asterisk network connections

Report
Question

4520 is for DUNDI. Obviously your install uses H323 in some flavor.
Mgcp-callagent is for jitterbuffering? And sieve and complex-main I have no
clue (perhaps H323 tag-alongs) From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Friday, February 17, 2012 1:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] asterisk network connections Hello, with the command lsof -i I notice the following network connections of the
asterisk proces : asterisk 23006 root 12u IPv4 1088961 UDP
*:mgcp-callagent
asterisk 23006 root 13u IPv4 1088964 TCP *:sieve
(LISTEN)

Asterisk Users 3.6 years ago 1 Answer

Headset Options

Report
Question

Hi, Jabra headsets work fine with Polycom. Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] På vegne af Blake Burgess
Sendt: 7. februar 2012 05:01
Til: asterisk-users@lists.digium.com
Emne: [asterisk-users] Headset Options Hey, I've heard recently from quite a few customers that there's cordless handsets around which don't require a lifter. Is anyone aware of any of these which will work with the cisco 69xx's, 79xx's or any of the current polycom range? -Blake

Asterisk Users 3.6 years ago 0 Answers

dial a queue

Report
Question

No :( 2012/2/6, Danny Nicholas :
> Queue(8888)?
>
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com
> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Alejandro
> Cabrera Obed
> Sent: Monday, February 06, 2012 1:40 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Custom extension: dial a queue
>
> No, Local/queue/8888 don't work at all :(
>
> 2012/2/6, Danny Nicholas
:
>> Local/queue/8888?
>>
>> -----Original Message-----
>> From: asterisk-users-bounces@lists.digium.com
>> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf…

Asterisk Users 3.6 years ago 1 Answer

Blind transfers being cancelled by asterisk & hanging up on remote caller

Report
Question

Are we the only 2 people on this list experiencing this issue? (surprised) Anyone else have any insights? My hunch is that this is likely some type of FreePBX issue with how it generates the [from-internal-xfer] context. From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Douglas Mortensen
Sent: Saturday, January 07, 2012 3:16 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Blind transfers being cancelled by asterisk & hanging up on remote caller Oh crap. I just reread the previous post & realized I'm not alone. Hallelujah! I'll post back more info soon. -
Doug Mortensen
Sent…

Asterisk Users 3.7 years ago 0 Answers

asterisk -> AGI (perl) -> sqlplus(oracle)

Report
Question

Yes, I already declared 'use lib
"/home/asterisk/lib/lib64/perl5/5.8.8/x86_64-linux-thread-multi/";' in my
AGI. When I execute the script as a user Asterisk, i.e. perl -wc test.pl in
return I'm getting OK and no error messages and script is running fine when
I try to run in shell. Even though I already declared the environmental variables in
.bash_profile. At the end I tired every method but still stuck in this
problem. Date: Thu, 5 Jan 2012 14:07:59 -0800
> From: "Ron Bergin"
> Subject: Re: [asterisk-users] asterisk -> AGI (perl) -> sqlplus
>…

Asterisk Users 3.7 years ago 0 Answers

registration not authorized - stale nonce

Report
Question


Asterisk destroys SIP dialogs in 32 secs, so increased the UAC registry expiration to 300 secs just in case, but that didn't help either.
From: mistral9999@hotmail.com
To: asterisk-users@lists.digium.com
Date: Sun, 1 Jan 2012 18:13:07 -0500
Subject: [asterisk-users] registration not authorized - stale nonce I have a very basic setup where a UAC registers with Asterisk 1.8.7.2 - both on the same subnet, no nat. The following is the flow of messages:
1. UAC sends the registration request
2. Asterisk responds with 401 Unauthorized with a new nonce
3.…

Asterisk Users 3.7 years ago 0 Answers

Asterisk Registrar / Trunk

Report
Question

Dears, 1-I have a GSM gateway (GOIP) with 8 ports, I used to let every port
register to VoIPSwitch in order to know how many minutes does this GSM
card, ASR ,ACD on each card. It's too simple on VoIPSwitch to add the registrar client to dial plan ,but
in asterisk only I can find trunks How can I do that with asterisk . 2-Do any one know from where I can download a2billing prompts in Arabic for
free. Regards Khaled Chehab NGN Eng. Description: xplorium Operations Office - Lebanon Office : +961 1 868686 ext…

Asterisk Users 3.7 years ago 0 Answers

performance/memory

Report
Question


I have a couple of performance/memory related questions:
Is there any downside to using long URIs as far as memory or database (mysql) performance is concerned, e.g. sip:1234567890_1234567890@abc.com? Or is this negligible?
Also is there a performance hit if no pattern matching is used? e.g. exten => _XXX,Noop(... vs exten => 100,Noop(..
exten => 101,Noop(...
exten => 102,Noop(...
...
exten => 999,Noop(... If a call comes to 999, does Asterisk go through each extension sequentially from 100 to 999 until it finds the matching one? Thanks,
Matt

Asterisk Users 3.7 years ago 1 Answer

Interesting attack tonight & fail2ban them

Report
Question

Maybe your logger is not setup properly?! You should get the IP in logs. I
can't think of when you won't get the IP in your logs unless the SIP
packets are manipulated. That IP is from Voxel.net. You don't have a VPS or
service from them do you? 2011/12/29 Michelle Dupuis > 1. I checked the log and I don't see any registration attempt, so I
> *assume* they simply send an invite, and so they are in the
> external/outside context of my dialplan. So they are trying to reach

Asterisk Users 3.7 years ago 0 Answers

Interesting attack tonight & fail2ban them

Report
Question

1. I checked the log and I don't see any registration attempt, so I *assume* they simply send an invite, and so they are in the external/outside context of my dialplan. So they are trying to reach extensions which don't exist. If they succesfully registered they would be on the internal context, and their calls would have succeeded. (Or am I missing something?). I actually see nothing in the log but the notice (and nothing on the CLI but the notice)...so I assume it is only an invite? 2. I got their IP by turning on SIP DEBUG while they…

Asterisk Users 3.7 years ago 0 Answers