r providers to see what they see or reset
the trunk when the issue comes up to see if it matters
Good luck
> On 08/03/22 11:54, Duncan Turnbull wrote:
> > It’s been a r we hike since we used these cards. This example may help
> >
> >
> https://wiki.freepbx.org/plugins/
It’s been a r we hike since we used these cards. This example may help
https://wiki.freepbx.org/plugins/servlet/mobile?contentId=73007457#content/view/73007457
My thinking is it sounds like a timing error. Make sure your provider is the
timing source. Once it loses time you will get dropped cal
> On 9/01/2022, at 7:11 PM, John Covici wrote:
>
> On Sat, 08 Jan 2022 19:17:57 -0500,
> Antony Stone wrote:
>>
>>> On Sunday 09 January 2022 at 00:50:27, John Covici wrote:
>>>
>>> Hi. I am using asterisk 18.3 and freepbx.
>>
>> Hm, which version of FreePBX uses Asterisk 18.3?
>>
>>> H
config
>
> Marek
>
>
> 2021-09-10 1:19 GMT+02:00, Duncan Turnbull :
>>
>>
>>>> On 10/09/2021, at 4:37 AM, Marek Greško wrote:
>>>
>>> There are other systems running on the same hardware. It would just
>>> leave open ports
> On 10/09/2021, at 4:37 AM, Marek Greško wrote:
>
> There are other systems running on the same hardware. It would just
> leave open ports here.
>
> Do not compare SIP ALG on a closed source device to an opensource
> software with active development. I had no such problems in the past
> when
}
>>
>> chain OUTPUT {
>>type route hook output priority mangle; policy accept;
>>...
>> udp dport 5060 ip dscp set 0x04
>>...
>> }
>> }
>>
>> table ip6 filter {
>> ct helper sip {
>>type "sip" protocol udp
>
s
> anybody have wide experience with nftables and sip?
If you publish your rule set then we could look. Did you write the rules? What
have you checked so far?
>
> Thanks
>
> Marek
>
>
> 2021-09-07 10:40 GMT+02:00, Antony Stone
> :
>> On Monday 06 September 2021
ots of debug advice on google.
>>>>>
>>>>> Asterisk cli did not show anything interesting. I tried pjsip set
>>>>> logger verbose on, but no logs showed anywhere. What am I doing wrong?
>>>>>
>>>>> Ma
is looking normal
>>>> except
>>>> asterisk doesn’t appear to beseeing the rtp packet
>>>>>
>>>>> Thanks
>>>>>
>>>>> Marek
>>>>>
>>>>>
>>>>>>
>>>>>> Hav
> On 6/09/2021, at 7:10 PM, Marek Greško wrote:
>
> Hello,
>
>
>
> 2021-09-06 2:51 GMT+02:00, Duncan Turnbull :
>> Hi Marek
>>
>> I didn't understand your setup originally.
>>
>> Can you confirm this is correct:
>>
>&g
gt; communication between asterisk and remote phones behind some internet
> provider. This is the only conversation to look at.
> The phone private address is 192.168.100.235.
>
> Thanks
>
> Marek
>
>
> 2021-09-05 1:11 GMT+02:00, Duncan Turnbull :
> >
> >
> On 5/09/2021, at 10:21 AM, Marek Greško wrote:
>
> Hello,
>
> could you please answer my previous question about anonymizing several
> parameters? I have the data ready, but will post after answer. I have
> no clue whether I could disclose some important data not deleting
> them.
>
> Regar
t; Hello,
>
> I agree my knowledge of SIP itself is poor, but I have quite well
> general tcp/ip understanding. What sip parameters should be
> anonymized? How about tag, branch, call-id, cseq values?
>
> Thanks
>
> Marek
>
>
> 2021-09-04 12:36 GMT+02:00, Duncan
>
> Thanks
>
> Marek
>
> 2021-09-04 0:40 GMT+02:00, Antony Stone
> :
>> On Saturday 04 September 2021 at 00:34:49, Duncan Turnbull wrote:
>>
>>>>> On 4/09/2021, at 7:53 AM, Marek Greško wrote:
>>>>>
>>>>>
er's router in the previous discussion. And it made a big
> improvement in the experience.
>
> Marek
>
> 2021-09-03 12:19 GMT+02:00, Duncan Turnbull :
>>> On Fri, Sep 3, 2021 at 8:47 PM Marek Greško wrote:
>>>
>>> Hello,
>>>
>>> I lo
another provider which had working SIP
> ALG I had no problem even without nat configuration on the asterisk
> side.
>
> The experience is clearly better after disabling SIP ALG, but we still
> face problems after asterisk side reboots.
>
> Could you point me for what should I lo
Hello,
it triggered again. Even disabling RTSp ALG did not help. My fantasy
ends here. It agains seems to be reboot triggered on asterisk side.
Not every one. But there was surely one before it was last working.
Reboot of the router on the phone side fixes the problem. Any other
suggestions?
T
> On 25/12/2020, at 3:08 PM, Turritopsis Dohrnii Teo En Ming
> wrote:
>
> Hi Duncan Turnbull,
>
> My final issue has been resolved.
Very well done
Merry Xmas
Cheers Duncan
>
> Please refer to the following post.
>
> Post: Addendum to Teo En Ming's Gu
> On 25/12/2020, at 12:40 AM, Turritopsis Dohrnii Teo En Ming
> wrote:
>
> Hi Duncan Turnbull,
>
> It is a newly created PJSIP extension with default settings. I have never
> configured Do Not Disturb settings before.
>
> Could it be something else?
>
>
> On 24/12/2020, at 6:39 PM, Turritopsis Dohrnii Teo En Ming
> wrote:
>
> Hi Duncan Turnbull,
>
> I have finally managed to get my Cisco 7960 IP phone to register on my
> Asterisk PBX appliance on Christmas Eve 2020.
>
> You can read my guide here:
>
&
Xmas
Cheers Duncan
> On 24/12/2020, at 1:11 PM, Turritopsis Dohrnii Teo En Ming
> wrote:
>
> Thank you for your replies, Duncan Turnbull.
>
> I am going to run tcpdump on my Asterisk PBX server.
>
> By the way, I found a Youtube video.
>
> Youtube video: Cisc
Sent from my iPad
> On 23/12/2020, at 5:33 PM, Turritopsis Dohrnii Teo En Ming
> wrote:
>
> Hi Duncan Turnbull,
>
> You can watch my Youtube video of my Cisco 7960 IP phone.
>
> The link is: https://www.youtube.com/watch?v=ip_F08jmmio
>
> My Youtube video s
Hi there
> On 23/12/2020, at 12:45 PM, Turritopsis Dohrnii Teo En Ming
> wrote:
>
> Good morning Duncan Turnbull,
>
> I have posted my Asterisk PBX server debugging output previously in my
> original post. The link is:
>
> http://lists.digium.com/pipermail/a
> Thank you very much for your kind assistance.
>
>
>
>
> On 2020-12-21 09:58, Duncan Turnbull wrote:
> > Hi there
> >
> > I would normally highlight the part but the email is so long I thought
> > I would just note what I can see
> >
> > It appea
Hi there
I would normally highlight the part but the email is so long I thought I
would just note what I can see
It appears the Cisco is downloading files.
None of the SIP traces show the IP of the phone of the extension
Your proxy is at 192.168.1.9
Your phone is at 192.168.1.130
These are t
Hi Carlos
On Tue, 8 Sep 2020, 12:36 pm Carlos Chavez, wrote:
> Some users have complained that their calls drop after about 30
> seconds.
The rtp timeout is usually about 30 seconds. If rtp is only 1 way then the
calls will drop after 30 secs. This is usually nat/firewall related so a
pac
Sent from my iPad
> On 15/01/2019, at 10:34 AM, Thomas Peters wrote:
>
> Duncan:
>
> You may have it right—I took one phone and set the ring time to 60 seconds. I
> now get about 4 rings on that one.
>
> I wonder how I can change the timing source.
In one version (and I can’t recall whic
Sent from my iPhone
> On 19/04/2017, at 11:43 AM, Ernie Dunbar wrote:
>
>> On 2017-04-18 03:38 PM, Duncan Turnbull wrote:
>> -- Original Message --
>> From: "Ernie Dunbar"
>> To: "'Asterisk Users Mailing List - Non-Commercia
-- Original Message --
From: "Ernie Dunbar"
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
Sent: 19-Apr-17 10:25:59 AM
Subject: [asterisk-users] SIP connections over OpenVPN connection get
one-way voice.
Hi everyone. I'm having some trouble with an OpenVPN tunnel th
> On 7/04/2016, at 6:01 AM, Carlos Chavez wrote:
>
>> On 4/5/16 3:17 PM, Joshua Colp wrote:
>> Carlos Chavez wrote:
>>> I am currently having a voice quality problem with one of our Asterisk
>>> servers. We have checked the network and we have found no problems that
>>> could cause the voice to
> On 4/03/2016, at 5:31 AM, Olivier wrote:
>
> Hello,
>
> I'm remotely managing an asterisk setup using an OpenVPN client on this
> Asterisk box, connecting to an OpenVPN server of mine).
>
> This box is mainly connected to PSTN.
> It is also connected to the Internet, only for remote managem
HI Kevin
Is your VPN set as a localnet? The externip only tends to cope with the
firewall address. If you put the VPNs in the localnet lists then it
won't use NAT to find them.
In answer to your question, the SIP session description in the call
setup has the IP for media for both parties, wh
-- Original Message --
From: "Tony Kasule"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: 30/06/2015 8:34:47 p.m.
Subject: Re: [asterisk-users] Help With Physical Layer
Hello,
Anyone to help me with this issue? It has never worked :(
Hi Tony
I'm not familia
Hi there
This has happened to me before when I changed the tone duration, it was
too long and the PSTN receiver no longer understood the tones, but it
seems unlikely nothing has changed.
The cli or logs should show you whats happening, something will be
blocking the call, either the group or
DNS failure could do this
Asterisk used to get stuck in a symmetric DNS request wait state which meant
everything ground to a halt as it waited for a reply while DNS timed out.
The recommended option was either ip only or a DNS proxy that failed fast this
letting asterisk continue
Cheers Dunc
If you use freepbx you can do it with endpoint manager
http://schmoozecom.com/endpoint-manager.php
It costs I think in the latest freepbx version but there will be earlier
versions around
It's just generating templates by mac for the tftp server
> On 10/04/2015, at 4:37 am, Tafadzwa Nyabasa
On 10 Feb 2015, at 12:22, Jose Flores Galicia wrote:
2015-02-09 14:36 GMT-06:00 jg :
Hi!
Sometimes IAX peers are not reachable and with "iax2 set debug on" I
get
something like this
Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
PONG
Timestamp: 00014ms SCall: 1
On 29 Jan 2015, at 11:07, Administrator TOOTAI wrote:
Le 28/01/2015 22:03, Steven McCann a écrit :
Hello,
Hi
I'm investigating a situation where there was a hundreds of minutes
of
calls from an internal SIP extension to an 855 number in Cambodia,
resulting in a crazy ($25,000+) bill from
On 9/04/2014, at 10:42 pm, Positively Optimistic
wrote:
> We are using vpn routers to connect home users back to our office network.
> Basically, shipping a mikrotik router that 'calls home' and establishes a vpn
> connection for the pc and phone that are connected to the mikrotik... user
Another option we like, but i depends on your preferences is to run them over
openvpn. Works for Mac, Linux and Windows clients.
Since all out clients are under our control we use openvpn a lot and yealink
and other phones have it built in so they can connect directly once initially
setup
Che
Cool
That looks like it is arriving at Asterisk - are you sure asterisk is not
getting it? If you turn on sip debug in asterisk can you see the SIP packets?
It maybe asterisk is ignoring them or replying to them but its going out an
interface you hadn’t thought of, I have had that a few times.
On 21/01/2014, at 6:40 pm, David Cunningham wrote:
> Hi Paul,
>
> Using ngrep/tcpdump shows the packet clearly going from the Kamailio server
> and arriving at the Asterisk server. This is why it's a mystery that Asterisk
> doesn't see the call coming in. We tried removing the firewall (so ip
On 21/01/2014, at 10:24 am, David Cunningham wrote:
> Hi Paul,
>
> The ngrep on the Asterisk server does show it being received. Have you any
> idea what would prevent it getting from the network stack to Asterisk on that
> machine?
>
>
Have you got a static route on asterisk or your defaul
I think that's a good idea, I turned an AA50 into just a trunk device for a sip
box and it worked for a long time
The other things that are small and work are the atcom ip Pbx series
http://www.atcom.cn/products_ippbx.html
They are pretty cheap in NZ but not as low as the beaglebone suggestion
We use Atoms with SSDs for customers and they work well
We have a some with PCI on the motherboards and haven’t had any issue other
than a single issue where a reinstall of the OS cleared up poor ethernet
performance. Use almost no power, and with SSDs can almost avoid fans and thus
moving part
Any chance DNS is dying about the same time the problem occurs
I get this occasionally every 6-12 months and usually because DNS got messed up
and then something didn’t fall back into place when it recovered - networking
looks okay on the machine but asterisk is stuck.
I have been meaning to fo
On 29/10/2013, at 9:55 am, Mike wrote:
> On Mon, 28 Oct 2013, Eddie Mikell wrote:
>
>> All,
>> The users in our organization are well, quite frankly, sick of phone service
>> that is being provided. The choppy phone
>> calls, and drop outs are detrimental to our sales force.
>> I've tried abo
On 29/08/2013, at 10:02 PM, Thorsten Göllner wrote:
> Permissions: take a look at "/etc/udev/rules.d/dahdi.rules". Last line. OWNER
> and GROUP should be the same as the user running the asterisk process (root
> or asterisk?).
>
> Am 29.08.2013 11:47, schrieb bilal ghayyad:
>> Hello;
>>
>> I
On 30/07/2013, at 4:22 PM, Akib Sayyed wrote:
> I didnt understand what you were saying.can you please explain
>
> I am using digium cards
>
> sent from android
>
E1 PSTN line interfaces are either unbalanced 75 ohm( and used to use BNC
connectors ) or a 120 ohm balanced twisted pair.
The
Hi Mitch
On 28/05/2013, at 5:14 AM, Mitch Claborn wrote:
> Asterisk 11.1
>
> We have a situation where one of our incomings POTS lines will not answer.
> There are 2 lines configured by the Telco as a rollover group (rings the line
> that is not busy) and they feed into a Digium AEX410 on t
We have had challenges with the latest kernel versions on Ubuntu and sangoma
wanpipe drivers
An older kernel - no problem, latest ones, sometime risky. There are release
notes on their site stating the supported versions so it might pay to check that
But if it compiled ok it might be something
On On Wed, 10 Apr 2013, Carlos Alvarez wrote:
>>
>>> Is anyone using something to log SIP results (connected/not, latency) that
>>> they really like? We do some logging using simple scripts writing the
>>> results of sip show peers to a text file if customers report issues, but it
>>> would
On 8/03/2013, at 7:46 AM, Leandro Dardini wrote:
> If I was in your shoes, I'll check in the elastix mailing list... Asterisk
> itself can't be blamed.
>
> Leandro
>
> I am typing from my mobile phone...
>
> Il giorno 07/mar/2013 19:06, "Luis H. Forchesatto"
> ha scritto:
> Greetings.
>
On 7/03/2013, at 9:29 PM, Kamlesh Kumar wrote:
>
> On Thu, 7 Mar 2013, Bharat Lalcheta wrote:
>
> You can use ATA box with pstn phone to reduce cost.
>
> Are you wiring a building where multiple-line SIP gateways make sense?
>
> How about a description of what you are trying to do?
>
> Pers
On 6/03/2013, at 9:06 AM, John Novack wrote:
>
> Carlos Alvarez wrote:
>>
>>
>> On Tue, Mar 5, 2013 at 2:32 PM, Hose
>> wrote:
>> We have an asterisk frontend terminating all our SIP phones to, and an
>> asterisk backend with a wildcard PRI card in it connecting to the PTSN.
>> The frontend
On 8/02/2013, at 6:49 AM, Frank wrote:
> I thought about that.
> I will give it a shot tonight and will post back my results in here.
> Thanks
>
> On 2/7/13 12:39 PM, Eric Wieling wrote:
>> The easiest thing to is renumber one of the networks so they are not using
>> the same address block.
>>
On 18/01/2013, at 4:28 PM, Jim Boykin wrote:
> Hi,
>
> We are looking for the web based console for our asterisk system. We
> came across AsteriskNow but it's kind of bundle and hence not usable
> for us. What we need is a separate GUI package which we can add to our
> existing asterisk install
Hi Joe
On 18/01/2013, at 9:05 AM, Joe Ruffolo wrote:
> Hi all! In need of some serious help. We currently run Trixbox and Cent Os on
> a 2u server for our small business’s phones system.
>
> We are using some Polycom Soundpoint IP phones. The whole thing came crashing
> down over the Holidays
On 13/01/2013, at 10:52 PM, Anselm Martin Hoffmeister
wrote:
> Am 13.01.2013 03:17, schrieb Adolphus Enaboifo:
>> Hi List Members ,
>> its been about one months since I built my first Asterisk server.
>> What I want to know is: are there ways to make Asterisk take recorded
>> reminders.
>> This
On 10/12/2012, at 8:54 AM, Stephen Brown wrote:
>
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> So a friend of mine and I setup a static key based point to point
> OpenVPN connection from my box to his for the express intent of carrying
> IAX traffic encrypted.
>
> His network on his
d luck
Cheers Duncan
> On 23 November 2012 19:39, Duncan Turnbull wrote:
>>
>> On 24/11/2012, at 2:19 AM, Tiago Geada wrote:
>>
>>> Hello Folks, I am looking for a way that makes asterisk tell remote SIP
>>> party that the IP where they will send RT
On 24/11/2012, at 2:19 AM, Tiago Geada wrote:
> Hello Folks, I am looking for a way that makes asterisk tell remote SIP party
> that the IP where they will send RTP is not the same as the one I am
> comunicating via SIP
>
> Can this be done anyhow?
>
> I can try and explain:
>
> We have pla
On 14/11/2012, at 10:16 AM, bilal ghayyad wrote:
> Dears;
>
> It seems my service provider is requesting a complicated settings to allow me
> to send from behind NAT.
>
> What they said:
>
> "It shouldn't matter as long as you are handling the NAT correctly your end.
> We do not fix NAT so
On 26/10/2012, at 10:09 AM, jon pounder wrote:
> On 10/25/2012 05:01 PM, Steve Totaro wrote:
>> That is just silly. You mean to say that the Adtran and the Adit
>> units are not as reliable as these new devices. No way.
>
> I have had channel banks fail yes, and I stick by my assertion that f
On 13/10/2012, at 7:54 AM, Christopher Harrington wrote:
> On Fri, Oct 12, 2012 at 1:44 PM, Philip Bennefall wrote:
>> Hi all,
>>
>> I have an Asterisk PBX under development, that I would like to link to a
>> Skype account if possible. The idea is that people would call a particular
>> Skype u
On 10/10/2012, at 9:54 AM, cov...@ccs.covici.com wrote:
> I am sure Mikrotik routers will do this also, although I have not tried
> it.
>
Mikrotik can do this but it takes some setup. They are very powerful but what
you are asking is complex and may require the following
- 2 ethernet upstreams o
Sorry pushed send too fast
On 2/08/2012, at 5:59 AM, Eric Wieling wrote:
> Yup, there is your problem. Tell hylafax to extend the amount of time before
> it times out.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digi
On 2/08/2012, at 6:37 AM, Tim Nelson wrote:
> - Original Message -
>> Yup, there is your problem. Tell hylafax to extend the amount of
>> time before it times out.
>>
>
> We're a bit off topic for the Asterisk list now, but in your Hylafax
> config.ttyIAX0 config file, add this:
>
On 1/08/2012, at 1:59 AM, "Kevin P. Fleming" wrote:
> I've been with Digium for just over seven years, and it's been an
> incredible experience that I wouldn't have traded for anything. When
> Mark Spencer invited me to visit Digium (and Huntsville) in early
> 2005, I could not have dreamed that
On 27/07/2012, at 3:42 AM, Richard Mudgett wrote:
>> I know the topic comes back like boomerang , but I did not find a
>> nice solution.
>> Does someone has/knows how to achieve "call back on busy" otherwise
>> called camping?
>> If one is calling the extension and it is busy, then caller should
On 27/07/2012, at 8:16 AM, Alejandro Imass wrote:
> On Tue, Jul 24, 2012 at 3:54 PM, Alejandro Imass wrote:
>> On Tue, Jun 12, 2012 at 4:04 PM, Alejandro Imass wrote:
>> we upgraded to 1.8.13.1 and we have much the same problem although after
>> the upgrade I don't seem to find any cases wh
On 25/07/2012, at 7:54 AM, Alejandro Imass wrote:
> On Tue, Jun 12, 2012 at 4:04 PM, Alejandro Imass wrote:
>> This has come up before on the list and archives but I don't seem to
>> find a solution for this. On just a few nodes we have this situation
>> where we see the IP disappear from the CL
On 25/07/2012, at 6:20 AM, equis software wrote:
> I was really worried because I have a rare problem fall of link (once per
> month) and I don't know why
These errors could be reflective of a timing failure if you are generating your
timing locally instead of syncing to your carrier
If
e:
> Great tip Duncan :)
>
>
> On Thu, Jul 12, 2012 at 10:29 AM, Duncan Turnbull
> wrote:
> You can also specify routes with an callerid qualifier as 09XX/20X
>
> This would only have it apply to extensions in the 200-209 range
>
> That route can then point to a tr
You can also specify routes with an callerid qualifier as 09XX/20X
This would only have it apply to extensions in the 200-209 range
That route can then point to a trunk going nowhere if you want to block them
In freepbx there is a field in outbound route page to select callerid that the
ro
Similar problem
On 12/07/2012, at 4:36 PM, Jeff LaCoursiere wrote:
> On Thu, 2012-07-12 at 15:49 +1200, Alec Davis wrote:
>> I've seen similar.
>>
>> We tried 4 interfaces. On 4 lans, are these considered to be overlapping?
>> 192.168.1.1
>> 192.168.2.1
>> 192.168.3.1
>> 192.168.4.1
>>
>
Runnin
The module is custom contexts - its a third party option in the module admin
But you can write contexts in the extensions_custom.conf if you want to
I wouldn't use freepbx to generate your code - its quite complex code for a
roll your own system, but very useful if you learn its gui and options
Hi James
On 29/06/2012, at 6:19 AM, James Lamanna wrote:
> Hi,
> I have a bunch of different customers on an Asterisk Box (the PBX).
> This Asterisk Box is behind another Asterisk box that provides a PSTN
> connection.
> Up to this point I've been using IAX between the 2 Asterisk boxes, but
> I w
I think you need the DSN in car_odbr.ini to refer to the one in res_odbc.conf
as below
On 19/06/2012, at 3:52 AM, Thorsten Göllner wrote:
> Hi,
>
> I am trying now for over 4 hours setting up cdr-logging via odbc into a mysql
> database. But with no success. Do you have any hint for me?
>
> c
Not sure about yum installs but in 1.8 I have had to move to using odbc as the
method to populate the mysql database
http://www.voip-info.org/wiki/view/Asterisk+cdr+odbc
Cheers Duncan
On 17/06/2012, at 4:22 AM, Bruce B wrote:
> Hello,
>
> I have done "yum install asterisk18 freepbx" and it ha
On 12/06/2012, at 12:00 AM, Tzafrir Cohen wrote:
> On Sun, Jun 10, 2012 at 10:10:29PM +1200, Duncan Turnbull wrote:
>> Hi All
>>
>> Just a quick check on the best way to ensure multiple cards/devices load in
>> the correct order.
>>
>> Asterisk 1.8 wit
Hi All
Just a quick check on the best way to ensure multiple cards/devices load in the
correct order.
Asterisk 1.8 with Sangoma A101 had no problems until we introduced an Astribank.
root@pabx377:/etc/asterisk# dahdi_hardware -v
usb:001/004 xpp_usb+ e4e4:1162 Astribank-modular FPG
On 1/06/2012, at 1:24 AM, Danny Nicholas wrote:
> My guess is that your email provider is forwarding the message since Asterisk
> should send the same content to both places.
Thanks but they are two different messages i.e. one is the standard voicemail
one, the other the pager email as below
Thanks but my voicemail conf line looks like this
>> 121 => 1234,Duncan
>> testing,dun...@e-simple.co.nz,,attach=yes|saycid=no|envelope=no|delete=no
There is no pager email address so I am not sure why its sending a pager email
Cheers Duncan
On 1/06/2012, at 1:51 AM, cov...@ccs.covici.com wrot
Hi All
I am not sure why but I am getting a pager email as well as a voicemail email
when a voicemail is left. I am guessing its a setting somewhere but I can't
find it
The system is Ubuntu with Asterisk 1.8.12 from source. I am using Freepbx for
the configs but freepbx doesn't do much to voic
I had Hylafax sending 1000s of faxes a day twice a week connected to analogue
lines using asterisk and iaxmodem for about 4 years. People don't use fax much
anymore though
No problems whatsoever
Cheers Duncan
On 31/05/2012, at 6:49 AM, Danny Dias wrote:
> Just to clarify, i were using fax ma
On 30/05/2012, at 10:02 AM, Danny Dias wrote:
> Hi all,
>
> Does Hylafax and IAXmodem works with analog lines? or only with E1?
>
Hylafax can use any fax modems: available E1 or analogue, ISDN as long as it
can talk to it to send the commands
If you add asterisk and iaxmodem then hylafax can
Hi Ashish
On 4/05/2012, at 3:41 AM, Ashish Agarwal wrote:
> Hello,
>
> We are currently working on a project where using .call file on asterisk
> spool, outbound calls will be made from a pri line and a voice clip will be
> played.
>
> We know that pri has a capacity of handling only 30 chann
Hi Anita
On 4/05/2012, at 12:27 AM, Anita Hall wrote:
> Hi
>
> We are using Spandsp + FreeSWITCH for receiving Fax over T.30 E1/PRI and the
> results make us sad :(
>
I am presuming you do mean T.30 (standard fax protocol but people don't mention
it much) not T.38 as I am not familiar with t
Usually its a firewall issue, or at least it has been for me
Its saying it can't form sip packets, and that will be because something isn't
letting it,
Cheers Duncan
On 26/04/2012, at 8:15 PM, Olivier CALVANO wrote:
> Anyknow know this problems ?
>
>
> I read on the net that it's a possible
Hi
I have had issues with wiring for incoming calls causing what looks like a
hangup when answered but in those cases the call stays up and asterisk thinks
its a new call. Have seen it on Avaya too
If it is wiring can you test a different incoming line?
Cheers duncan
On 19/04/2012, at 1:54
Either give it a 2nd address on the nic that can access the VPN modem
You can have lots of addresses on a nic to access different sinners on the LAN
Or just make sure the gateway knows to route the ipvpn traffic via the vpn modem
Cheers Duncan
On 24/03/2012, at 3:55 PM, Eliezer Croitoru wr
On 5/01/2012, at 12:21 PM, James Cloos wrote:
>>>>>> "DT" == Duncan Turnbull writes:
>
> DT> I have a new install of asterisk 1.8.8.1 on ubuntu server
> DT> 3.0.0-14-server #23-Ubuntu SMP Mon Nov 21 20:49:05 UTC 2011 x86_64 x86_64
> x86_64 GN
On 5/01/2012, at 8:06 AM, Steve Edwards wrote:
> On Wed, 4 Jan 2012, A J Stiles wrote:
>
>> If you stick a /* harmless comment */ in this file and re-save it, this will
>> give the file a new modification time. Then run `make` again. It will
>> recompile just localtime.c (this being the only s
const char *name, struct state * const sp, const int
doextend)
On 5/01/2012, at 12:13 AM, Duncan Turnbull wrote:
> On 4/01/2012, at 11:47 PM, A J Stiles wrote:
>
>
>>
>> For what it's worth, I once tried installing Asterisk on an old VIA C7 box;
>> and it
On 4/01/2012, at 11:47 PM, A J Stiles wrote:
>
> For what it's worth, I once tried installing Asterisk on an old VIA C7 box;
> and it turns out that this processor, while detecting as an i686, doesn't
> implement the full i686 instruction set -- and Asterisk is trying to use one
> of the non-
Hi there
Happy New Year
I have a new install of asterisk 1.8.8.1 on ubuntu server
3.0.0-14-server #23-Ubuntu SMP Mon Nov 21 20:49:05 UTC 2011 x86_64 x86_64
x86_64 GNU/Linux
It has a Sangoma A200 card and I thought should be fairly standard but I have a
new error when trying to start asterisk
On 28/07/2011, at 8:41 PM, Paul Hayes wrote:
> On 28/07/11 02:58, Mike Diehl wrote:
>>
>> Any ideas?
>>
>> Mike.
>
> I'd go on site if possible and see what actually happens at 19:00. Set up a
> wireshark trace capturing all traffic through their router.
>
> --
I am picking a cleaner pluggi
Asterisk can run operator phones with no problem, there are multiple phones out
there with addon buttons for automating shared line appearances forwards and
other functions
For example yealink have the t38 with 6 lines and 16 buttons and the ex 38 with
38 additional programmable buttons to add
Shorewall is a useful way of setting up iptables
http://www.shorewall.net/
Cheers Duncan
On 15/05/2011, at 1:46 PM, Jeremy Kister wrote:
> On 5/14/2011 9:45 PM, Jeremy Kister wrote:
>> http://jeremy.kister.net/code/asterisk/iptables.init
>
> oops, that's:
> http://jeremy.kister.net/code/iptabl
1 - 100 of 160 matches
Mail list logo
