Ok Good. It always feel good to add +1 to the list of resolved user-list
issues.
As I'm not a POWERFUL guy, I can't simply ask my ISP to unblock it.
It takes a VPN or in near future WebRTC(in other words "Knowledge") to
become one powerful guy. With these technologies you don't need to care
what
On 12-08-10 06:20 PM, Daniel Pocock wrote:
Debian is very conservative about accepting updates during the `freeze'
process - they will most likely want to see a 1.8.13.2 release with ONLY
the most essential fixes
a) is anyone else aware of these bugs?
b) what essential changes should go into
>> Debian is very conservative about accepting updates during the `freeze'
>> process - they will most likely want to see a 1.8.13.2 release with ONLY
>> the most essential fixes
>>
>> a) is anyone else aware of these bugs?
>>
>> b) what essential changes should go into 1.8.13.2 for Debian?
>>
> W
On 12-08-10 04:47 PM, Daniel Pocock wrote:
Debian 7 is currently in the `freeze' status with 1.8.13 - that means
Debian 7 is very likely to release 1.8.13 and be carrying it for the
next 2-3 years (typical lifetime of a Debian release)
I run 1.8.8. TLS has a bug: it fails to receive BYE over
The Asterisk Development Team is pleased to announce the first beta release of
Asterisk 11.0.0. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases
All interested users of Asterisk are encouraged to participate in the
Asterisk 11 testi
Debian 7 is currently in the `freeze' status with 1.8.13 - that means
Debian 7 is very likely to release 1.8.13 and be carrying it for the
next 2-3 years (typical lifetime of a Debian release)
I run 1.8.8. TLS has a bug: it fails to receive BYE over the TLS
connection from my Polycom phone.
I
I am starting to use ConfBridge and not MeetMe in asterisk 10.
I have everything converted over EXCEPT.
I am using an AGI and AMI to bring phones into a conf automatically.
When I do that the conf is going just fine - however - I head beep,
beep, beep.
I have every sound listed in confbridge.
- Original Message -
> From: "Matt Hamilton"
> To: asterisk-users@lists.digium.com
> Sent: Friday, August 10, 2012 12:15:14 PM
> Subject: Re: [asterisk-users] asterisk and meetme
> > ConfBridge is the preferred conference application in Asterisk 10+.
> > While
> > MeetMe is currently d
On Tue, 31 Jul 2012 09:44:26 +0100
Sebastian Arcus wrote:
> I have two setups with SIP hardware phones as extensions and POTS
> lines as trunks. Internal SIP to SIP calls are crystal clear, but all
> calls bridged to POTS have a significant amount of static noise. The
> problem is that if I plug
> ConfBridge is the preferred conference application in Asterisk 10+. While
> MeetMe is currently deprecated, you can still enable it and run it in
> Asterisk 10+.
What's going to happen to SLA (which is heavily integrated with MeetMe)? Will
the functionality be ported to ConfBridge?
Thanks,
M
Hi everyone,
We are getting cotinueous error messages over the past few days from iCall:
-- Called iCall/01144
-- Got SIP response 500 "Server internal failure" back from
72.249.14.242
Is this something everyone else is getting? They are very bad at support
and I am not sure if it'
On 08/10/2012 11:23 AM, Jerry Geis wrote:
I have a profile in confbridge
[MessageNetConfBridge]
and more...
Asterisk is reading it at startup.
[1;30m == ^[[0mParsing '/etc/asterisk/confbridge.conf': ^[[1;30m ==
^[[0mFound
^[[1;30m ^[[0mapp_confbridge.so => (^[[0;33mConference Bridge
Applicat
On Fri, Aug 10, 2012 at 2:49 AM, Kannan wrote:
> In contrast, hosted IVR will have only one number dedicated to a business,
> and the business can maintain the call flow and sound files. The system
> will integrate with their CRM and offer personalized services to the
> customers of the business.
I have a profile in confbridge
[MessageNetConfBridge]
and more...
Asterisk is reading it at startup.
[1;30m == ^[[0mParsing '/etc/asterisk/confbridge.conf': ^[[1;30m ==
^[[0mFound
^[[1;30m ^[[0mapp_confbridge.so => (^[[0;33mConference Bridge
Application^[[0m)
When I try to use it I get a wa
>
> But still contact your ISP and get them to un-block your port 5060. You
> paid
> them for an Internet connection; and an Internet connection means *all*
> ports,
> not just *some* ports.
>
There has been lots misuses of VOIP here and government policies are
strange. Instead, lots of illegal V
>
> Oh, I see - check if your country blocks the SIP port 5060 ? try changing
> the default poert from 5060 to something else like and then try this.
> I think your ISP is blocking the SIP.
>
> HEY!!
You rock. Yeah it seems this is the main problem. I tried with and saw
the message at in
>
> 1. Can you send UDP packets across your LAN?
> If not, check your client machine.
>
> Ok, I can't register a SIP phone from another PC to my Asterisk server
under same net. Can you tell me one thing? When I call to 127.0.0.1 to my
local Asterisk server it works. When I call to the IP of my eth
- Original Message -
> From: "Jerry Geis"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Friday, August 10, 2012 8:25:54 AM
> Subject: Re: [asterisk-users] asterisk and meetme
>
> On 08/10/2012 09:00 AM, Jerry Geis wrote:
> My bad - "make menuconfig" was no
On 08/10/2012 09:00 AM, Jerry Geis wrote:
I just downloaded and compiled from source asterisk 10.7.0
after installing and running I tried to do a meetme, did not work.
I looked in the apps/app_meetme* and there is only the C file, there
is no .o
seems like it did not compile.
Is that a new def
What you want can be done by OpenVBX, why dont you try exploring that model
?
Regards,
Mitul Limbani,
Chief Architech & Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@
I just downloaded and compiled from source asterisk 10.7.0
after installing and running I tried to do a meetme, did not work.
I looked in the apps/app_meetme* and there is only the C file, there is
no .o
seems like it did not compile.
Is that a new default behavior?
Looking for the trick to ge
On Friday 10 August 2012, Patrick Lists wrote:
> On 10-08-12 10:12, SamyGo wrote:
> > Oh, I see - check if your country blocks the SIP port 5060 ? try
> > changing the default poert from 5060 to something else like and
> > then try this.
> > I think your ISP is blocking the SIP.
>
> If that i
Hi all.
I have this problem with my Digium 2E1 card and PRI, for hours It works
well, with some meesages...
[Aug 10 09:20:31] NOTICE[32270] chan_dahdi.c: PRI got event: HDLC Abort (6)
on Primary D-channel of span 1
But PRI continue uphours later... PRI go down.
I thought the problem was in th
On 10-08-12 10:12, SamyGo wrote:
Oh, I see - check if your country blocks the SIP port 5060 ? try
changing the default poert from 5060 to something else like and
then try this.
I think your ISP is blocking the SIP.
If that is the case, setup an IAX connection and see if that works.
Regard
Hi Carlos,
The idea is this. We are planning to offer customized version of Asterisk
for specialized purposes. When we offer hosted PBX, using multi-tenancy
support, it is just going to be PBX, as opposed to a fully blown IVR. It
will have automated attendant feature, but not IVR.
In contrast,
Oh, I see - check if your country blocks the SIP port 5060 ? try changing
the default poert from 5060 to something else like and then try this.
I think your ISP is blocking the SIP.
On Fri, Aug 10, 2012 at 1:10 PM, A J Stiles
wrote:
> On Thursday 09 August 2012, Sazzad wrote:
> > Hi,
> >
> >
On Thursday 09 August 2012, Sazzad wrote:
> Hi,
>
> I've successfully setup Asterisk on my local PC and can make call using
> Twinkle to the server. But, I cannot call to my Asterisk server at
> Rackspace. .
>
> My question is how can I troubleshoot this scenario? (Is this question
> within t
>
> 1-a:Are your SIP packets from PC/SoftPhone reaching the server !! On
> Asterisk CLI execute "*CLI>sip set debug on"
>
> Yeah I've done that, and no UDP packets are reaching my Asterisk server
and neither I can catch any UDP packets at my server using nc -u.
- are you even able to ping
> your
28 matches
Mail list logo
