  I agree 
it could be a firmware issue.  I will get another vendors phone hooked 
up to the pbx before going crazy with support.


Thanks,
Neal



On Sun, Oct 12, 2008 at 6:14 AM, Vieri <[EMAIL PROTECTED] 
> wrote:



--- On Sat, 10/11/08, Eric "ManxPower" Wieling <[EMAIL PROTECTED]
> wrote:

> Try setting canreinvite=no in each of the device sections on
> a couple of
> phones, reload chan_sip.so and see if that fixes things.
> It has fixed
> the issue when I've tried it.
>
> [EMAIL PROTECTED]  wrote:
> > Hello,
> >
> > We are using asterisk 1.6, sangoma pri card, and Cisco
> 7960 phones.  When we
> > make or receive calls there is a delay before voice is
> heard.  Anyone have
> > any ideas on where to start to debug or has anyone
> seen this before.  We
> > have played with settings on pri, asterisk, and phones
> with no change.

I'm having the same problem but with ATA-connected analog phones.
The ATAs are Grandstream GXW4008 with firmware v. 1.0.1.15
. The "canreinvite" option in sip.conf doesn't
change anything for me. Downgrading the GXW4008 solves this issue
so this is obviously a firmware bug in my case. I had a vague
report once of a user in another installation having this 1-second
delay on call connection. That user had a Cisco phone but I don't
remember which one. I suggest you check this with Cisco Support if
you can.

Vieri





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