Re: [asterisk-users] Custom PHP for Call Files

[Eherr]

Thanks Dale!

I will try your method in the one web directory. 

Also while I was waiting for a response, I decided to start from scratch in a 
different web directory and rewrite my code for a database based system. 

I am writing so that the system inserts into table the extension and time of 
call with those two grouped with a unique key to avoid duplicates. 

Then have a cron check every minute to move any files with time of call 
datertime as current to the outgoing directory. 

I have been considering two methods
1 - create the call file and leave in directory where Apache can write to. Then 
put the uniqueID of the call file into the database so when cron runs, it'll 
grab the file name and move that file to outgoing/

2- run cron and when it gets time for the call file, have the perl grab the 
extension and write the call file then and there and then move it to outgoing/ 
and deleting the database entry. 

I currently going for option 2. 

Is there any benefit in a database driven application or does best practices 
favor the original directory permissions?

I am not expecting a high level of wake up calls. Maybe the most of 5-10 in the 
system nightly. 

Thanks,
--Eric

Sent from my phone.

> On Dec 28, 2015, at 8:12 AM, Dale Noll  wrote:
> 
> 1. The 'Total' line that is displayed with the 'ls -l' command output is NOT 
> the total number of files, it is the total number of system blocks used by 
> the files in the directory.
> 
> 2. In order to truly understand the situation you need to understand the 
> Linux file system permissions.,,, Every file in Linux has 3 basic permissions 
> for 3 types of users.  The basic permissions are  Read, Write, eXecute.  The 
> types of users are User, Group and Other.  The permission 'mode' is displayed 
> in the 'ls -l' output in the first column.  In your example of the 
> callfile... '-rw-r--r--'.  This gives the User Read and Write to the file 
> while Group and Other have Read to the file.
> 
> Directories are simply special kinds of files. A directory (very short 
> explanation) is a file that contains the names of other files and a pointer 
> the the place on disk where each file's data is stored(blocks in Question 1). 
>  With that understanding... in order to create and delete files in a 
> directory, you need write permission to the directory. This you may have 
> figured out since you changed the permissions on the outgoing directory to 
> get PHP to be able to delete the files.  Now, you may not be able to get 
> Asterisk to properly handle the files if Asterisk is not running as root.
> 
> There are several ways to handle this, each with pros and cons, but how I 
> probably would do it is change the permissions on the 
> /var/spool/asterisk/outgoing directory so that the User and Group both have 
> write access to the directory.  
> 
> There are two commands to do that.
> 
> To change the permission mode, use the chmod command as root
> 
> chmod u=rwx,g=rwx,o=rx /var/spool/asterisk/outgoing
> 
> 
> To change the ownership of the directory, use the chown command as root
> 
> chown asterisk:apache /var/spool/asterisk/outgoing
> 
> 
> This will allow both the 'asterisk' User and the 'apache' group to create and 
> delete files in the directory.
> 
> 
> You may also find it beneficial to change the ownership of the call file when 
> you move them into the outgoing directory.  Asterisk will modify the files as 
> it make call attempts. Therefore the asterisk user should be able to write to 
> the call file itself.  In your script that moves the files, you may want to 
> add a 'chown asterisk:apache {callfile pathname}'. 
> 
> I don't know if your perl cron job does other things, but I would simply do 
> it all using a set of bash commands.
> 
> chown asterisk:apache /tmp/*.call && mv /tmp/*.call 
> /var/spool/asterisk/outgoing
> 
> That can be put into a shell script or run directly in a crontab entry.
> 
> 
> 
>> On Sun, Dec 27, 2015 at 9:14 PM, er ic  wrote:
>> I am hoping to get some help here with building custom PHP to manage a 'wake 
>> up call' system.
>> 
>> I have the script where the user can set the schedule for an extension wake 
>> up call.
>> 
>> It appears to write to the /var/spool/asterisk/outgoing/ directory.
>> 
>> My two issues:
>> 
>> 1 - when the files do get moved over to outgoing/ directory via a cron job, 
>> the permissions show "-rw-r--r-- 1 apache apache 100 Jan  1  2016 
>> 5680a312a28b2.call" and the calls get sent when the date comes to pass. But 
>> my question is, if I mv 3 files from my php script,  'll 
>> /var/spool/asterisk/outgoing/' shows 'total 12' when there are only three 
>> files in the directory. What does total mean? Is my perl script doing 
>> something that I am not aware of and really there are 12 files overlapped or 
>> something funky?
>> 
>> --- cron job perl script
>> my @list = glob("/tmp/*.call");
>> for( 0 .. $#list )
>> {
>> system "mv $list[$_] 

[asterisk-users] Caller ID not real nor showing in call logs.

[Eherr]

I have a multi tenant asterisk box where on tenant is receiving calls from the 
caller ID as1as and they cannot pickup the call.

The caller ID also does not show up in the call log. 

Thoughts?

Thanks,
--Eric
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Re: [asterisk-users] Caller ID not real nor showing in call logs.

[Eherr]

Does not show up in the cdr log. 

I am going to enable an asterisk cli dump tonight and try to catch it. 

I am thinking its a straight sip attack or IP attach on the sip client vs a 
real call or problem with asterisk. 

It's also a polycom IP 335

Thanks,
--Eric

Sent from my phone.

 On Jan 8, 2014, at 12:02 PM, Tiago Geada tiago.ge...@gmail.com wrote:
 
 logs ?
 
 full log containing the call?
 
 
 On 8 January 2014 14:56, Eherr email.eherr9...@gmail.com wrote:
 I have a multi tenant asterisk box where on tenant is receiving calls from 
 the caller ID as1as and they cannot pickup the call.
 
 The caller ID also does not show up in the call log.
 
 Thoughts?
 
 Thanks,
 --Eric
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[asterisk-users] fail2ban filter issue

[eherr]

Not sure if this has been answered but I cannot find a solution.

 

I am running Asterisk 1.4.26.3

 

I am seeing the following lines in my log files:

 

A: [2013-03-05 13:54:27] NOTICE[6928] chan_sip.c: Failed to authenticate user 
sip:192.210.138.12;tag=DmVIjOlfYiiL

 

B: [2013-03-05 12:20:00] NOTICE[6928] chan_sip.c: Failed to authenticate user 
101sip:1...@my.asterisk.server.ip;tag=eec630f1

 

C: [2013-03-03 05:15:02] NOTICE[31158] chan_sip.c: Registration from 
'101381sip:101...@my.asterisk.server.ip' failed for
'85.25.23.129' - No matching peer found

 

Now Two Part Question:

 

Part 1: 

 

I understand that line C is from some soft phone like Xlite, IP phone, or 
program trying to register extension 101381 to my server
and the user exten does not exist. I don't understand the method for A or B. I 
don't understand what generates that error message.
Can someone explain?

 

Part 2:

 

Fail2ban blocks line C as per the regex in filters/asterisk.conf. What I don't 
understand is why doesn't lines A or B have a built
in regex line? This goes back to not knowing the method that generates the 
error message in part 1. 

 

Also, can I update the regex in asterisk.conf,  

 

From: NOTICE.* .*: Failed to authenticate user .*@ mailto:.*@%3cHOST%3e.* 
HOST.* 

To:  NOTICE.* .*: Failed to authenticate user .*

 

It should ban both A and B, along with the original Regex line that I modified.

 

Question is, would this present a problem under normal circumstances? I know 
when the line comes up with my.asterisk.server.ip it
will get ignored because I am in the ignoreip list but I want to make sure it 
will be OK to adjust.

 

Thanks community!

 

-E

 

 

 

 

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[asterisk-users] SIP, Polycom, Asterisk - VPN

[eherr]

I am trying to configure the following scenario but have failed.

 

Currently, I have an Asterisk box sitting on a Static Public IP address in my 
office.

 

I have a remote office with 3 Polycom IP335s that are registering back to my 
local office's publically address Asterisk box.

 

The remote office Polycom phones are getting IP information from an RV042 and 
using the local ISP for internet access.

 

I want to set up a VPN on the remote side.

 

Has anyone done this? Does it make sense to do this?

 

Thanks,

--E

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Re: [asterisk-users] SIP, Polycom, Asterisk - VPN

[eherr]

Thanks for the reply!

 

Why IAX over SIP?

 

In what environment/setup are you using a VPN for the phones?

 

--E

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Wednesday, October 03, 2012 10:59 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SIP, Polycom, Asterisk - VPN

 

The easiest way to accomplish this is probably going to be to set up an 
asterisk server in the remote office and just use IAX to
talk between the two boxes. We do VPN here for two phones but I can't really 
tell you all that you need to know.

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr
Sent: Wednesday, October 03, 2012 9:22 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] SIP, Polycom, Asterisk - VPN

 

I am trying to configure the following scenario but have failed.

 

Currently, I have an Asterisk box sitting on a Static Public IP address in my 
office.

 

I have a remote office with 3 Polycom IP335s that are registering back to my 
local office's publically address Asterisk box.

 

The remote office Polycom phones are getting IP information from an RV042 and 
using the local ISP for internet access.

 

I want to set up a VPN on the remote side.

 

Has anyone done this? Does it make sense to do this?

 

Thanks,

--E

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[asterisk-users] Timeout for Huntgroup

[eherr]

I have not found a solution so I am checking with the Masses here.

 

I have a client who has a old 5 line key system without voicemail.

 

Currently, I can set up a huntgroup and ring each line for 15 seconds and after 
the 5th line has reached its limit, the call goes to
voicemail.

 

The problem with this is that the caller will hear roughly 1m15s worth of 
ringing.

 

I trying to give them the same ability to always ring the huntgroup, starting 
with line 1, and hunting only if its busy or the
timeout is reached.

 

The modification needed is to only ring the huntgroup for 25 seconds ( roughly 
5 rings ) and then pull the call back and send it to
voicemail if the line was not answered.

 

Has anyone seen this feature created and implemented and if so, how?

 

Thanks,

--eherr 

 

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[asterisk-users] Least Machine Specs to run a production asterisk server

[eherr]

What is the lowest end machine to run a production asterisk server.

 

Currently I have an MTE version running on a $2500 server.

 

I want to get a single Tenant asterisk server for one company but don't want to 
waste a $2500 server for one tenant.

 

Do you guys have any recommendations for a unix machine specs and 
pricing..granted everything is rough estimates.

 

Thanks,

--E

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Re: [asterisk-users] Least Machine Specs to run a production asterisk server

[eherr]

Well..

 

I would say that there would be about 40 Polycom Soundpoint IP335s

 

G711u

 

I would like to restrict this to say 15 inbound lines and 40 outbound lines.

 

Probably looking at 8 concurrent calls.

 

But I need it to be reliable machine since it will be a production server.

 

Thanks,

--E

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez
Sent: Friday, May 11, 2012 10:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Least Machine Specs to run a production asterisk 
server

 

On Fri, May 11, 2012 at 6:15 AM, eherr email.eherr9...@gmail.com wrote:

What is the lowest end machine to run a production asterisk server.

 

Depends on a lot of variables.  I've got some old 1.8GHz 1U servers running 
hundreds of calls.

 

How many calls, how much transcoding, etc etc.

 

You can run Asterisk on a Linksys WRT router, so I guess that's about a minimum.

 

-- 

Carlos Alvarez

TelEvolve

602-889-3003

 

 

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[asterisk-users] SLA - Shared Line Appearance - Polycom

[eherr]

Has anyone been able to get SLA ( Shared Line Appearance ) to work with Polycom 
phones.

I want to be able to emulate a key system but I cannot figure it out.

Everything I tried so far is just not working together.

Thanks,
--E


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[asterisk-users] SLA - Shared Line Appearance - Polycom

[eherr]

Has anyone been able to get SLA ( Shared Line Appearance ) to work with Polycom 
phones.

 

I want to be able to emulate a key system but I cannot figure it out.

 

Everything I tried so far is just not working together.

 

Thanks,

--E

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Re: [asterisk-users] SLA - Shared Line Appearance - Polycom

[eherr]

Oh yeah, damn small things.

 

Asterisk 1.8.7.1

Polycom IP650

 

Thanks,

--E

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, May 10, 2012 10:47 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom

 

Which flavor of Asterisk and what model Polycom?  Both are factors in the 
possible success of the task.

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr
Sent: Thursday, May 10, 2012 9:45 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] SLA - Shared Line Appearance - Polycom

 

Has anyone been able to get SLA ( Shared Line Appearance ) to work with Polycom 
phones.

 

I want to be able to emulate a key system but I cannot figure it out.

 

Everything I tried so far is just not working together.

 

Thanks,

--E

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Re: [asterisk-users] SLA - Shared Line Appearance - Polycom

[eherr]

Yes but I am not able to follow it to completion.

 

I think the guide is quite incomplete.

 

I am also trying to follow another guide which leaves things out.

 

I was hoping that someone has it working so they can tell me exactly what is 
missing and needs to be done.

 

--E

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, May 10, 2012 2:18 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom

 

You read this? - 
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/SLA.html

 

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr
Sent: Thursday, May 10, 2012 12:00 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom

 

Oh yeah, damn small things.

 

Asterisk 1.8.7.1

Polycom IP650

 

Thanks,

--E

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, May 10, 2012 10:47 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom

 

Which flavor of Asterisk and what model Polycom?  Both are factors in the 
possible success of the task.

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr
Sent: Thursday, May 10, 2012 9:45 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] SLA - Shared Line Appearance - Polycom

 

Has anyone been able to get SLA ( Shared Line Appearance ) to work with Polycom 
phones.

 

I want to be able to emulate a key system but I cannot figure it out.

 

Everything I tried so far is just not working together.

 

Thanks,

--E

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Re: [asterisk-users] SLA - Shared Line Appearance - Polycom

[eherr]

Unfortunately I am still a sitting duck.

 

Any old configs you can show me?

 

Its driving me insane.

 

I am trying to set it up in this fashion:

Everything is SIP based. I do not have ZAP/Dahdi/etc.

6 IP650 phones.

line keys 3,4,5,6 will be emulating the Key System with line keys 1 and 2 be 
set to have their extension.

A call comes in, it rings SLA_line 1 ( line key 3) on all phones. If I pick up, 
that line is now active on all phones. If another
call comes in, it goes to SLA_Line 2 ( line key 4 ). If I put SLA_line1 on 
hold, it shows on hold for all phones. A coworker then
picks up SLA_line1 and continues to the conversation from their phone.

I do not necessarily care if they can call out on the SLA_line(n)s since they 
will have line key 1 and 2 set to their personal
extension.

-E

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, May 10, 2012 2:34 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom

 

I was able to do it with 1.4 and an IP501. You just get your hints set up 
correctly and life should be good.  You set up the hints
as contacts in the directory.

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr
Sent: Thursday, May 10, 2012 1:31 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom

 

Yes but I am not able to follow it to completion.

 

I think the guide is quite incomplete.

 

I am also trying to follow another guide which leaves things out.

 

I was hoping that someone has it working so they can tell me exactly what is 
missing and needs to be done.

 

--E

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, May 10, 2012 2:18 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom

 

You read this? - 
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/SLA.html

 

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr
Sent: Thursday, May 10, 2012 12:00 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom

 

Oh yeah, damn small things.

 

Asterisk 1.8.7.1

Polycom IP650

 

Thanks,

--E

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, May 10, 2012 10:47 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom

 

Which flavor of Asterisk and what model Polycom?  Both are factors in the 
possible success of the task.

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr
Sent: Thursday, May 10, 2012 9:45 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] SLA - Shared Line Appearance - Polycom

 

Has anyone been able to get SLA ( Shared Line Appearance ) to work with Polycom 
phones.

 

I want to be able to emulate a key system but I cannot figure it out.

 

Everything I tried so far is just not working together.

 

Thanks,

--E

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Re: [asterisk-users] SLA - Shared Line Appearance - Polycom

[eherr]

Would those be the shared lines?

 

Continuing with the assumption they are, I need to set them up in stations in 
the SLA.conf file?

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, May 10, 2012 2:44 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom

 

Since you are doing all SIP, you would want something like this:

exten = 1000,hint,SIP/100

exten = 2000,hint,SIP/200

exten = 3000,hint,SIP/300

exten = 4000,hint,SIP/400

 

Then set up your lines to look for 1000@default, 2000@default, etc.

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr
Sent: Thursday, May 10, 2012 1:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom

 

Unfortunately I am still a sitting duck.

 

Any old configs you can show me?

 

Its driving me insane.

 

I am trying to set it up in this fashion:

Everything is SIP based. I do not have ZAP/Dahdi/etc.

6 IP650 phones.

line keys 3,4,5,6 will be emulating the Key System with line keys 1 and 2 be 
set to have their extension.

A call comes in, it rings SLA_line 1 ( line key 3) on all phones. If I pick up, 
that line is now active on all phones. If another
call comes in, it goes to SLA_Line 2 ( line key 4 ). If I put SLA_line1 on 
hold, it shows on hold for all phones. A coworker then
picks up SLA_line1 and continues to the conversation from their phone.

I do not necessarily care if they can call out on the SLA_line(n)s since they 
will have line key 1 and 2 set to their personal
extension.

-E

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, May 10, 2012 2:34 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom

 

I was able to do it with 1.4 and an IP501. You just get your hints set up 
correctly and life should be good.  You set up the hints
as contacts in the directory.

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr
Sent: Thursday, May 10, 2012 1:31 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom

 

Yes but I am not able to follow it to completion.

 

I think the guide is quite incomplete.

 

I am also trying to follow another guide which leaves things out.

 

I was hoping that someone has it working so they can tell me exactly what is 
missing and needs to be done.

 

--E

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, May 10, 2012 2:18 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom

 

You read this? - 
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/SLA.html

 

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr
Sent: Thursday, May 10, 2012 12:00 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom

 

Oh yeah, damn small things.

 

Asterisk 1.8.7.1

Polycom IP650

 

Thanks,

--E

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, May 10, 2012 10:47 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom

 

Which flavor of Asterisk and what model Polycom?  Both are factors in the 
possible success of the task.

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr
Sent: Thursday, May 10, 2012 9:45 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] SLA - Shared Line Appearance - Polycom

 

Has anyone been able to get SLA ( Shared Line Appearance ) to work with Polycom 
phones.

 

I want to be able to emulate a key system but I cannot figure it out.

 

Everything I tried so far is just not working together.

 

Thanks,

--E

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Re: [asterisk-users] SLA - Shared Line Appearance - Polycom

[eherr]

Then on my polycom what are the configurations?

 

Is it with the line keys such as reg.1 be my extension with line key set to 2 
and reg.2 be 1000 ( SLA line 1 ) reg.3 = 200 ( SLA
line 2 )

 

Or is this set up in the bottom part of the configs under the attendant section?

 

--E

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, May 10, 2012 2:55 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom

 

Correct - the sla.conf makes the line active the hint makes it accessible by 
the phone.

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr
Sent: Thursday, May 10, 2012 1:51 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom

 

Would those be the shared lines?

 

Continuing with the assumption they are, I need to set them up in stations in 
the SLA.conf file?

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, May 10, 2012 2:44 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom

 

Since you are doing all SIP, you would want something like this:

exten = 1000,hint,SIP/100

exten = 2000,hint,SIP/200

exten = 3000,hint,SIP/300

exten = 4000,hint,SIP/400

 

Then set up your lines to look for 1000@default, 2000@default, etc.

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr
Sent: Thursday, May 10, 2012 1:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom

 

Unfortunately I am still a sitting duck.

 

Any old configs you can show me?

 

Its driving me insane.

 

I am trying to set it up in this fashion:

Everything is SIP based. I do not have ZAP/Dahdi/etc.

6 IP650 phones.

line keys 3,4,5,6 will be emulating the Key System with line keys 1 and 2 be 
set to have their extension.

A call comes in, it rings SLA_line 1 ( line key 3) on all phones. If I pick up, 
that line is now active on all phones. If another
call comes in, it goes to SLA_Line 2 ( line key 4 ). If I put SLA_line1 on 
hold, it shows on hold for all phones. A coworker then
picks up SLA_line1 and continues to the conversation from their phone.

I do not necessarily care if they can call out on the SLA_line(n)s since they 
will have line key 1 and 2 set to their personal
extension.

-E

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, May 10, 2012 2:34 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom

 

I was able to do it with 1.4 and an IP501. You just get your hints set up 
correctly and life should be good.  You set up the hints
as contacts in the directory.

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr
Sent: Thursday, May 10, 2012 1:31 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom

 

Yes but I am not able to follow it to completion.

 

I think the guide is quite incomplete.

 

I am also trying to follow another guide which leaves things out.

 

I was hoping that someone has it working so they can tell me exactly what is 
missing and needs to be done.

 

--E

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, May 10, 2012 2:18 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom

 

You read this? - 
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/SLA.html

 

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr
Sent: Thursday, May 10, 2012 12:00 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom

 

Oh yeah, damn small things.

 

Asterisk 1.8.7.1

Polycom IP650

 

Thanks,

--E

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, May 10, 2012 10:47 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom

 

Which flavor of Asterisk and what model Polycom?  Both are factors in the 
possible success of the task.

 

From: asterisk-users-boun

[asterisk-users] fail2ban restarts

[eherr]

I have fail2ban running on my Asterisk box.

 

Every so often I receive emails stating that the jails stopped and then started.

 

Why does this happen?

 

Why isn't it just continuously running?

 

Thanks,

-E

 

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Re: [asterisk-users] Sip Registration Hijacking

[eherr]

It is accessible from HTTP.

However, the access list only allows access from my home and the password is 
strong.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Hayes
Sent: Thursday, January 26, 2012 10:30 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Sip Registration Hijacking

On 20/01/12 01:36, eherr wrote:

 It is also register on an AudioCodes MP-118.

 Thanks,

 -E

Is the Audiocodes gateway accessible online?  Have you set a strong 
password for it's web interface (and cli if it has one)?  It is possible 
someone is breaking into that and getting the SIP password out of it.

cheers,
Paul.

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Re: [asterisk-users] Sip Registration Hijacking

[eherr]

Can you please elaborate on rate limiting. Not how to implement it but rather 
how implementation is beneficiary.

 

Reading up on it, it appears that it just checks the tcp connections and denys 
connection if limit is passed.

 

In my thoughts, this is essentially a live fail2ban monitor in respects to 
attempted authentications. 

 

Thanks,

--E

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim DeVito
Sent: Saturday, January 21, 2012 12:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Sip Registration Hijacking

 

Rate limiting (google) via iptables FTW! Good luck! 

- Original message - 

 
 Alejandro Imass wrote 20.01.2012 18:09: 
 
  I would like to know how 
 to block this MF because he makes calls at 1-2 AM 
 
 I use this 
 construction on my servers 
 
 [users] 
 
 exten = 
 _XXX,1,GotoIfTime(1:00-2:00,*,*,*?block,1,1) 
 
 [block] 
 exten = 
 _X.,1,HangUp(1) 
 
 -- 
 With Best Regards 
 Mikhail Lischuk 
 


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Re: [asterisk-users] Sip Registration Hijacking

[eherr]

This is actually an interesting concept however I do think I want to restrict 
dialing during a specific time period.

 

If someone is in the office, I would have to reprogram the route so allow 
dialing which adds overhead.

 

Again, I do like the concept though.

 

Thanks,

--E

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mikhail Lischuk
Sent: Friday, January 20, 2012 7:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Sip Registration Hijacking

 

Alejandro Imass wrote 20.01.2012 18:09:

 I would like to know how to block this MF because he makes calls at 1-2 AM

I use this construction on my servers

[users]

exten = _XXX,1,GotoIfTime(1:00-2:00,*,*,*?block,1,1)

 

[block]
exten = _X.,1,HangUp(1)

 

-- 
With Best Regards
Mikhail Lischuk mailto:mlisc...@itx.com.ua 
 
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Re: [asterisk-users] Sip Registration Hijacking

[eherr]

I appreciate your 2-cents worth.

 

However, I do not believe they have access to machine

 

If so, they are clever to create three failures in the logs for my benefit 
before entering the correct one for hijacking.

 

Additionally, I have a lot of sip extensions to hijack and he keeps going for 
the same one.

 

I was hoping this was something with the MP-118 and someone experienced the 
same thing with that device.

 

Either way, I posed two questions which are still unanswered and probably I 
will never get answered: 

1 - is this a vulnerability in the MP-118

2 - what method could they possibly be using to hijack a number-alpha extension 
which is creative to begin with ie)
203-Joes_Insurance_Service with an openssl generated password of 12 characters.

 

Thanks,

--E

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Larry Moore
Sent: Saturday, January 21, 2012 1:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Sip Registration Hijacking

 

On 20/01/2012 9:36 AM, eherr wrote: 

I have a honey pot box with extensions that are not just numbers ie )

 

100-MySipUserName

 

And the passwords are from an openssl generated password ie)

 

Gq5VNIjDFWIQoUT6

 

 


Is the password stored in sip.conf in plain text or as an MD5?

If it is stored in plain text then it may suggest the hijacker has greater 
access to your system than you realise.

My 2-cents worth.

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[asterisk-users] Cordless SIP phone

[eherr]

I have an asterisk box which has Polycom Soundpoints IP335 and IP650s 
registering to it both locally and remote.

 

I want to be able to incorporate a cordless phone at the remote location; not a 
wireless phone.

 

I want it to also be able to register to the same asterisk box so it can take 
calls and transfers.

 

Any idea?

 

Thanks,

--E

 

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Re: [asterisk-users] Cordless SIP phone

[eherr]

To be honest I am not sure.

 

I was under the impression that there was a cordless SIP phone that 
communicated back to the base which was hardwired into the
network.

 

Anything like that?

 

--E

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez
Sent: Monday, January 23, 2012 10:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cordless SIP phone

 

On Mon, Jan 23, 2012 at 7:35 AM, eherr email.eherr9...@gmail.com wrote:

I have an asterisk box which has Polycom Soundpoints IP335 and IP650s 
registering to it both locally and remote.

 

I want to be able to incorporate a cordless phone at the remote location; not a 
wireless phone.

 

I want it to also be able to register to the same asterisk box so it can take 
calls and transfers.

 

 

What is the difference between a cordless phone and a wireless one?

 

We use and recommend the Panasonic KX-TGP500 and 550.


 

-- 

Carlos Alvarez

TelEvolve

602-889-3003

 

 

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Re: [asterisk-users] Cordless SIP phone

[eherr]

I have one of these already installed. 

I have a plain household cordless phone plugged into an SPA which then SIPs 
back to the server.

Where I want to put the new on is outside the range.

I thought SIP cordless phones would be better on the range.

Either way, does the two mentioned:

Gigaset
Panasonic KX-TGP500

Allow to program an extension from the asterisk so it is integrated into the 
phone system as if it were just another Polycom
extension?

Thanks,
--E

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A J Stiles
Sent: Monday, January 23, 2012 10:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cordless SIP phone

On Monday 23 January 2012, eherr wrote:
 I have an asterisk box which has Polycom Soundpoints IP335 and IP650s
 registering to it both locally and remote.
  
 I want to be able to incorporate a cordless phone at the remote location;
 not a wireless phone.
  
 I want it to also be able to register to the same asterisk box so it can
 take calls and transfers.
  
 Any idea?

Some sort of FXS adaptor  (Grandstream HandyTone 286 is still available)  and 
a generic cordless phone from any electronics store?  You may also need an 
adaptor with an RJ11 plug and whatever phone socket your local telephone 
company use; and in some countries this will need to be a master type  (with 
bell capacitor),  or else the phone may not ring.  (A half-master type -- 
which has the bell capacitor but lacks a surge arrester -- will also work 
fine.)

-- 
AJS

Answers come *after* questions.

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Re: [asterisk-users] Sip Registration Hijacking

[eherr]

I always thought Sip Vicious only does numbers ( 0 - 100 ) not 
Numberic-Alpha ( 100-MySipUserName ).

To make my situation more interesting is that I also have fail2ban installed 
banning after 5 failed attempts.

This hijack is only happening to an extension on the honeypot audiocodes with 
the sip reg authenticating back to my honey pot
asterisk which is why I thought it might be a vulnerability in the audiocodes.

However, the hijacker manages to make it past the fail2ban and gets the sip reg.

I see sipvicious attempts all the time where they run checks against extensions 
0 - . 

Sometimes I see alpha extension name attempts but I do not know how that's done.

--E

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro Imass
Sent: Friday, January 20, 2012 11:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Sip Registration Hijacking

On Thu, Jan 19, 2012 at 8:36 PM, eherr email.eherr9...@gmail.com wrote:
 I have a honey pot box with extensions that are not just numbers ie )



 100-MySipUserName




I have the same problem and I use contactpermit with specific ip blocks!

I know for a fact I'm getting hijacked by sip vicious on extension 100
but I can't understand how because I don't even have an extension 100
declared anywhere. I would like to know how to block this MF because
he makes calls at 1-2 AM

-- 
Alejandro Imass

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[asterisk-users] Sip Registration Hijacking

[eherr]

I have a honey pot box with extensions that are not just numbers ie )

 

100-MySipUserName

 

And the passwords are from an openssl generated password ie)

 

Gq5VNIjDFWIQoUT6

 

However, this one extension keeps getting hacked and showing up on a different 
IP address.

 

It is also register on an AudioCodes MP-118.

 

I wanted to know if anyone else ran into this and if it's a vulnerability on 
the MP-118 or with Asterisk.

 

Thanks,

-E

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Re: [asterisk-users] random digits dialing during call

[eherr]

What are you using for hardware?

I have experienced SPA2102s supplying a DTMF tone when someone was talking.

This was caused by the talker reaching a certain frequency while talking in 
which the SPA popped out a DTMF tone.

I haven't experienced this behavior on polycoms or anything else.

--E

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Skyler
Sent: Thursday, December 08, 2011 9:27 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] random digits dialing during call

Hi List,

 When a user is on a call, sometimes they hear digits dialing as if the
other end is randomly pressing the keypad with their face...but they
aren't. It has happened while I've been on calls also, very odd and
annoying.

 Has anyone come across this on Asterisk before?

TIA,
Skyler


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Re: [asterisk-users] Issue with Polycom SPIP650 and its sidecar

[eherr]

When the side car looses it entries, what does the config file show for the 
entries. 

 

This happened to me one time but that was only because for some reason, the 
contacts file was deleted by accident and I had to
recreate it. ( I have a backup now too! )

 

It probably as Dan said, check the firmware and get the latest. Make sure your 
sidecar is pushed down all the way. Possible get a
new IP650 and see if it exhibits the same behavior.

 

I am pretty positive its not asterisk related.

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Wednesday, November 30, 2011 10:32 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Issue with Polycom SPIP650 and its sidecar

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Wednesday, November 30, 2011 9:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Issue with Polycom SPIP650 and its sidecar

 

Hello,

On one location, I've got from time to time (let say one a week) the following 
issue :
the phone SoundPoint 650 works ok (can call or answer, display and sound are 
ok),
the sidecar looses its display : entries on sidecar's LCD screen are not 
displayed anymore, or names are truncated, or BLF are not
shown or updated.

I only have one SPIP650 on this system so I can't compare with others.

What could be the root cause of this ?

Regards

 

I doubt it is an Asterisk related issue, but what version of Asterisk is the 
phone running on?  The most likely probabilities as I
see them are (1) network or power issue (2) hardware problem with the phone (3) 
provisioning issue with the phone (bad sip.ld or
such).  You could investigate /var/log/asterisk/full to see what kind of call 
activity occurs around these failures.

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[asterisk-users] # of Polycoms on a DSL line?

[eherr]

Out of curiosity, how many concurrent phone calls for an office that uses 
Polycoms could be sustained on a DSL ( 3meg down, 768 up )
line using g711?

 

Not sure if its 64kbps or 87kbps.

 

I would say roughly 8 but I don't know if the polycoms add any more payload to 
the network for presence and all that jazz.

 

Thanks,

--E

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Re: [asterisk-users] sip show peers

[eherr]

I believe it is set by a character length for formatting the output.

What are you trying to accomplish? Are you just viewing it in the CLI or are 
you writing monitoring scripts?

Can you switch names so that they are unique in the beginning?

--E

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Tuesday, November 22, 2011 10:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] sip show peers

  Is there a way with the command (1.4.42) for sip show peers to
see the FULL Name/Username field???

I have long names and mine are being truncated.

Thanks

Jerry

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[asterisk-users] Polycom Phantom Ringing

[eherr]

I have a Polycom Soundpoint IP335.

 

There are no inbound routes set to the phones yet.

 

However, the phones are getting phantom rings.

 

What is the legitimacy of these calls? 

 

Is there something I need to block to stop it?

 

I believe its people trying to hack the phones/phone system but I cannot find 
where I read that before.

 

Thanks,

--E

 

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Re: [asterisk-users] Polycom Phantom Ringing

[eherr]

Well this is a remote site.

I am running 1.4.26

I have multiple polycoms that do not experience this.

They are getting dhcp from their local router.

I am wondering if it could either be a bug in the polycom firmware or something 
like a probe into the phone.

I am pretty sure I read that phantom rings are sip calls to a phone where they 
are probing for extensions or something; cant
remember.

--E


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis
Sent: Friday, November 18, 2011 9:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom Phantom Ringing

Nick.

On Fri, Nov 18, 2011 at 9:38 AM, Danny Nicholas da...@debsinc.com wrote:
 If your phones are being “hacked” you have a firewall problem.  Your phones
 should only be registering to your local DHCP server and your Asterisk box.

 DHCP Server 192.X.X.X

 Asterisk Server 192.X.Y.Y

 Phone 192.X.Z.Z



 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr
 Sent: Friday, November 18, 2011 8:34 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: [asterisk-users] Polycom Phantom Ringing



 I have a Polycom Soundpoint IP335.



 There are no inbound routes set to the phones yet.



 However, the phones are getting phantom rings.



 What is the legitimacy of these calls?



 Is there something I need to block to stop it?



 I believe its people trying to hack the phones/phone system but I cannot
 find where I read that before.



 Thanks,

 --E



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I don't think that's it. I have had the Polycom on my test site
ring unexpectadly. It could be a bug in asterisk. Are you using
1.8?


Nick

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Re: [asterisk-users] Polycom Phantom Ringing

[eherr]

They do.

It shows up asterisk on the physical phone.

Nothing in the raw cdr file on the server.

--E

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Friday, November 18, 2011 10:16 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Polycom Phantom Ringing

I have Polycom 501's and they keep a log of all calls.  I would expect the
335's to have that capability as well.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr
Sent: Friday, November 18, 2011 8:50 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Polycom Phantom Ringing

Well this is a remote site.

I am running 1.4.26

I have multiple polycoms that do not experience this.

They are getting dhcp from their local router.

I am wondering if it could either be a bug in the polycom firmware or
something like a probe into the phone.

I am pretty sure I read that phantom rings are sip calls to a phone where
they are probing for extensions or something; cant remember.

--E


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis
Sent: Friday, November 18, 2011 9:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom Phantom Ringing

Nick.

On Fri, Nov 18, 2011 at 9:38 AM, Danny Nicholas da...@debsinc.com wrote:
 If your phones are being “hacked” you have a firewall problem.  Your 
 phones should only be registering to your local DHCP server and your
Asterisk box.

 DHCP Server 192.X.X.X

 Asterisk Server 192.X.Y.Y

 Phone 192.X.Z.Z



 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr
 Sent: Friday, November 18, 2011 8:34 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: [asterisk-users] Polycom Phantom Ringing



 I have a Polycom Soundpoint IP335.



 There are no inbound routes set to the phones yet.



 However, the phones are getting phantom rings.



 What is the legitimacy of these calls?



 Is there something I need to block to stop it?



 I believe its people trying to hack the phones/phone system but I 
 cannot find where I read that before.



 Thanks,

 --E



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   http://lists.digium.com/mailman/listinfo/asterisk-users


I don't think that's it. I have had the Polycom on my test site ring
unexpectadly. It could be a bug in asterisk. Are you using 1.8?


Nick

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Re: [asterisk-users] 10.0.0-rc1: dahdi doesn't see card

[eherr]

I am running into an issue installing asterisk 10.0.0-rc1

 

I have centos  2.6.18-194.el5 #1 SMP Fri Apr 2 14:58:35 EDT 2010 i686 i686 i386 
GNU/Linux installed.

 

I am at the point of trying to install the dahdi and I am getting the error 
message when I do a make all:

You do not appear to have the sources for the 2.6.18-194.el5 kernel installed.

 

Any ideas?

 

Thanks,

--E

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James zhu
Sent: Sunday, November 13, 2011 8:19 PM
To: asterisk-user
Subject: Re: [asterisk-users] 10.0.0-rc1: dahdi doesn't see card

 

hi:

make sure chan_dahdi.conf with the data and stop now, restart asterisk.

Best regards,
James.zhu
Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP).
website: www.voipviews.com 



 From: ewiel...@nyigc.com
 To: asterisk-users@lists.digium.com
 Date: Fri, 11 Nov 2011 19:38:56 -0500
 Subject: Re: [asterisk-users] 10.0.0-rc1: dahdi doesn't see card
 
 Show us /etc/asterisk/chan_dahdi.conf (and any #include'd files)
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
 Sent: Friday, November 11, 2011 5:51 PM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] 10.0.0-rc1: dahdi doesn't see card
 
 From asterisk -cv
 
 == Parsing '/etc/asterisk/chan_dahdi.conf': == Found
 == Parsing '/etc/asterisk/users.conf': == Found
 -- Automatically generated pseudo channel
 [Nov 11 17:43:28] WARNING[5756]: chan_dahdi.c:18155 process_dahdi: 
 Ignoring any changes to 'userbase' (on reload) at line 23.
 [Nov 11 17:43:28] WARNING[5756]: chan_dahdi.c:18155 process_dahdi: 
 Ignoring any changes to 'vmsecret' (on reload) at line 31.
 [Nov 11 17:43:28] WARNING[5756]: chan_dahdi.c:18155 process_dahdi: 
 Ignoring any changes to 'hassip' (on reload) at line 35.
 [Nov 11 17:43:28] WARNING[5756]: chan_dahdi.c:18155 process_dahdi: 
 Ignoring any changes to 'hasiax' (on reload) at line 39.
 [Nov 11 17:43:28] WARNING[5756]: chan_dahdi.c:18155 process_dahdi: 
 Ignoring any changes to 'hasmanager' (on reload) at line 47.
 == Registered channel type 'DAHDI' (DAHDI Telephony Driver)
 == Manager registered action DAHDITransfer
 == Manager registered action DAHDIHangup
 == Manager registered action DAHDIDialOffhook
 == Manager registered action DAHDIDNDon
 == Manager registered action DAHDIDNDoff
 == Manager registered action DAHDIShowChannels
 == Manager registered action DAHDIRestart
 chan_dahdi.so = (DAHDI Telephony Driver)
 ..
 
 But, no dahdi.
 
 dahdi show channels shows nothing:
 
 Chan Extension Context Language MOH Interpret 
 Blocked State Description
 pseudo default default 
 In Service
 
 
 dahdi_cfg -vvv
 DAHDI Tools Version - 2.4.1
 
 DAHDI Version: 2.5.0.1
 Echo Canceller(s): HWEC, OSLEC
 Configuration
 ==
 
 
 Channel map:
 
 Channel 01: FXO Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 01)
 Channel 02: FXO Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 02)
 Channel 04: FXS Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 04)
 
 3 channels to configure.
 
 Setting echocan for channel 1 to oslec
 Setting echocan for channel 2 to oslec
 Setting echocan for channel 4 to oslec
 
 From dmesg:
 dahdi: Telephony Interface Registered on major 196
 [ 9.849036] dahdi: Version: 2.5.0.1
 [ 9.899531] ACPI: PCI Interrupt Link [LNKC] enabled at IRQ 19
 [ 9.899919] wctdm :01:0a.0: PCI INT A - Link[LNKC] - GSI 19 
 (level, low) - IRQ 19
 [ 9.901845] Freshmaker version: 73
 [ 9.902565] Freshmaker passed register test
 [ 11.287170] Module 0: Installed -- AUTO FXS/DPO
 [ 12.172362] Module 1: Installed -- AUTO FXS/DPO
 [ 12.172795] Module 2: Not installed
 [ 12.373273] Module 3: Installed -- AUTO FXO (FCC mode)
 [ 12.374071] Found a Wildcard TDM: Wildcard TDM400P REV I (3 modules)
 
 Any help appreciated.
 
 sean
 
 
 
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Re: [asterisk-users] Becoming a CLEC

[eherr]

But what is the correct physical setup of a CLEC.

Do you get rack space at a carrier hotel and equipment in there?

Do you get rack space at the local ILEC CO?; which is Verizon here.

What are the types of voice platforms used by CLECs?

Thanks,
--E

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov
Sent: Tuesday, November 15, 2011 12:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Becoming a CLEC

There are clever ways to be a CLEC, and keen reasons for becoming so.  But 
cheaper stuff ain't one of them.

--
This message was painstakingly thumbed out on my mobile, so apologies for 
brevity, errors, and general sloppiness.

Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/

On Nov 14, 2011, at 10:02 PM, Nick Khamis sym...@gmail.com wrote:

 Yeah! That is what I was thinking... Bringing Voice and Video under
 one umbrella, things like that...
 I actually come from a speech recognition and natural language
 processing background. Trying to
 build the voice network, and seeing how I can bring it all together.
 
 P.S. I started by getting acquainted with the proxies of course ;)
 
 Nick
 
 On Mon, Nov 14, 2011 at 9:42 PM, Alex Balashov
 abalas...@evaristesys.com wrote:
 Only through new, innovative applications. They will always deliver 
 transport and dialtone cheaper than you.
 
 --
 This message was painstakingly thumbed out on my mobile, so apologies for 
 brevity, errors, and general sloppiness.
 
 Alex Balashov - Principal
 Evariste Systems LLC
 260 Peachtree Street NW
 Suite 2200
 Atlanta, GA 30303
 Tel: +1-678-954-0670
 Fax: +1-404-961-1892
 Web: http://www.evaristesys.com/
 
 On Nov 14, 2011, at 9:15 PM, Nick Khamis sym...@gmail.com wrote:
 
 Hahah! Yeah it does doesn't it? What do we do? How do we stay
 a float, It almost seems like the ILECs will drop their rates to a
 penny once the people in this, and Kamailio lists ;) actually put a
 dent in their underline.
 
 Nick
 
 On Mon, Nov 14, 2011 at 9:08 PM, Jeff LaCoursiere j...@sunfone.com wrote:
 On Mon, 2011-11-14 at 20:51 -0500, Nick Khamis wrote:
 The ride is over before it even began A local ILEC here in Canada,
 is already offering
 Unlimited World service. And this on a Tier 1 network, not the crap
 we're use to doing
 business on. Choose a different angle before you get anymore grey
 hairs on that head...
 
 http://www.bell.ca/Home_phone/Products/Unlimited_World?INT=HP_hpldpg_BAN_flatfee_Mass_20101110_cb_on_en
 
 
 
 The Unlimited service seems pretty limited to me.  Vonage may even
 have more reach than this.
 
 j
 
 
 
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Re: [asterisk-users] Skype For Asterisk (SFA)

[eherr]

I would agree, unfortunately.

However, I still see it as a glorified webcam chat and not a telecommunication 
device like a SIP/soft phone.



-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gordon Henderson
Sent: Wednesday, November 16, 2011 11:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Skype For Asterisk (SFA)

On Wed, 16 Nov 2011, A J Stiles wrote:

 You would be better off persuading Skype users to transition to something 
 else.

 Skype is the absolute antithesis of the whole point of telephony, which is to
 connect people together.  This includes, implicitly, the ability for
 subscribers on one telecommunications provider's network to call subscribers
 on another network.  Imagine if, say, Vodafone subscribers were unable to call
 up BT subscribers?  Well, this is *exactly* what Skype are trying to create by
 keeping their protocols proprietary.

 Of course there remains a small but finite probability that Skype will be
 successfully reverse-engineered, the Source Code leaked, or Skype's owners
 forced to publish its communications protocols before the 2013 deadline.  But
 it would be extreme folly to bet the family farm on this happening.

 It's time to start seriously evaluating Asterisk-compatible alternatives to
 Skype.

Sadly, my experience in the SOHO environment is that Skype wins.

I tried to get my family to all use SIP videophones - and it worked for a 
couple of years - mostly. The downside was that they're mostly using crap 
domestic quality broadband and trying to use a videophone, or even a 
soft-phone on a PC just seemed too hard for them to grasp. They *all* 
moved to Skype recently - and I have to say I've been totally blown away 
at the ease of use and the quality of the calls - both sound and video. 
(And I'm using Linux too)

The other thing - LAN to LAN calls STAY ON THE LAN! So I can Skype my 
wife next door and it doesn't use up any of my own broadband bandwidth 
wheras if I use a hosted SIP service, calls go out  come back in again. 
Skype also seems to be able to run the lines at max. rate too - some sort 
of adaptive bandwidth - we get large and high resolution video calls from 
one end of the country to the other with the output bandwidth running at 
near max (800Kb sec in our case)

And now I'm seeing some of my smaller business customers using Skype. For 
serious business calls too. It's free. They get video. It just works. No 
fiddling with NAT, port forwarding, never any hint of one-way audio.

I really was skeptical at first, but Skype is here to stay - mostly 
because it just works. Even a complete computer idiot can install it and 
make it work. Give them a SIP phone, or SIP softphone and tell them to set 
it up and they'll just leave it alone as too complicated.

As for interoerability - well there's Skype-Out. It works, it's set at a 
reasonable price level, so what more do you need?

Once upon a time I would block Skype from working inside a corporate LAN 
and would recomend against it's use - now I'm told to explicitly allow it.

Times are changing and I'm finding it harder to persuade small businesses 
to use SIP phones - and why should they...

Gordon

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[asterisk-users] polycom soundpint ip650 question

[eherr]

On the polycom soundpoint ip 650 six line phone:

 

Say I have 4 lines on hold, is there way to tell who I put on hold.

 

I cannot see the caller ID of the other lines, only the last line I placed on 
hold.

 

Thanks,

--E

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[asterisk-users] Polycom Attended Transfer

[eherr]

When you perform an attended transfer, the extension of the person transferring 
is displayed to the co-worker.

 

Can I override the caller ID to display the caller's callerID during a blind 
transfer?

 

Thanks,

--E

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Re: [asterisk-users] polycom soundpint ip650 question

[eherr]

Thanks for the response.

 

What you described is for the CLI.

 

I am asking is there a way on the phone itself or is there a phone that does 
have this capability.

 

Thanks,

--E

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Wednesday, November 16, 2011 1:49 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] polycom soundpint ip650 question

 

Core show channels verbose - if you do asterisk -rx cscv from bash

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr
Sent: Wednesday, November 16, 2011 12:06 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] polycom soundpint ip650 question

 

On the polycom soundpoint ip 650 six line phone:

 

Say I have 4 lines on hold, is there way to tell who I put on hold.

 

I cannot see the caller ID of the other lines, only the last line I placed on 
hold.

 

Thanks,

--E

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Re: [asterisk-users] Polycom Attended Transfer

[eherr]

So there is no way to do these with programming.

For instance, setting a variable in the DB and grabbing it to override the 
field when transferring?

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Mudgett
Sent: Wednesday, November 16, 2011 2:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom Attended Transfer

 When you perform an attended transfer, the extension of the person
 transferring is displayed to the co-worker.
 
 Can I override the caller ID to display the caller’s callerID during a
 blind transfer?

 Upgrade to 10.0 – this isn’t available in any of the 1.X flavors
 because they had to re-invent the background stuff for it.

Asterisk v1.8 added the connected line support not Asterisk v10.

Richard

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Re: [asterisk-users] Polycom Attended Transfer

[eherr]

Unfortunately, I only have 1.4.26 installed.

What's the next stable version?

Should I go to 1.6, 1.8, or 10

Thanks,
--E

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Mudgett
Sent: Wednesday, November 16, 2011 2:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom Attended Transfer

  When you perform an attended transfer, the extension of the person
  transferring is displayed to the co-worker.
 
  Can I override the caller ID to display the caller’s callerID
  during a
  blind transfer?
 
  Upgrade to 10.0 – this isn’t available in any of the 1.X flavors
  because they had to re-invent the background stuff for it.
 
 Asterisk v1.8 added the connected line support not Asterisk v10.
 
 So there is no way to do these with programming.
 
 For instance, setting a variable in the DB and grabbing it to override
 the field when transferring?
 

See
https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information

This applies to v1.8 and later.

Richard

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Re: [asterisk-users] Polycom Attended Transfer

[eherr]

I understand and agree.

There is one client who prefers having the attended transfer still display the 
original caller ID because some users still just hit
transfer and hangup. The boss has a few times got caught saying What when he 
thought it was an internal call but really wasn't.



-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan Wagoner
Sent: Wednesday, November 16, 2011 3:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom Attended Transfer

On Wed, Nov 16, 2011 at 1:49 PM, eherr email.eherr9...@gmail.com wrote:
 When you perform an attended transfer, the extension of the person
 transferring is displayed to the co-worker.



 Can I override the caller ID to display the caller's callerID during a blind
 transfer?



 Thanks,

 --E


The point of an attended transfer is to announce the calling party.
When you hit transfer on the Polycom you have the option to select
Blind on the screen. A blind transfer will use the caller id of the
incoming call, not the person making the transfer.

Ryan

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[asterisk-users] Becoming a CLEC

[eherr]

I know this questions is not really asterisk related, however I know a lot of 
people here are in the industry.

 

I was curious if anyone here could provide insight on how to become a 
facilities based CLEC.

 

I did a lot of google-ing and read info on voip-info.org but it's all the same 
generic stuff and they tell you how to contact an
agency.

 

If anyone has some insider knowledge or advice, please shoot me an email.

 

Currently, we are in the Westchester County and will be renting rack space in 
60 Hudson down in NYC.

 

We are already are an ISP/ITSP but we are a reseller and a paper CLEC.

 

Thanks,

--E

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Re: [asterisk-users] How do extensions stay registered

[eherr]

I think the question is more along the lines of how does asterisk know 
immediately when a sip phone becomes on line and when you
unplug the phone from the network, how does asterisk essentially know 
immediately that it status is UNKNOWN

 

If I am not mistaken.

 

--E

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Monday, November 14, 2011 5:01 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] How do extensions stay registered

 

Extensions do not register - peers do.  A peer can register itself or be 
registered by Asterisk.  In most cases the extension is
equivalent to the peer (301 = 301) but it can be quite different (301 = 
sipuser1) or (301 = d...@impalanetworks.com).

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Douglas Mortensen
Sent: Monday, November 14, 2011 3:52 PM
To: 'asterisk-users@lists.digium.com'
Subject: [asterisk-users] How do extensions stay registered

 

I know this is probably a very basic question for many on this list. But in 
troubleshooting an issue, I wanted to take a step back 
ask the question. In Asterisk (or maybe all SIP), how do extensions stay 
registered with the SIP server?

 

Do the extensions simply register repeatedly as a means of telling asterisk 
I'm still here, or are there actual keepalive packets
that are transmitted to actually keep a TCP session alive? My guess is the 
former.

 

But am I oversimplifying it? Is there more to the process?

 

Thanks,

-

Doug Mortensen

Network Consultant

Impala Networks Inc

CCNA, MCSA, Security+, A+

Linux+, Network+, Server+

.

www.impalanetworks.com

P: (505) 327-7300

F: (505) 327-7545

.

 

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Re: [asterisk-users] How do extensions stay registered

[eherr]

I think the wrap up answer is the interval of registration compacted, if used, 
with the SIP OPTION packet.

 

I like the SIP OPTION packet because we have scripts to monitor the status and 
lets us know when a phone is up or down.

 

--E

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez
Sent: Monday, November 14, 2011 5:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How do extensions stay registered

 

I think the registration part was answered.  The de-registration part is 
different.  If the phone is gracefully taken off line it
specifically de-registers.  If it just can't be reached because it powers off 
or the router closes NAT, or whatever, then Asterisk
won't know this until it times out.

 

On Mon, Nov 14, 2011 at 3:19 PM, eherr email.eherr9...@gmail.com wrote:

I think the question is more along the lines of how does asterisk know 
immediately when a sip phone becomes on line and when you
unplug the phone from the network, how does asterisk essentially know 
immediately that it status is UNKNOWN

 

If I am not mistaken.

 

--E

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Monday, November 14, 2011 5:01 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] How do extensions stay registered

 

Extensions do not register - peers do.  A peer can register itself or be 
registered by Asterisk.  In most cases the extension is
equivalent to the peer (301 = 301) but it can be quite different (301 = 
sipuser1) or (301 = d...@impalanetworks.com).

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Douglas Mortensen
Sent: Monday, November 14, 2011 3:52 PM
To: 'asterisk-users@lists.digium.com'
Subject: [asterisk-users] How do extensions stay registered

 

I know this is probably a very basic question for many on this list. But in 
troubleshooting an issue, I wanted to take a step back 
ask the question. In Asterisk (or maybe all SIP), how do extensions stay 
registered with the SIP server?

 

Do the extensions simply register repeatedly as a means of telling asterisk 
I'm still here, or are there actual keepalive packets
that are transmitted to actually keep a TCP session alive? My guess is the 
former.

 

But am I oversimplifying it? Is there more to the process?

 

Thanks,

-

Doug Mortensen

Network Consultant

Impala Networks Inc

CCNA, MCSA, Security+, A+

Linux+, Network+, Server+

.

www.impalanetworks.com

P: (505) 327-7300 tel:%28505%29%20327-7300 

F: (505) 327-7545 tel:%28505%29%20327-7545 

.

 


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-- 

Carlos Alvarez

TelEvolve

602-889-3003

 

 

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[asterisk-users] DID Provider Issues

[eherr]

How do people get around when their DID provider is experiencing routing issues 
and cannot terminate the DID to your box and all
your clients from that provider can no longer receive inbound calls to their 
company?

Thanks,
--E


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Re: [asterisk-users] DID Provider Issues

[eherr]

Understandable.

I know you can reroute out a different trunk or forward to another number in 
the event the end user loses connectivity to the world.

I was wondering what if happens when your upstream DID provider can no longer 
route the calls to you?

Do they have redundancy? Sometimes it can take a provider 5 - 20 minutes to 
settle into their backup routing before a call gets
completed again.

Also, I know you can forward calls directly from the upstream providers website 
but when they are having issues, logging in to
accomplish this is like asking someone to grow wings and fly.

You could also forward from your asterisk box again, that is assuming the calls 
are making it to you from the upstream.

I even know of an instance where a main TDM switch failed in NYC and calls 
couldn't be re-routed for awhile.

And thank luck for that internet outage this week only lasted a minute or so, 
or everyone would have been in bad shape.

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez

As we were discussing off-list, I don't think there's a solution for problems 
on the PSTN side.  Our solution has been to fire
carriers that have these problems frequently.  We haven't had an origination 
carrier issue in a very long time.  If your
origination carrier has good redundancy, and you do too, then SIP/IP issues 
shouldn't get in the way.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Friday, November 11, 2011 12:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DID Provider Issues

I work for a CLEC which as VoIP services.

We allow customers to specify a telephone number to send calls to if, for some 
reason, the call cannot be sent to the customer.
Usually this is for when the customer's circuit is down.  

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr
Sent: Friday, November 11, 2011 12:43 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] DID Provider Issues

How do people get around when their DID provider is experiencing routing issues 
and cannot terminate the DID to your box and all
your clients from that provider can no longer receive inbound calls to their 
company?

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Re: [asterisk-users] Frequent Asterisk Restarts

[eherr]

I am having similar issues with Asterisk 1.4.26

It happens at random times; could be once a day or a few hours in between up to 
a month or so.

Haven't found a solution to my problem yet either.

--E 

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl
Sent: Thursday, November 10, 2011 1:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Frequent Asterisk Restarts

Hi all,

I've recently started experiencing frequent * restarts, usually caused by a 
segfault.  So, I moved my users to a different server and the same thing 
happened.  It's a fairly busy box, so I considered memory exhaustion.  Nope, 
not even swapping.

Now I'm thinking file handles, but I won't be able to test until tonight.  Is 
there anything else I should look at?

I'm (re)starting * via inittab.  I'm assuming I can just use a script that 
contains the proper ulimit incantation to increase the file handles...?

Any other advise on how to improve stability?  BTW, both servers are running 
Asterisk 1.6.2.9 w/ realtime sip/voicemail.



-- 

Take care and have fun,
Mike Diehl.

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Re: [asterisk-users] Frequent Asterisk Restarts

[eherr]

Unfortunately, I didn't compile with DON'T_OPTIMIZE. Would this render my 
backtrace.txt completely useless or should I still submit?

Thanks,
--E

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Madsen
Sent: Thursday, November 10, 2011 1:47 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Frequent Asterisk Restarts

On 11-11-10 01:15 PM, Eric Wieling wrote:
 The Asterisk source tree has a document with instructions on getting a 
 backtrace from the segfaults so you can report it on the
issue tracker.

Most up to date documentation is on the Asterisk wiki:

https://wiki.asterisk.org/wiki/display/AST/Debugging

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