On 28 Mar 2011, at 14:19, vip killa wrote:
> Yes I followed directions on that page
> Running Asterisk 1.6.1.22, anybody else experiencing this?
How often does fail2ban check the logs? It can only block that often, so if
more attempts happen in that time period it can't do anything until it knows
On 24 Mar 2011, at 16:46, tahar .H wrote:
> so plz is there any one who can give me a puch to learn this extraordinary
> Asterisk plz(video things will be better :))
Learn to ask questions. Learn to read books. Learn to use google.
S
--
___
On 24 Mar 2011, at 16:38, Gordon Henderson wrote:
> 1.2 has been the most stable version for me.
>
> Same setups with 1.4 +DAHDI has never been as stable with random crashes and
> re-starts - however they're not predictable and sometimes months apart. I had
> one instance of 1.2 run for over a y
On 23 Mar 2011, at 10:40, Nikhil wrote:
> I am planning to use asterisk as a IP phone(Porting asterisk into a hardware).
Interesting..
> Is there any limitations if I use asterisk as a SIP client?,and asterisk has
> any advantages if use like this?
It's not really designed as a SIP client. It's
On 22 Mar 2011, at 01:09, Outback Dingo wrote:
> Even worse... now it smells of a scam
At least their website isn't hideous...
Oh..wait.. ;)
S
--
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New to A
On 15 Mar 2011, at 15:21, Jonas Kellens wrote:
> On 03/15/2011 12:39 PM, Steven Howes wrote:
>>
>> On 15 Mar 2011, at 11:30, Jonas Kellens wrote:
>>> On 03/15/2011 12:24 PM, Steven Howes wrote:
>>>>
>>>> On 15 Mar 2011, at 09:08, Jonas Kellens wrot
On 15 Mar 2011, at 11:30, Jonas Kellens wrote:
> On 03/15/2011 12:24 PM, Steven Howes wrote:
>>
>> On 15 Mar 2011, at 09:08, Jonas Kellens wrote:
>>> I also notice the presence of a "Remote-Party-ID" SIPheader... Where does
>>> this come from ?! Not
On 15 Mar 2011, at 09:08, Jonas Kellens wrote:
> I also notice the presence of a "Remote-Party-ID" SIPheader... Where does
> this come from ?! Not from my dialplan...
sendrpid in your sip.conf
Steve--
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On 14 Mar 2011, at 16:24, satish patel wrote:
> I test page application and it works but i am worried about i have 200 SIP
> phone. Do you think asterisk page application can handle that number of page
> ?
Do they support multicast?
S--
_
On 14 Mar 2011, at 15:58, Jonas Kellens wrote:
> dialplan :
>
> exten => 67121212,1,NoOp()
> exten => 67121212,n,Set(CALLERID(all)="3259" <3259>)
> exten => 67121212,n,SIPAddHeader(P-Preferred-Identity:
> )
> exten => 67121212,n,SIPAddHeader(Privacy: id)
> exten => 67121212,n,Dial(SIP/32
On 3 Mar 2011, at 20:53, Danny Nicholas wrote:
> Not having an in-depth knowledge of how EU numbering works
Sadly there is no 'EU numbering'. Europe isn't a country, thus doesn't share
any dial plan. There appears to be some tendency towards having a '7' at the
front of a mobile number, but it's
On 28 Feb 2011, at 10:33, Rizwan Hisham wrote:
> The problem I have been experiencing since last month is that some of my
> customers are getting calls with "Asterisk " caller id. Most of them
> in the middle of the night. And my asterisk server has no record of these
> calls. The customers were
On 17 Feb 2011, at 17:52, Fred Posner wrote:
> Awesome. Any institution that issues a masters to someone who asks a
> question to a mail list on open-source software must be held in high
> regard.
Careful! You might end up with a Masters in Debating.
S
--
On 17 Feb 2011, at 10:04, Nikhil wrote:
> Do I need to modify chan_phone application to make it works or it is
> available in net.
Why not use a proper sip client?
S
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Same here. But, can the genie ever be put back in the bottle?
> Cary Fitch wrote:
>> Has anyone else noticed "new spam" in the last 2-3 weeks?
>>
>
> No,
>
> But I run ASSP in front of my MTA.
>
> Doug
--
_
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conds into the call.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steven C. Blair
Sent: Wednesday, August 25, 2010 2:08 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk 1.6.1.17 ACK/BYE question
We'
We're running Asterisk 1.6.1.17 for our campus voicemail server and Juniper
M120s as our SBC. Unanswered calls, which arrive via the SBC, are diverted to
voicemail using a 302 redirect when the called party doesn't answer. In this
case the caller is able to hear the greetings and begin to lea
I'm using P0S3-8-12-00 and things are working great with piaf and asterisk
1.4. Drop me a direct line to email, and I can send you my configs and such
if that would help diag things for you.
On Feb 3, 2010 3:02 PM, "i...@comtek.co.uk" wrote:
David Gibbons wrote:
>
> I have upgraded the phones t
Sorry to bump this one...
Anyone have any other ideas on it?
Regards
Steven Davison
Net Technial Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steven Davison
Sent: 21 January 2010 08:41
To
Hi,
Couple of questions...
Are you allowing reinvites, and what happens if you change the dialplan to this?
exten => 1,n,Dial(SIP/14168724...@6135551212-sw1|120|gtT)
exten => 1,n,Playback(vm-goodbye)
exten => 1,n,Hangup()
help this helps :)
Steven Davison
Net Technial Soluti
You last question : why are DTMF tones not audible in the recording?
WE had issues with DTMF not recording, and found it was due to the handset only
sending the DTMF in data, rather than inline, as a beep... that could be your
reason :)
Steven Davison
Net Technial Solutions
-Original
l address.
The phone site has no static Nat in place for Sip or RTP, so we are reliant on
the routers ability to sort that out. There is a firewall on that router, which
allows ALL traffic out, and also allows SIP and RTP in.
Hope that clears up a few things! :)
Steven Davison - Network Engineer
t:
This email is not a question, but a potential solution to any who have
had the same issue I have faced.
If you have agents logged in to multiple queues at the same time,
Asterisk does not handle the answering of those queues in any set order
or sequence. It has no way of prioritizing calls in
r the configs on the routers, sip.conf etc trying
to work this out... we have also checked that the users are using the above
sequence to transfer a call...
Thanks to anyone who may have ideas for this... ☺
Steven Davison - Network Engineer
t: 0845 0034567
f: 0845 0034543
w: www.ntsols.com
Is SELinux enabled on the machine? If it is, you might have a problem with
the asterisk process being able to execute in that directory.
On Wed, Nov 11, 2009 at 4:38 AM, her Garcia wrote:
> Hello. I am trying to execute an fax reception script and i am getting the
> following:
> [Nov 11 08:40:52
I'm under the impression that this sometimes happens when a firewall
decides that the port you've opened no longer needs to be so. Are you
using sip_nat? Do you have a firewall between the asterisk host and
public? How are your VoIP related firewall rules configured?
Has anyone seen somet
/automatic-asterisk-extensions-for-skype.html
Enjoy,
-S
--
Steven Sokol
Digium Inc. | Product Manager - Asterisk
1568 S. Yorktown Place - Tulsa OK - 74104
direct: +1 256-428-6101
mobile: +1 816-806-8844
fax: +1 816-817-0441
twitter: ssokol | jabber: sso...@digium.com | skype: ssokol.digium
Visit
On about 25% of inbound calls to a ring group, picking up any one
extension as it rings results in dead air.
Some details regarding my VoIP network to make the following logs more
readable:
192.168.7.130 resolves to the trixbox host.
192.168.7.135 resolves to endpoint 812.
192.168.7.137 resolv
--[ UxBoD ]-- wrote:
> - "Steven J. Douglas" wrote:
>
>> --[ UxBoD ]-- wrote:
>>
>>> - "Gordon Henderson" wrote:
>>>
>>>
>>>> On Fri, 1 May 2009, --[ UxBoD ]-- wrote:
>>>&
Hi Jonas,
Maybe you can try leaving out bindport and bindaddr parameters first.
The port defaults to 4569 anyway. As for the bindaddr, you should be
using the IP Address of your interfaces. I am assuming you are using the
IP Address obtained from your router. If that is the case, then asterisk
--[ UxBoD ]-- wrote:
> - "Gordon Henderson" wrote:
>
>> On Fri, 1 May 2009, --[ UxBoD ]-- wrote:
>>
>>
>>> Okay, getting somewhere now ! I am now getting the following :-
>>>
>>> == Starting post polarity CID detection on channel 1
>>> -- Starting simple switch on 'DAHDI/1-1'
>>>
Thanks to all who replied. The problem was due to a faulty NTU box from
the telco. It has been up for almost a week now without any downtime.
Regards,
Steve
Steven J. Douglas wrote:
> Thanks for the tip, Harry. I will try that when I have exhausted all
> avenue. My problem is that if I u
) I updated to Asterisk 1.4.24, replaced Zaptel with latest
> DAHDI. In the DAHDI case I even had to use latest Subversion revision
> due to some bug (but that was related to the TE121-cards I think).
> Since then I haven't had any issues at all, so consider updating
> Asterisk and Z
Maybe your network is not ready when asterisk first fires up?
-steve
Yahya Mohammad wrote:
> I'm running asterisk on Ubuntu 8.10. I have two 'register' lines in
> iax.conf for registering with two remote servers. However only the
> first one registers at system startup. I always have to issue an
.org/wiki/view/crossover+T1+cable
>
> On Tue, Mar 31, 2009 at 12:37 AM, Steven J. Douglas <mailto:stev...@moij.biz>> wrote:
>
> Hi guys,
>
> I've been trying to get my ISDN-10 line up for the past few days, but
> its been going up and down. I am using Ope
Hi guys,
I've been trying to get my ISDN-10 line up for the past few days, but
its been going up and down. I am using OpenVox D110P card on
asterisk version 1.4.21. It seems to me like a cable problem. I tried
using Ethernet straight cable (12, 45, 36, 78) and also a "straight"
cable where
Hi Ken,
If you run "ulimit -c" on the command line and get a "0" output, then
you need to run "ulimit -c unlimited" on the command line.
-Steve
Ken D'Ambrosio wrote:
> Asterisk segfaulted on me the other day; how do I tell it to generate a
> core file so -- if it happens again -- I can attempt
Thanks! I'll give that a try.
Regards,
Steve.
Tilghman Lesher wrote:
> On Monday 02 March 2009 00:27:00 Steven J. Douglas wrote:
>
>> Hi,
>>
>> My asterisk segfaults a few times each day and the crash problem seems
>> weird. When I run gdb on the core
Hi,
My asterisk segfaults a few times each day and the crash problem seems
weird. When I run gdb on the core dump, it almost always segfaults on
free() or malloc(). When I run the back trace, I see something weird.
Here's one of the back traces.
#0 0x4017f87f in _int_free () from /lib/libc.so
Hi,
Have you tried using "externip" in your sip.conf? By setting the correct
"localnet", any SIP packets that goes elsewhere will use the value in
"externip". This might solve your problem.
Regards,
Steve
nik600 wrote:
> On Sat, Feb 7, 2009 at 8:31 AM, nik600 wrote:
>
>> hi
>>
>> is it pos
I think you need to use ParkAndAnnounce instead of Park to get the call back.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeremy G. Gault
Sent: Wednesday, February 04, 2009 11:16 AM
To: Asterisk Users Mailing List - Non-Commercial D
Hi Lincoln,
The fact that you can hear and respond to the voice mail (even if its
for the first 20 seconds), means that your phone has received the OK
message properly. The problem is the missing ACK after receiving OK.
When asterisk did not receive the ACK after a few retries of the OK, it
te
sip-trunk-100-1] user 100102
> [sip-trunk-100-2] and user 100103 to use [sip-trunk-100-3] for outbound
> calls. can i route it based from the source. TIA
>
> Regards,
> Nhadie
>
> Steven J. Douglas wrote:
>
>> Use different context for both users in sip.conf. In
Don't use g729 in the iax.conf for the IAXY device. It doesn't support it.
Regards,
Steve
Adam Robins wrote:
> I am using a Polycom SIP phone (ext 2042) to call an analog phone
> connected via an IAXY (ext 2120). The analog phone rings, and when I
> answer, I can hear the person speaking on the
Hi Dimitar,
You can use the Read command in your 5051 extension to wait for a
response after the user answers the phone.
Regards,
Steve
Dimitar Dimitrov wrote:
> Hi,
> I have some troubles with early media with Zapatel TDM400P adapter. I
> made a simple callback function wich works by followin
In your Read command, leave out the .wav extension in the file name.
Regards,
Steve
Artifex Maximus wrote:
> Hi all!
>
> I would like to make a service with recording sounds and playing back
> to caller. I had wrote the script but it failed at Read statement with
> file not found error. I have pu
Use different context for both users in sip.conf. In the context for
user 100300, include the context sip-trunk-100. For user 101300, include
the context sip-trunk-101.
Regards,
Steve
Nhadie wrote:
> Hi,
>
> Is it possible to detect where the call came from and route it out to
> different sip
Hi Steve,
Thanks for the tip. But unfortunately it doesn't help. The Avaya is
passing on the MFC codes to the SIP phone when it answers the call. I
think the solution might be in the Avaya configuration to properly
convert the signaling.
Regards,
Steve
Steve Totaro wrote:
> Answer() is the cu
a did not recognize this and stop the ringing
on the PSTN side. I'll give your suggestion a try and see if it makes a
difference.
Thanks.
-Steve
David fire wrote:
> try a answer() before the dial(sip/xxx)
> and if you are using originate try local/ and start whit and an
Hi,
I'm trying to integrate my asterisk PBX to an Avaya PBX. I am using
chan_ooh323 from asterisk-addons.
I am able to make a call from SIP Phone -> Asterisk -> Avaya -> Station
(phone) and vice versa.
I am also able to make a call from SIP Phone -> Asterisk -> Avaya -> PSTN.
However I face pr
here that might be able to give
me a few pointers? Any and all help is much appreciated.
Thanks.
Kind Regards,
Steven Moughan
-
LAKE Communications,
Beech House, Greenhills Road,
Dublin 24, IRELAND
int. +353 1 4031112
fax. +353 1 452
On 24 Oct 2008, at 03:57, David Gibbons wrote:
> Dare I ask why you want to do this?
Cheaper than buying an AIM-CUE? And certainly more flexible.
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To U
On 22 Oct 2008, at 20:29, Craig Van Ham wrote:
>
> HI all,
>
>
This appears to be the same message you posted earlier.
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On 16 Oct 2008, at 14:57, jonathan augenstine wrote:
> I am trying to build app_confcall and it is failing. Are there
> known build issues with this module. I am running Asterisk 1.6.0-
> beta9.
Ah yes. 'failing'. I bet that is all it says eh? its not like
compilers give descriptive errors
On 14 Oct 2008, at 18:05, Christian Victor wrote:
> Steven Howes schrieb:
>> Have created a system that involves using call files in the outgoing
>> spool folder. On some occasions it retries which is fine is there
>> any way to view calls waiting retries from the CLI? U
Hi All,
Have created a system that involves using call files in the outgoing
spool folder. On some occasions it retries which is fine is there
any way to view calls waiting retries from the CLI? Using 1.4 btw.
Have googled to no avail (although it is near the end of the day so I
might
On 14 Oct 2008, at 11:00, Chris Rowson wrote:
>> Hi folks,
>>
>> I'm working on a solution using the Asterisk voicemail component and
>> wondered if anyone knew the answer to this question please?
>>
>> I understand that Asterisk saves voicemail to
>> /var/spool/asterisk/voicemail///INBOX/ but I
>
Hi
triXbox.org can answer these questions. Google may also give a
balanced view. But yes, i can assure you, people are using Trixbox
from Fonality.
Steve
On 6 Oct 2008, at 10:24, broadband Voice wrote:
> Anyone using Tribox from Fonality. I understand its open source and
> free. Can I use
Just copy the src folder and do `make install` on each machine?
Then tar and copy the /etc/asterisk folder if config is important too.
On 29 Sep 2008, at 08:41, Jim Boykin wrote:
> Is there a script to create an Asterisk binary package after it is
> compiled on one system.
>
> We do not want to c
On 25 Sep 2008, at 18:38, Shyju K wrote:
> I was configuring asterisk with TE110P Card.When run zttool
> It is showing "Blue Alarm/Yellow Alarm/Recovering" and the
> card's LED is blinking RED and GREEN.
> I have connected 1&2,4&5 Lines from ISDN modem(RAD ASMi-52)
> to 1&2,4&5 of the PRI card r
Hi,
Agreed. Asterisk on a VM appears to work sometimes, only if magic is
involved. It is not the way to run anything for a business.
Steve
On 25 Sep 2008, at 02:36, Dean Collins wrote:
> Mike,
>
> Buy an asterisk appliance like http://www.taa.com/products-vdex-40.html
> problem solved.
>
>
Hi,
We saw this between Asterisk and an Audiocodes gateway. Whilst the
voicemail is being recorded asterisk is not sending *ANY* rtp. Silence
detection will always detect silence if it listens to this side of the
conversation. Adjusting the threshold wont work, you need to find the
timeout
Perhaps you could supply some sort of log.
On 19 Sep 2008, at 08:41, Brad wrote:
> we just did a brand new installation of asterisk 1.4 on ubuntu with
> a sagnoma t-1 card
>
> everything went smooth (other than fighting a little outbound call
> issue that we are sure is a tdm network to sagno
n Öberg, Skype: VoIP For Business
* 5 Tracks (Business, Tech-Intro, Carrier/Large Scale, Advanced 1, Advanced 2)
* CodeZone - 9:45 AM - 1:00 AM
Register Now: http://www.astricon.net/2008/glendale/web/attendRegister.php
See you in Phoenix.
-S
--
Steven Sokol
AstriCon 2008
P.S. Remember, ticke
On 12 Sep 2008, at 10:13, Tim Panton wrote:
>>
> I'd guess the battery on your motherboard has died so it is going back
> to 1970 at
> boottime.
>
> Watchout, because this can also mean that your BIOS is about to
> loose all settings too which can cause it to forget how to talk to the
> harddrive
Press *
On 11 Sep 2008, at 14:31, Joseph L. Casale wrote:
> Now that we have voicemail working, people have asked to be able to
> dial in externally and be able to access their voicemail. My dial
> plan is
> simple, after ringing a few extensions for some time, it goes to
> voicemail.
> What
glendale/web/attendRegister.php
If you've not yet registered, please get signed up as soon as
possible. The main hotel is rapidly running out of rooms --
fortunately there are plenty of alternate hotels within walking
distance. Remember that prices go up by $100 once the conference has
started
That us a bit like wanting to know what the person calling you wants
to talk about without picking up the phone..
On 8 Sep 2008, at 17:42, JD wrote:
> Generic question,
>
> Is there a way to detect a fax call without actually taking it as a
> fax
> call? In a non-universal manner?
>
> In other
On 7 Sep 2008, at 15:06, Bruce Komito wrote:
> I recently installed 1.4.21.2 on Debian 2.6.18-6 and since then, I am
> experiencing occassional garbled voicemail messages. Specifically,
> what
> happens is that the first 15-20 seconds of the message is fine, but
> sometimes after that the sound
On 2 Sep 2008, at 10:36, Steven Howes wrote:
> asterisk-1.4.21.2
> libpri-1.4.7
> zaptel-1.4.11
>
> I might be being a muppet here (not used PRI with Asterisk before) so
> humor me.. I am using SetCallerPres on an outbound call over PRI...
> Console shows:
>
> -- Ex
asterisk-1.4.21.2
libpri-1.4.7
zaptel-1.4.11
I might be being a muppet here (not used PRI with Asterisk before) so
humor me.. I am using SetCallerPres on an outbound call over PRI...
Console shows:
-- Executing [EMAIL PROTECTED]:8] SetCallerPres("SIP/XXX.XXX.
209.243-08b81d68", "prohib_
Sip debug please.
On 1 Sep 2008, at 10:07, daniele visaggio wrote:
> Hi guys,
>
> I need to create a SIP trunk between my * (trixbox) and a legacy
> Samsung pbx. I create the SIP trunk as usual: the calls from my * to
> the Samsung pbx worked immediately, but I can not place any calls
> fro
, 2008 at 6:47 PM, Steven Howes <[EMAIL PROTECTED]>
> wrote:
>> Did you tab complete it to make sure it was right?
>>
>> On 28 Aug 2008, at 11:39, Rilawich Ango wrote:
>>
>>> I got the message below after I issue the soft hangup.
>>> sip01
asterisk linkedin group
I have created an asterisk linkedin group for anyone interested.
http://www.linkedin.com/e/gis/45252/66270A773F53
Thank You,
Steven BerkHolz
- MCSA - MCSE -
Manager of Information Systems
HIROTEC AMERICA
Please visit us on the web at
y to kill the call without affecting other queues and
> calls?
>
> On Thu, Aug 28, 2008 at 4:09 PM, Steven Howes <[EMAIL PROTECTED]>
> wrote:
>> Try CLI> soft hangup Local.
>>
>> On 28 Aug 2008, at 09:01, Rilawich Ango wrote:
>>
>>> Hi ,
&
Try CLI> soft hangup Local.
On 28 Aug 2008, at 09:01, Rilawich Ango wrote:
> Hi ,
>
> Actually, there are 3 queues in the server. Only one queue (2700)
> has problem. I want to reset or remove the caller only in 2700
> without affecting other queues or calls. Does it work for my case?
>
>
On 28 Aug 2008, at 08:22, Andreas M. wrote:
> http://www.api-digital.com --
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asterisk-users mailing list
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On 27 Aug 2008, at 14:21, Olivier wrote:
> I think we're getting closer now as obviously Asterisk's greeting
> ("...UNIX connection") is mixed with its output.
> (I can't understand why this happens now as I never noticed this
> before and didn't change anything).
>
> I tried to use asterisk
On 27 Aug 2008, at 13:23, Olivier wrote:
> 2008/8/27 Steven Howes <[EMAIL PROTECTED]>
> Probably another left over word from another message. Is it
> repeatable?
> At the moment, yes.
>
> Now, I'm looking for a way to flush input/output, to protect shell
> scri
Probably another left over word from another message. Is it repeatable?
On 27 Aug 2008, at 13:00, Olivier wrote:
> Hello,
>
> On a 1.2 Asterisk / Debian Sarge, I noticed that :
>
> ipbx*CLI> sip show peers
> Name/username HostDyn Nat ACL Port Status
> 4201/4201
On 26 Aug 2008, at 18:33, Drew Gibson wrote:
> Is there a maximum string length for use with the legacy interface
> chan_string?
> Does it depend on the type of cup used? Does styrofoam give better
> range
> than paper?
>
> regards,
>
> Drew
DTMF modes include: as audio, tugging on the string c
On 5 Aug 2008, at 09:16, Budacsik Attila wrote:
> Hi Everyone,
>
> I am currently running Trixbox 2.6 and I have a problem with Asterisk.
>
> /etc/init.d/asterisk status
> Asterisk dead but subsys locked
>
> I deleted all files in /var/run/asterisk folder and asterisk
> restart...
> It's ok for a
On 29 Jul 2008, at 15:31, Steve Totaro wrote:
> Yes.
Beat me to it ;)
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asterisk-users mai
On 22 Jul 2008, at 14:36, Nhadie wrote:
> Pinging my.sipserver.com [202.203.204.205] with 32 bytes of data:
>
> Reply from 202.203.204.205: bytes=32 time=250ms TTL=56
> Reply from 202.203.204.205: bytes=32 time=250ms TTL=56
> Reply from 202.203.204.205: bytes=32 time=651ms TTL=56
Never going to w
Fail.
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Afternoon All,
Does anyone here have any experience with an Audiocodes Mediant 2000?
I know its a bit 'non asterisk' but i figured you guys are as likely
as any to have come across them. I'm having a few problems with one,
i.e. its not sending screening/privacy flags although it is sending
Fail.
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On 4 Jun 2008, at 11:43, Joao Ferreira gmail wrote:
> can I connect 2 FXS plugs to the same analog phone ?
No. Fire and death.
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Hi All,
We have a Cisco CME linked to our Asterisk PBX (named 'epstein'). Off
said PBX we have numerous other PBX's, some IAX and some SIP. On a
call placed from CME (SIP) to 'epstein' it all works fine except for a
few quirks.
When calling through epstein to an IAX peer we get '100 trying'
On 19 May 2008, at 05:42, Lee, John (Sydney) wrote:
> Lee, John (Sydney) would like to recall the message, "[asterisk-
> users] Newbie Asterisk: Install Asterisk as non-root".
>
Fail.
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On 18 May 2008, at 23:42, Andrea Cristofanini wrote:
> Hi
> I just saw this now !
> does the microphone and speaker works ?
> Can you use it like softphone for recive calls ?
> Regards Andrea
Since when is Asterisk a SIP client.
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> Is there any reason why I should be experiencing such bad line quality
> on inbound calls from PSTN? Call quality is perfect when plugging in a
> regular analogue phone.
So outgoing PTSN calls are fine but incoming PTSN calls have poor
quality. Do both parties hear the crackling, etc? Can y
Jerry Geis wrote:
> I have xinet tftp running on centos 5.1
>
> It seems to be running on the local network eht0 fine. My box has 2 nics.
> however when I connect to eth1 for tftp I get:
>
> in.tftpd[5084]: tftpd: read(ack): Connection refused
>
> How can I get tftp working on BOTH eth0 and eth1 f
han all of the RTP ports
required for SIP.
--
--
Steven
http://teamvie.blogspot.com/
http://www.connectech.org/
"equis software" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED]
Hi, I need to make Click-to-Call web application to connect with an asterisk
server.
I´
ve used it.
Some do not want to open up the iax2 port in their firewall, but that is their
issue.
I wanted to use IAX2 because I knew with NAT and firewalls, that IAX2 was
easier for people to use than all of the RTP ports
required for SIP.
--
--
Steven
http://teamvie.blogspot
Brian J. Murrell wrote:
> Does anyone have an implementation of this they'd like to share?
>
I cut out the authentication stuff we do, but this is part of the macro
we use to spy and record calls arbitrary calls. All of our users have
sip handsets. Asterisk 1.2.
exten => s,n(getext),Read(SP
We are using Asterisk and SER with Polycom 550 phones running SIP version
2.2.2.0084. The phones register to SER. If an AOR appears on more than one
phone when a call arrives for that AOR one, some or all of the Polycom phones
reboot. I can't seem to find the source of this problem. Has any
John covici wrote:
> OK, this is exactly what I would like to do, can you either write me
> on or off list for further details. This would be the first baby step
> toward the 20th Century!!
I'd love some pointers on integrating * with a sx-200. I have a system
where a fork lift upgrade is imposs
Marius Muja wrote:
> My guess is that the asterisk server tries resolving the names of the
> SIP providers when it tries to re-register to them and because there
> is no internet connectivity it hangs there for a while. However in
> that time all the local calls to the asterisk server stop worki
rther comments, concerning my improper use of the list,
off-list.
Thank You,
Steven B
[EMAIL PROTECTED]
http://teamvie.blogspot.com/
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