Hello Sean,
Sunday, September 8, 2013, 11:25:24 PM, you wrote:
The problem is that once a phone has used the server, no other phone can
use it. The servers sees all the phones as having the same ip address
(though different ports).
This sounds like the Peer v Friend problem I have had
Hello Doug,
Monday, March 3, 2014, 7:13:52 PM, you wrote:
I was successful in compiling asterisk in raspbien except for the
following error If I enable the gsm codec. It appears there is
something in the Makefile n this directory that needs to be changed.
Probably involving optimization. Not
Hello motty,
Thursday, September 18, 2014, 6:35:40 PM, you wrote:
Hello, I would to allow users to place calls overseas such as India
and Malaysia but only with a security code. if they don't have a
security code I want to be able to drop the calls.
I use this
exten =
Hello Kantharuban,
Friday, September 4, 2015, 8:19:28 AM, you wrote:
> Thanks for your info, What is the impact of the following line in
> dialpla Dial(SIP/19201/19202,300)
It does not look like a valid format. If you are trying to dial two
SIP devices (19201 and 19202) with a timeout
Hello Daniel,
Wednesday, November 25, 2015, 7:02:08 PM, you wrote:
> It still asks for a client certificate.
> See this screen shot, hopefully it showswhat I mean.
> http://firestar-hosting.com/clientcert.png
It is probably a Keychain issue, and not a problem with the Issue
Tracker website.
Title: Re: [asterisk-users] Including doesn't have any effect
On Monday, June 6, 2016, 4:55:12 PM, AJS wrote:
> Are you lazy enough to edit a text file and reload your dialplan, *every single
> time* someone calls you, that you don't want to have to speak to ever again?
> Not sure about you,
Title: Re: [asterisk-users] Including doesn't have any effect
Hello Frank,
Monday, June 6, 2016, 9:46:47 PM, you wrote:
> As far as I know, Asterisk's database/blacklist function only supports
> exact match of caller ID.
> If you want to block a specific area code or a block of numbers (eg.
>
Hello Phil,
On Saturday, April 23, 2016, 12:19:15 PM, you wrote:
> I have checked that the username and password in my config agree both
> ends, and have even tried changing them.
> The bulk of my calls come in on A, so I am obviously trying to find
> out what has gone wrong. No-one else is
Hello Phil,
On Saturday, April 23, 2016, 11:11:29 PM, you wrote:
> Actually, this is now sorted. It turns out the latest recommended
> configs on the A wiki had peer vs. user confusion. On correcting
> this, all was well.
I'm glad you found it. It look me a while to track down that problem
when
Hello Guillermo,
On Tuesday, September 13, 2016, 7:10:29 PM, you wrote:
> I have tried to install Wanpipe 7.0.20 in a fresh-installed Debian
> 8.5 (Jessie) server, but when I reach to the point where I have to
> enter the Linux Headers directory (installed in default with
> “apt-get install
On Thursday, December 14, 2017, 10:05:23 PM, Tony Mountifield
(t...@softins.co.uk) wrote:
> So I think you really do need to have a single peer section for all sipgate
> calls, pointing to one sipgate context in your dialplan that contains all
> your various extensions like se2489, sj0151, etc.
Hello,
Following the recent discussions about CentOS, I am interested to hear whether
anyone is successfully running Sangoma's Wanpipe (for an analog telephony card)
using Centos 7? Upgrades of Wanpipe have been tricky at best in the past, and I
don't want to make the leap to CentOS 7 without
Hello Thelma,
Friday, February 16, 2018, 2:16:02 AM, you wrote:
> Contact: "sip:pstn-"
> And it found in sip.conf only:
> Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060
> Is perhaps the name effected by the special character "-" (dash) that is
> why it only matches "pstn" and
Hello Doug,
Friday, December 13, 2019, 11:03:37 AM, you wrote:
>> This is exactly what I do - “press 1 for a human”
>> Works great
> I do this as well, but I also do a database lookup to see if the number
> is on our speeddial list and if so, pass the call directly on without
> the IVR
Hello Jeff,
Friday, December 13, 2019, 7:42:38 PM, you wrote:
> Mind posting your dialplan code? I was thinking the same thing - very
> much like an old spam control program I used to use whose name now
> escapes me. First time senders would have to respond to an auto-reply,
> then were added
Hello Doug,
Tuesday, October 15, 2019, 5:07:45 PM, you wrote:
> Personally, I don't think MACROS are going anywhere any time soon,
> so I have not bothered looking into a substitution.
Haven't they gone in Asterisk 16? I've just upgraded from Asterisk 13
and had to do some hasty re-writing of
Hello Joshua,
Wednesday, October 16, 2019, 10:39:27 AM, you wrote:
> The module is still present, it just isn't built by default. It
> requires explicit enabling using menuselect.
I was obviously too concerned about the rebuilding of wanpipe with a
new kernel version to notice anything about
Title: Re: [asterisk-users] which linux for asterisk?
On Wednesday, December 9, 2020, 11:07:55 AM, Antony Stone wrote:
> Upgrading a Debian server to the next release is a whole lot easier than doing
> a CentOS one.
It will be slightly easier for me having all Debian-based boxes, but my main
Hello Thelma,
Thursday, December 24, 2020, 9:26:53 AM, the...@sys-concept.com wrote:
> In astersik-11 MWI light was cleared as soon as I checked the message.
> In asterink-13 it takes about 20min to set the light ON and the light
> takes over an hour to clear.
I had this problem following an
Hello Antony,
Monday, August 9, 2021, 4:14:11 PM, you wrote:
> You want to look for firewall rules which will allow UDP in both directions
> on
> ports 1 - 3 (typically, may vary a bit, but something like that), or
> alternatively, look for any rules which would block this, and remove
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