Hi again,
I've got a user who's using a bunch of Grandstream GXP2xxx's. For the most
part, they work, except for when they try to do a phone-based call transfer.
Here's what it looks like is happening:
Phone A is on a call with phone B. (B could be another phone, or a PSTN
endpoint.)
The use
Sent from my Verizon Wireless 4G LTE DROID
Mike Diehl wrote:
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El 27/03/14 11:59, Alex Villacís Lasso escribió:
I am having an issue that prevents WebSockets over SSL/TLS (or any kind of
encrypted HTTP traffic to Asterisk) from working after an openssl library
update.
My setup is CentOS 6 x86_64, and initially, with openssl[-devel]-1.0.0-20.el6_2.5.x86_6
Well, I went to an online xml validation site and found an error. After
correcting the error, my problem is gone!
Thank you.
Mike.
On Thu, Mar 27, 2014 at 2:56 PM, Noah Engelberth
wrote:
> To me, the settings you've sent look correct. However, one thing I've
> found with SPA configuration f
On 03/27/2014 08:36 PM, Eric Wieling wrote:
> I have an iptables file which blocks all traffic except traffic from networks
> allocated by ARIN or are Legacy networks. I pulled the information from
> http://www.iana.org/assignments/ipv4-address-space/ipv4-address-space.xhtml
>
> My iptables
To me, the settings you've sent look correct. However, one thing I've found
with SPA configuration files is that they're very picky - if they don't parse
as valid XML anywhere in the file, it will pretty much silently discard the
entire file. The first troubleshooting step I use for SPA provis
Hi all,
I've got a provisioning file that I use to configure Cisco SPA112's.
I'm wanting to get this file to do 3 things for me. I want to turn T.38
on, Call forwarding off, and Call waiting, off for both lines. but it's
not working.
This is what I'm using to enable T.38 for line 1.
Yes
1
Yes
- Original Message -
> From: "Michelle Dupuis"
> To: "Asterisk Users List"
> Sent: Thursday, March 27, 2014 12:55:21 AM
> Subject: [asterisk-users] Security log format / content
> I've noticed that the Asterisk (v11) security log captures attempts
> do dial without first authenticating
I have an iptables file which blocks all traffic except traffic from networks
allocated by ARIN or are Legacy networks. I pulled the information from
http://www.iana.org/assignments/ipv4-address-space/ipv4-address-space.xhtml
My iptables script can be found at the link below.
http://help.n
On 03/26/2014 05:05 PM, Michelle Dupuis wrote:
> I see a lot of attempts by hackers to call 00972595301123
> or 011972595115207 or variations but that same 972595 is often present.
>
>
> Can someone break down that dial string with an explanation? The 011
> look like an overseas call (from Ame
I am having an issue that prevents WebSockets over SSL/TLS (or any kind of
encrypted HTTP traffic to Asterisk) from working after an openssl library
update.
My setup is CentOS 6 x86_64, and initially, with openssl[-devel]-1.0.0-20.el6_2.5.x86_64 . With this openssl versions, https over TCP port
Wouldn't it make more sense to handle this by just monitoring the calls and doing everything
else with normal data processing?
jg
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On 26 Mar 2014, at 19:14, Mickael MONSIEUR wrote:
> Hello,
>
> When I get a SIP INVITE as follows:
>> INVITE sip:s@10.1.0.191:5060 SIP/2.0
>> Max-Forwards: 69
>> From: "0475XX" ;tag=as7df9ab18
>> To:
>> Contact:
>> Call-ID: 344d42bd16975a54141d11f635bdf...@sip.domain.com
>> CSeq: 102 INV
On 26.03.2014 11:30, jg wrote:
What do you mean with "voice recorders"? Voice mail, if nobody answers,
or do want to monitor calls?
I mean voice recording (just like a voice box does), but the recorder is
not Asterisk, but a dedicated VoIP recorder.
regards
Klaus
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In my experience iaxmodem + Hylafax is very stable and work in my setups
fine.
But in these setups I either use ISDN uplinks or SIP trunks with low RTT
and jitter (highspeed links to the service provider, no WAN links).
Thus, T38 is in my setups not necessary.
regards
Klaus
On 24.03.2014 06
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