JR Richardson gave a very nice presentation at Astricon on how to do that with
DUNDI
check
http://www.astricon.net/files/usa06/Friday-General_Conference/JR_Richardson.ppt
http://www.astricon.net/files/usa06/Friday-General_Conference/JR_Richardson_Whitepaper.pdf
Stelios
> -Original Message--
Hi,
Anyone there could figure me out on how to install my unicall. I followed the instruction below in the stated site at; http://soft-switch.org/unicall/installing-mfcr2.html.
Questions:
1. On what directory/folder should I copy the chan_unicall.c, channels_makefile.patch?
2. On what directory/f
There is definitely wrong in your setup . I have ipkall setup on my asterisk and dont have ports 1000-2000 open ( only 1-2,5060,4569 open ) . and incoming calls word fine for me .
On 14/11/06, Al Bochter <[EMAIL PROTECTED]> wrote:
No 1000 to 2000 is not a typo.Well let me put some light on
Steve poses some good questions. In addition, I'd wonder how your trunk group in the definity is configured? Are you sending calling party name and number? If so, is your DS1 card set for protocol A,B,C, or D? For both calling name and number, I believe you need B, but I don't have my docs here
Jason,If you must stick with analog phones, you can find higher density channel banks that will host 8, 16 or up to 24 ports each. They communicate back to your asterisk server via your LAN. Or, as has been stated, you can purchase IP phones that also communicate back to your asterisk server via
Anthony Rodgers wrote:
> Greetings,
>
> Has anyone noticed that attempting to place a call from the "Placed
> Calls" list on a Polycom IP501 by pressing the 'Dial' softkey sometimes
> simply returns the phone to the idle screen? It is not related to the
> number being dialed, as we have observed t
hi, I have some qustions ... first one is ..chanspy in asterisk 1.4 version really work with whisper mode...
and second is i got following error,when i using chanspy in 1.4. version.. when i dial 6008 ,it is connected ,but i can't able to hear the voice of the any one. when coversation b
Maybe you should try this http://www.digium.com/en/products/hardware/aadk.php .
Is very heavy loaded if 9PCI cards at a server. But is possible but not encourge. Maybe you can consider to have digital extension with IP phone. THis is my opinion. :-) good luck
On 11/14/06, Jason Flatt <[EMAIL PRO
Hello all.
My company currently has an older Executone PBX system that we are outgrowing.
Rather than wait until the last minute to make a hasty decision, I thought it
would be a good idea to do some research and compare options first. My
expertise is in computers and networking, and telephon
SER: www.iptel.org
OpenSER: www.openser.org
> -Original Message-
> From: voiplist [mailto:[EMAIL PROTECTED]
> Sent: Monday, November 13, 2006 7:46 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Load balance Asterisk servers?
>
>
> We are look
Dovid B wrote:
I have a client that has a dedicated box. Running asterisk 1.2.10 with
ztdummy on Centos. He is connected via Bell South DSL in the Miami,
Florida area. He has been complaing about voice quality issues. The
person he is calling can hear him fine however he can not has terrible
On checking tail -f /var/log/atftpd, I can see that on reboots, other phones get served by the TFTP, but not the linksys ones. Now I don't understand how was it was updating itself and was being provisioned resyncing for so many hours initially, and why TFTP has stopped serving it now?
___
We are looking to be able to put a device in front of an array of
Asterisk systems which would do the job of load balancing them.
We would store all the particulars on one or more MySQL servers.
What want to accomplish is to have all calls sent to/from a single IP,
then push the calls off to ano
I was wondering how the Directory CMD can read the input of numbers from a phone and translate that to search in voicemail.conf. Essentially I want to be able to look up contacts with MySQL and have the user input 3 digits corresponding to the contacts last name and have it search for it in the da
I installed SPA942 and SPA2101, and experimented with TFTP and HTTP provisioning. It all went smooth for many hours. But then all of a sudden it stopped reading configs from both from TFTP and HTTP. Now I am trying to troubleshoot and cant't find the problem. Once in a while, it does read from TFTP
Tonight I made 3 calls, all which were answered at the remote end. All
three calls showed up in the CDR but only one showed a disposition of
"ANSWERED" the other 2 had a disposition of "NO ANSWER".
Few other things to note, on the calls with no answer the bill seconds
is of course 0. After furthe
- Original Message -
From: "Fred" <[EMAIL PROTECTED]>
To:
Sent: Tuesday, November 14, 2006 12:42 AM
Subject: [asterisk-users] "Username/auth name mismatch" + SIP phone
can'tconnect?
Hello
I'm trying to set up Asterisk on an older AMD Duron 700MHz with Fedora 5
for use with SIP p
How much did the hardware cost you to set this up for your door ?
- Original Message -
From: "mitcheloc" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Monday, November 13, 2006 11:51 PM
Subject: Re: [asterisk-users] Survey: In what ways do you
I want to setup a few remote locations with point-to-point T1s, I was wondering if it would be possible to have one host machine with two or three 4 port T1 cards, allocate a few channels for voice and the rest of the channels for data? I plan to terminate those connections to a PPPoE server. I wou
Hi List,
Does anyone know of a good dual wan router that can
handle SIP well and can failover between connections if there is a SIP issue on
one of the lines (meaning there still is a connection however there isnt enough
bandwith or sip packets arent going thru etc.) ?
Thanks.
Dovid
_
Keep in mind that CDR records show calls
sent to VM as answered, so you also have to look at the lastapp field
Disposition=answered and lastapp=voicemail
means the call was answered by voicemail (obviously)
If you are doing billing you do not care, because
the are both billable, but
You like trixbox Should try voxbox.
Best regards,
Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email
Are you outside of the US?
Do you need to call US Toll Free Numbers?
We can help you save money on calling US toll free numbers.
Email for information: [EMAIL PROTE
No 1000 to 2000 is not a typo.
Well let me put some light on this..
If you goto http://www.ipkall.com/
and your firewall is set to 1 to 2 you WILL NOT get SIP calls
from http://www.ipkall.com/ DID's
As soon as you OPEN ports 1000 to 2000 to the PBX Server the calls from
http://www
Is there an option in meetme.conf or the application meetme
to set a strict participant count limit on a per room basis?
I checked the sample meetme.conf and did a show application
meetme, as well as a couple of Google searches and came up empty handed.
This is for a system with SIP
Am Montag, den 13.11.2006, 16:51 -0500 schrieb mitcheloc:
> For about a year and a half now I've had Asterisk set up to unlock my
> front door at my house when calling a certain number. I locked it down
> by using caller id (not the most secure, but hey nobody knows the
> phone number to my door).
I've beeing using Asterisk for 3 years, principally in support of my home
office, but recently the home phones as well. I work from home full time, have
done for ten years.
At present I use Astlinux running on an H-P T5700 thin client. Prior to that it
was Astlinux on a Soekris Net4801, before
On Tue, 2006-11-14 at 09:28 +1100, Peter Howard wrote:
> On Mon, 2006-11-13 at 13:42 +0800, Rosli Sukri wrote:
> > any logs/errors when you do a verbose 6 and a sip debug ?
> >
>
> I've got a log from a call under asterisk 1.4.0-beta3 attached. The
> behaviour was the same; the call connected and
I have a client that has a dedicated box. Running
asterisk 1.2.10 with ztdummy on Centos. He is connected via Bell South DSL in
the Miami, Florida area. He has been complaing about voice quality issues. The
person he is calling can hear him fine however he can not has terrible quality
issues
Am Montag, den 13.11.2006, 23:42 +0100 schrieb Fred:
> Hello
>
> I'm trying to set up Asterisk on an older AMD Duron 700MHz with Fedora
> 5
> for use with SIP phones and the Linksys 3102 SIP gateway (ie. no FXO card,
> so no need for zaptel and libpri), but I'm stuck: The GrandStream Budg
I did not write a how to, but it essentially involves installing a
door strike that I purchased from smarthome.com, running some wiring,
and a serial port relay controller hooked up to the Asterisk server. I
then used the System command and a script to send the signal through
the board to send pow
I'm finishing up deploying an Asterisk (Trixbox) box at work. Wow, I
thought Asterisk was cool by itself, but Trixbox has made just about
everything turnkey. Great stuff!
So... we're using Grandstream GXP-2000 handsets to connect to the Trixbox,
which sits on our DMZ with a single public IP. I
On Tue, 2006-11-14 at 09:41 +1100, Peter Howard wrote:
> On Tue, 2006-11-14 at 09:28 +1100, Peter Howard wrote:
> > On Mon, 2006-11-13 at 13:42 +0800, Rosli Sukri wrote:
> > > any logs/errors when you do a verbose 6 and a sip debug ?
> > >
> >
> > I've got a log from a call under asterisk 1.4.0-b
We’re running Asterisk 1.2 on a network with a Cisco
Voice Gateway (2811) and 7940/7960 SIP Phones, and were wondering what the
recommended software versions on these were. Currently, we have 12(4)2T4
on the Voice Gateway, and 7.5 on most of the phones. Is this a fairly
standard, supporta
On 13/11/06, Al Bochter <[EMAIL PROTECTED]> wrote:
Yes you are right 1-2 are rtp ports used by asterisk by default
I have some that do set a custom range in /etc/asterisk/rtp.conf ..
After looking around.. There were not any notes about the 1000 - 2000 port
range on there website.
As you
FRom voip-info.org# SIP on UDP port 5060. Other SIP servers may need TCP port 5060 as well iptables -A INPUT -p udp -m udp --dport 5004:5082 -j ACCEPT # IAX2- the IAX protocol
iptables -A INPUT -p udp -m udp --dport 4569 -j ACCEPT # IAX - most have switched to IAX v2, or ought to iptables -A INPUT
Hello
I'm trying to set up Asterisk on an older AMD Duron 700MHz with Fedora 5
for use with SIP phones and the Linksys 3102 SIP gateway (ie. no FXO card,
so no need for zaptel and libpri), but I'm stuck: The GrandStream BudgeTone
phone fails registering with Asterisk :-/
Following the "Aste
On Tue, 2006-11-14 at 09:28 +1100, Peter Howard wrote:
> On Mon, 2006-11-13 at 13:42 +0800, Rosli Sukri wrote:
> > any logs/errors when you do a verbose 6 and a sip debug ?
> >
>
> I've got a log from a call under asterisk 1.4.0-beta3 attached. The
> behaviour was the same; the call connected and
Hi Again,
Ok so I have been able to load the driver now I need help configuring it for the PRI. The provider of the PRI is AT&T. I think I have set up zaptel.conf correctly now what is the next step?
Thanks
Julian
From: [EMAIL PROTECTED]To: asterisk-users@lists.digium.comSubject: RE: [a
Yes you are right 1-2 are rtp ports used by asterisk by default
I have some that do set a custom range in /etc/asterisk/rtp.conf ..
After looking around.. There were not any notes about the 1000 - 2000
port range on there website.
As you know if you don't know what the ports are it no
On Mon, 2006-11-13 at 13:42 +0800, Rosli Sukri wrote:
> any logs/errors when you do a verbose 6 and a sip debug ?
>
I've got a log from a call under asterisk 1.4.0-beta3 attached. The
behaviour was the same; the call connected and audio worked, but no
video.
> On 11/13/06, Peter Howard <[EMAIL
>
>asterisk-users mailing list
>To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>__ NOD32 1863 (20061113) Information __
>
>This message was checked by NOD32 antivirus system.
>http://www.eset
My * at home is a P-3 400 256 meg with a TDM 400. 2 cordless phones, 3 snom
200's. Termination is through an IAX provider. All of the standard stuff
works, transfer to cell, web voicemail,etc but the interesting thing that I
do is a script that polls the Canadian weather service every 10 min lookin
On 11/13/06, mitcheloc <[EMAIL PROTECTED]> wrote:
For about a year and a half now I've had Asterisk set up to unlock my
front door at my house when calling a certain number. I locked it down
by using caller id (not the most secure, but hey nobody knows the
phone number to my door). Speed dialing
On Tue, 14 Nov 2006, Vicky wrote:
This is more of mysql question then asterisk :D . Most voip providers use 6
second rounding for costing . My asterisk server stores call cdr's in mysql
properly with billsec field containing number of billed seconds . I want to
know some function to round this
On Mon, Nov 13, 2006 at 07:18:24PM +0100, Christian wrote:
> Hi all,
> Using the latest 1.4 of Asterisk. I have noticed that the music on
> hold files are in wav, isn't mp3 supported anymore?
It is supported as before.
However why pay the price of transcoding mp3? BTW: the files are
probably not
For about a year and a half now I've had Asterisk set up to unlock my
front door at my house when calling a certain number. I locked it down
by using caller id (not the most secure, but hey nobody knows the
phone number to my door). Speed dialing your front door is one of the
coolest things you ca
I have a query that query's my database based on the read input for an ID number.exten => s,4,MYSQL(Query resultid ${connid} SELECT\ `FirstName`\ `HomePhone`\ FROM\ `contacts`\ WHERE\ `ContactID`= \'${ID}\')
exten => s,5,MYSQL(Fetch foundRow ${resultid} var1 var2) ; fetch rowProblem is is that when
On 14:50, Mon 13 Nov 06, Earle Clubb wrote:
> The main thing I'm trying to do right now is replace my answering
> machine with *-based voicemail. I want to retain the ability to screen
> calls (listen on a speaker while a person is leaving a message), but I'm
> not sure of the best way to go ab
I'm happy to alter my AGI file in any way that will allow this to function.
If you can suggest a means of getting this to work please let me know.
I would suggest that you follow your original thought of using the file
system. You can turn on the phone's MWI by touching the msg.txt file
My s
Earle Clubb wrote:
The main reason for this e-mail is to see what other people are doing.
- What service provider/technology do you use for origination/termination?
- What hardware/software do you use and how does it all tie together?
- What tasks do you use * to accomplish?
- Any other pertinen
Donald Stahl wrote:
Right now I "Answer" and then send them into the voicemail function
with the list of mailboxes. How would I go about first recording the
message and _then_ sending it to voicemail in a loop, 50 at a time?
If there is a function I missed or am unaware of please let me know.
Right now I "Answer" and then send them into the voicemail function with
the list of mailboxes. How would I go about first recording the message and
_then_ sending it to voicemail in a loop, 50 at a time? If there is a
function I missed or am unaware of please let me know.
I was not clear on h
actually 1-2 are rtp ports used by asterisk .. its not really compulsary .. you can set a custom range in /etc/asterisk/rtp.conf .. check ur rtp.conf what range its using and open that in firewall . Default with asterisk is 1-2 unless changed .
On 14/11/06, Al Bochter <[EMAIL PRO
Donald Stahl wrote:
Define "Generate."
Right now I "Answer" and then send them into the voicemail function
with the list of mailboxes. How would I go about first recording the
message and _then_ sending it to voicemail in a loop, 50 at a time? If
there is a function I missed or am unaware of
Hey thanx for that Marnus . Thats working just exactly how i wanted :) . Damon i actually came up with same row/60+0.5 then roundup trick when i was doing something same in excel sheets :) and its usefulfor billing in 1 minute roundup ( 60 sec pulse ) but i failed to get it working for 6 second p
I was reading the posts and someone said about the default 1000 to 2000
I see in the .conf the default is 1 to 2
I found a service that gives inbound DID's in the firewall 5060 and
1 - 2 is setup
no workie on the DID
But when I set 5060 , 1 - 2 and (Unblocked) 1000 - 20
I'm trying to implement a voicemail distribution list using asterisk and
I've hit a bump. I've got an agi script that parses voicemail.conf and
generates a list of voicemail boxes to use as an argument to the
voicemail() function.
Generate in groups of 50 and loop it until you have them all?
I initiate a call with a callfile, specifying the "From" phone# as the
channel Dial(), and the "To" phone# as the Extension Dial(). I announce
the To phone# to the From listener with the A() option to the Dial()
command. It seems that the A() app plays audio while blocking return
from the F
Before I open a bug I'll ask again if anyone else is having trouble with
receiving MWI on SIP devices in 1.4. My configuration was working fine
in 1.2 but as soon as I change to any build of 1.4 I don't get
notification on any of several SIP devices. I can post my configuration
but since it w
Supposing you have an extra column called 6second:
UPDATE cdr SET 6second=billsec+(6-mod(billsec,6) where 6second=0
if you want a decimal minutes column called billmin
UPDATE cdr SET billmin=round((billsec/60)+0.5),1) where billmin=0
Vicky wrote:
Thx and what would the sql query be
? . I p
Donald Stahl wrote:
I'm trying to implement a voicemail distribution list using asterisk
and I've hit a bump. I've got an agi script that parses voicemail.conf
and generates a list of voicemail boxes to use as an argument to the
voicemail() function.
Generate in groups of 50 and loop it until
I have a two port TE205P Digium card. I have set everything up to
create a native zap bridge between the two spans. Everything works
perfectly except one thing. Our telco has a "password" that has to be
entered as soon as a long distance call is made. So if I dial a long
distance call from my
Hi
It's defiantly the branch
server. My main server handles 30 to 40 calls at a time on a regular basis. It
is only happening on the branch server and it acts like it is using up all the
bandwidth of the DSL. It is a 1.5 meg down and 512 up DSL line. I would think it
could handle 2 simul
Thx and what would the sql query be ? . I plan to put additional field as 6second . How can i make billsec of values of whole table get rounded and filled in field "6second" Sorry i am a noob with mysql :D
On 14/11/06, James Coberly <[EMAIL PROTECTED]> wrote:
sum(duration+(6-mod(duration,6) for s
All,
I'm starting to tinker with Asterisk for use in my home. Here's my
current setup:
Cox broadband telephone <--> spa3k-fxo
analog phones + answering machine (all on one line) <--> spa3k-fxs
I can pick up a phone in my house, dial a certain extension, and the
spa3k will connect me to Aste
Most usage charges are stored in various
billing databases as per MINUTE of use, not per 6 seconds of use.
6 second billing simply means that you
bill in decimal fractions of a minute, 66 seconds becomes 1.1 minutes.
1. Divide your billsec value by 60 and
round to 1 decimal place. Add
I'm trying to implement a voicemail distribution list using asterisk and I've
hit a bump. I've got an agi script that parses voicemail.conf and generates a
list of voicemail boxes to use as an argument to the voicemail() function.
The problem is that the argument exceeds 256 characters (100 mai
Dear all,
I'm trying to enable Asterisk to work with FAX using T38. I've tried
Asterisk 1.2.4 with the available patch found at URL
http://bugs.digium.com/view.php?id=5090 and also with the new 1.4 Beta3
that is announced to support it too.
With both Asterisk versions, I've sent with success
Hello All, as good?It would like to know if somebody has experience in
asterisk with ss7 protocol for isdn and asterisk with support to the
protocol sip-t.
Best RegardsJosué
___
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asterisk-users mailin
Sorry meant 2.6.12-27..
From: [EMAIL PROTECTED]To: asterisk-users@lists.digium.comSubject: RE: [asterisk-users] Modprobe ZaptelDate: Mon, 13 Nov 2006 17:31:24 +
Hi Eric, Your answer solved my problem. I did a uname -r which = 1.6.12-27mdksmp, the Makefile had -27mdkblahblahblah. Once
sum(duration+(6-mod(duration,6) for summary of seconds divisible by 6, /60 for minutes
On Tue, 2006-11-14 at 00:07 +0530, Vicky wrote:
This is more of mysql question then asterisk :D . Most voip providers use 6 second rounding for costing . My asterisk server stores call cdr's in mysql
Does it happen when you make more than one call from you main voip server alone ? Or it happens when there are more than 1 call on your branch server ? Pin the problem is in which server first , If main server can handle 2-3 calls with no lag then its probably problem in branch server .
On 13/11/
IF your asterisk server is behind NAT and no port forwarding is done then how can that static ip user/device reach it . You will have to keep asterisk server in static ip or do port forwarding to accept connections from outside .
OR maybe i didnt understand senario properly here . Is it like your
Please pardon the absolute noob questions. Someone has asked me to
interface with Asterisk and have it dial 4 numbers in succession to have
it track down an on-call person.
My initial reaction was to write an AGI program and return all 4 numbers
and have Asterisk hunt them - can Asterisk do t
Why not directly use ip address in host= line in extensions instead of dynamic address like sip.voipprovider.com .. temporary fix but it may work .
On 13/11/06, Steve Langstaff <[EMAIL PROTECTED]> wrote:
A search of google should turn up some recommendations about running alocal cacheing DNS proxy
I am planning to use asterisk with Digium TDM2404E card as a media
gateway to terminate traffic to Cell phones. Anyone got this working
before with no problmes, specially with Answer/Disconnect supervision?
Thanks
___
--Bandwidth and Colocation provided b
This is more of mysql question then asterisk :D . Most voip providers use 6 second rounding for costing . My asterisk server stores call cdr's in mysql properly with billsec field containing number of billed seconds . I want to know some function to round this to 6 seconds ( or any custom valud li
Hi
I have 2 asterisk servers
running 1.2.12.1 and IAX2 with trunking and no jitterbuffer. Both servers are
using sccp2 with 7940's and 7960's with 7914. Server 1 is my main VOIP server
and is connected to the pstn and VOIP wholesale provider. Sever 2 is a branch
site and all calls go to
Hi all,
Using the latest 1.4 of Asterisk. I have noticed that the music on hold files
are in wav, isn't mp3 supported anymore?
Many thanks,
Christian
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Setup overview:
We have an asterisk server serving a small number of SIP phones. The
asterisk server is connected to an old phone system via a T1. The
asterisk server is also connected to a second T1 used for
inbound/outbound calls.
Scenario:
We are using a call file to do auto-dialing.
OK, to simplify the reading I'll resume my problem...
Is there a way to make Asterisk send a call to Ser witch reroutes it
back to the same asterisk server ,without resulting in a "loop detected"
error in Asterisk?
Thanks,
Ricardo.
___
--Bandwid
Hello,I know little to nothing about asterisk. I am currently reading info online but was looking for other links as to good places to start. I basically just need to use asterisk to create a prototype. I need to demo a system that will allow the user to call a number, enter in some data, the ba
On 17:27, Mon 13 Nov 06, Dave Cotton wrote:
> On Mon, 2006-11-13 at 17:00 +0100, Michiel van Baak wrote:
> > On 15:34, Mon 13 Nov 06, bails wrote:
> > > We use Asterisk Desktop Manager http://adm.hamnett.org/ very
> > > successfully with both debian and windows desktops.
> >
> > Firefox can't fin
Hello,
I try to configure Asterisk to send voicemail in the language of the user's
mailbox.
But the only way I see is to modify the app_voicemail.c anybody has an
alternative idea for me ?
Lot of thanks
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Hi Eric,
Your answer solved my problem. I did a uname -r which = 1.6.12-27mdksmp, the Makefile had -27mdkblahblahblah. Once I switched the makefile everything worked. Thanks for your help.
Julian
> Date: Fri, 10 Nov 2006 12:05:37 -0600> From: [EMAIL PROTECTED]> To: asterisk-users@lists.d
Hi Moises,
Coul you give more details about how to use Cacti for CDR analysis,
there is some special pluggin, additional conf?
Your help will be appreciated.
Rgds.
On 10/31/06, Moises Silva <[EMAIL PROTECTED]> wrote:
of course you can always use http://cacti.net/download_cacti.php
On 10/31/0
On Monday 13 November 2006 11:49, Yu Safin wrote:
> when you use snap, does the call go to your iax hardphone connected to
> asterisk or do you need a softphone on your PC?
Please trim your posts, you don't need to keep the headers and signature lines
of the entire thread to ask one sentence, now
Hi all,I've 2 * servers with static IP, and i notice that if i set both sip peers with host=server_ip and qualify=yes it presents UNREACHABLE on asterisk CLI.When i changed the host parameter to host=dynamic and set the register string in the [general] of
sip.conf on both servers, the connection h
joe a.<[EMAIL PROTECTED]> Wrote on: 11/13/2006 10:37 AM:
> Making custom "voicemail greetings" seems fairly straight forward, and
> I've done it.
>
> However, I'm looking for a way to make the actual extension answer with
> "You've reached my Jim Dandy voice mailbox, go take a flying . . .".
>
I have this very
weird situation where some callers hear the playback of sound prompts on half
speed. It only lasts a few second but it can happen at any time during playback.
My server is a 3.4
Ghz Xeon with 1 GB RAM and 80 GB SATA disk. I run Asterisk 1.2.13 on FreeBSD
6.1
Anyone who h
Am Montag, den 13.11.2006, 10:37 -0500 schrieb joe a.:
> Making custom "voicemail greetings" seems fairly straight forward, and I've
> done it.
>
> However, I'm looking for a way to make the actual extension answer with
> "You've reached my Jim Dandy voice mailbox, go take a flying . . .". (OK,
Hi:
I'm having audio dropouts in ChanSpy when the call is originated in a
Unicall (E1/MFCR2) channel and the destination is an Agent using a SIP
phone. If the agent is using a traditional phone (going from the PBX to
asterisk via another Unicall line) no dropouts are present. The dropouts
seem
List members,
Is it possible to record outbound analog calls using an X100P?
I was asked if I knew how to record all calls for a shop with 4 analog
phones transparently to the end users. I thought Asterisk was a good
fit for this and I envisioned using either Digium TDM400Ps or Sangoma
A200s
On 11/13/06, Steven <[EMAIL PROTECTED]> wrote:
I have been using http://www.snapanumber.com/ 's Windows tray utility, and it
works great.
--
--
Steven
http://www.glimasoutheast.org
"Ondrej Valousek" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED]
Hi all,
I am interested in inte
On Mon, 2006-11-13 at 17:00 +0100, Michiel van Baak wrote:
> On 15:34, Mon 13 Nov 06, bails wrote:
> > We use Asterisk Desktop Manager http://adm.hamnett.org/ very
> > successfully with both debian and windows desktops.
>
> Firefox can't find the server at adm.hamnett.org.
Just downloaded it wit
On Mon, Nov 13, 2006 at 07:10:12AM -0500, Brian Rogan wrote:
> On Mon, Nov 13, 2006 at 12:46:14PM +0100, nik600 wrote:
> > Hi
> >
> > i have an application developed with bayonne.
> >
> > Recentely i'm experiencing some problems and i am planning to migrate
> > to asterisk.
> >
> > I would like
I've figured out the problem, alaredy, and posted in another thread,
but no one seems to have an answer yet.
It is a problem in the IAX trunk. If I turn the jitterbuffer on, I
get one-way-audio when I put someone on hold. If I turn the
jitterbuffer off... I still have two way audio. THis is ru
A search of google should turn up some recommendations about running a
local cacheing DNS proxy, or similar.
I've never done it myself (the "cacheing proxy", not the "searching on
google") so I don't know the specifics.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PRO
Dear all,
My architecture is having some problems with redirects. In the following
diagram is shown a simple erroneous test. When someone dials from the
PSTN, signalling of the incoming call is passed to Asterisk which routes
to SIP Express Route (Ser), and then Ser routes to the phone. The us
I believe that the problem really is fault of DNS lookups, but as I
should proceed for resolve that??
see the first point at
http://www.voip-info.org/wiki/view/Asterisk+administration
The best solution for now is probably to have a caching dns server on
your Asterisk box or in your LAN
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