For the time being try putting 212.41.253.181in hostname= line in ur sip config and it should work . Also check if you /etc/resolv.conf has correct dns list ( i guess it does bcoz OS can resolve) . Also check /etc/asterisk/dnsmgr.conf .
Here's xample :[general]enable=yes ; enable crea
Recently, we just migrate our PABX from 1.0.10 to the latest asterisk
1.2.13, most of the features are migrated smoothly, only blind
transfer behaves differently, and quite annoying,
We use options 't' and 'T' in dial command to enable using asterisk
feature to do blind transfer, in 1.0, after yo
Yeh voxee is really of no use now . You can try voipjet though .. even though they dont have good support but i hardly had any big problems with them . Also there is icall but i wont recommend that right now but their rates are pretty good .
- Original Message - From: [EMAIL PROTECTED] To
On 00:38, Fri 10 Nov 06, Christian wrote:
> Hi all,
> Since i cant get latet beta of zaptel installed on the latest test version of
> Debian with kernel 2.6.17-2-686 can someone who is using debian give me some
> tips on how to get it working and installed?
> Many thanks,
> Christian
Chris,
You
Asterisk Development Team ha scritto:
The Asterisk Development Team is pleased to announce the release of
version 1.2.11 of Zaptel.
Where is it??? The link on asterisk.org is broken...
Also, no Changelog anywhere.
--
--
Alberto Pastore
B-Press Srl - Gruppo MSoft
P.IVA 01697420030
P.le Lomb
I’m running asterisk 1.2.8. I would like PSTN inbound
calls to do the following:
1-once PSTN callers enter their desired extension; they have
to record their name
2-recording then announces that it is trying to locate the
user
3-asterisk calls local extension and announces callers
re
Never mind I got DID billing to work with a2billing
it was in the conf files
needed retyped to the right info.
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 ca
Bruce Ferrell wrote:
I'm having problems with a new asterisk PBX install. the phones/ATAs
are all linksys/cisco. They all worked before with a commercial
softswitch.
Most of the linksys devices offer auto, inband, INFO and AVT. I'm
looking for suggestions.
I believe that AVT is RFC283
On Fri, Nov 10, 2006 at 02:34:31PM +1100, Paul Hales wrote:
>
> I had a 200, and it worked fine with POE.
>
> The standard power connector was the RJ-11 style as mentioned below.
> Weird item that one.
>
> The successor to the 200, known as a 190 does NOT support poe, while the
> 320 does.
>
Y
I had a 200, and it worked fine with POE.
The standard power connector was the RJ-11 style as mentioned below.
Weird item that one.
The successor to the 200, known as a 190 does NOT support poe, while the
320 does.
later,
PaulH
On Thu, 2006-11-09 at 22:13 -0500, Christopher Aloi wrote:
> > He
Dovid B wrote:
Are you trying to get FOP to monitor
the SIP account that you are using to dial the cell phone on ?
The SIP extension, yes. So, as long as a call that has been forwarded
to that cell phone is still in progress, that extension should still
show busy.
Thanks again,
Hi Alls,In Asterisk-1.4 there is new config file, users.conf, but i don't know how mechanism between users.conf and sip/iax.conf, usually i add new user in sip.conf, but when i try use asterisk-gui, it write to
users.conf
and when i type "sip list peer" on asterisk console, there is no user that
Hey Brad -
I have a Snom 200 at the office, If I remember correctly the power is
about the size of an RJ11 jack; it's a weird connector.
I haven't used PoE with the 200 though.
Not sure if that helps or not :)
-chris
On 11/9/06, Brad Templeton <[EMAIL PROTECTED]> wrote:
>
> Ok, not exactl
I'm having problems with a new asterisk PBX install. the phones/ATAs
are all linksys/cisco. They all worked before with a commercial softswitch.
Most of the linksys devices offer auto, inband, INFO and AVT. I'm
looking for suggestions.
Thanks in advance
--
One day at a time, one second
Ok, not exactly an Asterisk problem, but...
I picked up some SNOM 200 phones because SNOM's have been recommended for use
with Asterisk and they have line buttons that can subscribe to presence.
However, they don't appear to power up when connected to my Negear FS108P,
which is an 802.3af Power-
After
running 'make install', do a 'depmod -a'.
Then
check /lib/modules for the file:
find
/lib/modules | grep zaptel
Be sure
the path /lib/modules//extra/zaptel.ko matches up with your
currently running kernel (from uname-a) as that is where it will be
checking.
From: [EMAIL PROTECT
Then the "make install" in the Zaptel directory didn't work or installed
it in the wrong location.
Julian Varanini wrote:
Hi Eric,
Tried that but I am still getting the same error.
Thanks
Julian
Date: Thu, 9 Nov 2006 17:25:02 -0600> From: [EMAIL PROTECTED]> To: asterisk-users@lists.d
Hi Eric,
Tried that but I am still getting the same error.
Thanks
Julian
> Date: Thu, 9 Nov 2006 17:25:02 -0600> From: [EMAIL PROTECTED]> To: asterisk-users@lists.digium.com> Subject: Re: [asterisk-users] Modprobe Zaptel> > Julian Varanini wrote:> > Hi,> > > > Can someone walk me through
Hi David,
Packet 8 are restricted to a single call
because they use an ATA which I then route into my asterisk server via a tdm400p
Cheers,
Dean
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dovid B
Sent: Thursday, 9 November 2006
7:35 PM
To: A
Are you trying to get FOP to monitor the SIP
account that you are using to dial the cell phone on ?
- Original Message -
From:
Alexander
Burke
To: asterisk-users@lists.digium.com
Sent: Wednesday, November 08, 2006 11:07
PM
Subject: [asterisk-users] Off-Site
I have the same issue. Just went in to my box.
Seems I am still registering with them. This is from when I tested them. The
call quality was horrible. If you want IAX specificly I would recomend
teliax.com. Call quality is great and during business hours some one actually
answers the phone.
Post away.
- Original Message -
From: "Jay Moore" <[EMAIL PROTECTED]>
To:
Sent: Thursday, November 09, 2006 6:58 PM
Subject: [asterisk-users] Quick Q...
Before I make any serious gaffes, is this an acceptable place to post
PHPAGI questions as well? I can't seem to find a dedicat
More than "1 incoming line" will depends on your
provider. I have a SIP provider that will send me up to 10 channels at a time. I
personally use myphonecompany.com. They have been really good for
me.
And now for the disclaimer:No I do not work for
them. Just a reall happy customer for orig
Sup pigs,
I think I found the solution to the problem!
It's a one line of code fix to rtp.c that people have been going daffy
about for almost a year! A patch was uploaded in August but for some
reason it hasn't found its way in to 1.2.
Here's the bug:
http://bugs.digium.com/view.php?id=5970
Hi Gustavo,
I correct myself. Voicemail is possible if you make a supervised
transfer (I was talking about blind transfer).
Sorry for my too fast response.
Jorge Mendoza
===
Hi Gustavo,
Auto attendant is easy, voicemail I don't think so (there are not
extension information
Hi Gustavo,
Auto attendant is easy, voicemail I don't think so (there are not
extension information when call is back to Asterisk).
We use the following topology:
- pstn line -> norstar (ext 123) -> (fxo) asterisk
Jorge Mendoza
Gustavo Berman wrote:
> Hi there!
>
> We have an old legacy norstar
Hi all,
Since i cant get latet beta of zaptel installed on the latest test version of
Debian with kernel 2.6.17-2-686 can someone who is using debian give me some
tips on how to get it working and installed?
Many thanks,
Christian
___
--Bandwidth and
I know that on my blog I have a flash
player which is just html generated from xml feeds.
http://deancollinsblog.blogspot.com/
Can a html web page be auto generated from
within the Asterisk voicemail module and be sent to an email?
What about auto generating a html email
with a “p
Julian Varanini wrote:
Hi,
Can someone walk me through compiling and loading the Zaptel 1.2.10 driver for Mandriva 2006 kernel 2.6.12-12? When I compile and attempt a modprobe I get "module zaptel not found"
You need to edit /usr/src/linux/Makefile to remove make the EXTRAVERSION
variable eq
Hello List,
So I have a few MiTel 5224 IP phones running in SIP mode. Per the
phones documentation they honor SIP distinctive ring tones. I am able
to send the correct ALERT_INFO message in an invite from Asterisk to
the phone, but I don't know what ring tone to call. From the reading
I've don
Felipe Amaral wrote:
Hi,
There's anyone here who go to "Estacao Voip" in Brazil???
http://www.estacaovoip.com.br/
I was think to go
Anyone here ??
--
Felipe Amaral
Vento Livre Internet
Felipe,
I will be there, and so will Mark :).
--
Kristian Kielhofner
__
Hi,
Can someone walk me through compiling and loading the Zaptel 1.2.10 driver for Mandriva 2006 kernel 2.6.12-12? When I compile and attempt a modprobe I get "module zaptel not found"
Thanks
Julian
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The Asterisk Development Team is pleased to announce the release of
version 1.2.11 of Zaptel.
This release includes a small number of fixes, primarily to support
recently updated hardware products from Digium. It also contains a very
large XPP driver update from Xorcom for their Zaptel-compatible
I still don't see a reason to use it. If you want immediate information
about phones even in the event of a catastrophic failure, bypass the
cache altogether (that's what we do) and have it do a lookup every time.
Also, set your lookup time to an acceptable value in the event that the
primary DUND
Can you try a sip.conf entry with the port= parameter as well.
For example:
[lucent]
type=friend
host=
port=5060
insecure=port,invite
context=default
Your INVITE header is including the port, and maybe Asterisk is having
trouble matching the sip.conf entry.
So the insecure=port,invite opt
Yes, same WEP key, SSID and channel for all the AP.On 11/9/06, Altus Snyman <[EMAIL PROTECTED]> wrote:
Everything is working beside roaming
Yes im using encryption, should I turn it
off, or uses the same wep key, and same ssid
Should I then also just add 1 config with
1 access poi
Justin Tunney wrote:
UPDATE:
I've installed ztdummy and have chan_zap.so loaded in to the system so
Asterisk can use psuedo zap channels or whatever it does for timing.
I also specified relaxdtmf=yes in sip.conf.
DTMF is still completely awful. Digits are still getting doubled up
and my IVR is
Does anyone know if it’s possible to port a number AWAY
from packet8?
I’ve been with them for 2 years and really want to
move to an IAX based service so I can have more than 1 incoming line at a time.
Cheers,
Dean
UPDATE:
I've installed ztdummy and have chan_zap.so loaded in to the system so
Asterisk can use psuedo zap channels or whatever it does for timing.
I also specified relaxdtmf=yes in sip.conf.
DTMF is still completely awful. Digits are still getting doubled up
and my IVR is still nearly impossib
This exact problem is solved by the short cache time, this is one of the reasons behind the low cache time in the white paper by JR.Your example is correct also, but your are expecting pbx2 to "push" the information to the server whereas JR has the central server pull the information. Both work, bu
Hi there!
I'm setting up a connection between Asterisk ver. 1.2.13 and a Harris
20-20 PBX. More less everything went fine, but the problem I have now
is that when dialing to the Harris PBX it seems to pick up my call as
soon as it reaches it.
For example if from the Asterisk outgoing folde
On Thu, 2006-11-09 at 14:05 +0200, yusuf wrote:
> Khaled wrote:
> > I installed libsrtp can any one help me how to ingrate it with asterisk
> > .to make SRTP
> >
> > Regards
> >
> Hi,
>
> I dont think SRTP is supported in Asterisk. There is some work to have RTP
> over TCP, where be
> def
Hi everybody,
I've got a very strange problem:
As far as I remember I didn't change anything on my Asterisk side. I
have 2 SIP providers to which I can place outbound calls.
Today I noticed that outbound calls to provider "inode" fail (and
inbound from this provider too). On the CLI I get every
I also just realised a distinct advantage of the precache model.
Lets say you have a central DUNDi cache server. He has in his cache the
knowledge that appearance 2944093 is registered to pbx1 for the next hour. If
pbx1 where to crash, then for the next hour, calls to 2944093 would fail.
Howev
Hi Vicky,
I used to use their termination services, but I had the same problems ... It is impossible to work with that latency ... A lot of gaps and dead spots ...
If you know ... I'm looking for a good termination provider that I can use the combination IAX/iLBC ... If you know some .. can yo
Everything is working beside roaming
Yes im using encryption, should I turn it
off, or uses the same wep key, and same ssid
Should I then also just add 1 config with
1 access point , not 2?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew Joakimsen
Sent: T
Bruce,
After
thinking about it a bit, I can see how setting the cache time to some value
higher than 0 could be effective. However, I'm trying to figure out what
benefits a 'central' DUNDi cache server provides over a completely distributed
architecture. If you have three asterisk boxes, a
Disable WDS but set all the AP to the same channel and same SSID and then make sure they are connected to the same LAN (IE: no NAT on the AP). Are you using encryption?Something like:
Try RxLevel -60PreRoaming Enable RxLevel -75Try over TxErrcnt 15Try Over RxError Count 10Play with the PreRoaming m
If you have a large number of servers, a mesh relationship may not scale well,
and maintaining relationships between all the servers is difficult for
starters. It also cuts down on the physical distance to perform queries if you
have a centralised DUNDi server, rather than having to query the pe
Could it be you did not bind it correctly in http.conf? Something similar
happened to me today while I was doing the same thing.
Try:
enabled=yes
enablestatic=yes
bindaddr=0.0.0.0
Hope this helps
l.
On Thu, 09 Nov 2006 19:28:17 +0100, Curt Shaffer <[EMAIL PROTECTED]>
wrote:
I was just g
Good day all
I cant get my WIP 5000 to roam 100%
I have 2 access points, different SSI’s
I make a config1 and config2 on the phone, each for the different
SSID’s(A & B)
Im standing next to A and I walk to B, but…the phone
does not want to change its signal to B, it still keeps the bad s
I was just going to test out the new Asterisk 1.4 GUI. I
downloaded it from source make;make install. I added my http.conf and modified
manager.conf. I restarted Asterisk and did a make checkconfig and it says
everything looks good. But I notice that the port 8088 is not listening when I
do
Why would you want to do that? Defeats the purpose of *having* the
DUNDi protocol. Why not just program the extensions in at regular
intervals or something?
On Thu, 2006-11-09 at 10:16 -0700, Douglas Garstang wrote:
> Aaron.
>
> Thanks. JR sent me that article before it was published. He's not
This is how I'm able to record my outbound calls, hope this helps you.
exten => _407NXX,1,Set(CALLFILENAME=${EXTEN:1}-${TIMESTAMP}-OUT)
exten => _407NXX,n,Monitor(wav,${CALLFILENAME},m)
exten => _407NXX,n,Dial(ZAP/g1/1${EXTEN:0})
exten => _407NXX,n,Congestion
Ed Nuñez
IT/Telecom
On 10:16, Thu 09 Nov 06, Douglas Garstang wrote:
> Aaron.
>
> Thanks. JR sent me that article before it was published. He's not precaching
> registrations. He's doing something different. In his configuration, when a
> registration server gets a request for the location of a phone, it queries
>
Doug,JR's example does cache but only for a very short time, like 5 seconds, so that if the device registers else where then the lookup is able to find it. You can change the cache time to the default hour or what ever you want. As far as precahe, I know it is a dundi cli command and you could prob
Hi,There's anyone here who go to "Estacao Voip" in Brazil???http://www.estacaovoip.com.br/I was think to goAnyone here ??
-- Felipe AmaralVento Livre Internet
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asterisk-users mailing list
Here's how to unsubscribe:
First, ask your Internet Provider to mail you an Unsubscribing Kit.
Then follow these directions.
The kit will most likely be the standard no-fault type. Depending on
requirements, System A and/or System B can be used. When operating
System A, depress lever and a plast
Aaron.
Thanks. JR sent me that article before it was published. He's not precaching
registrations. He's doing something different. In his configuration, when a
registration server gets a request for the location of a phone, it queries the
DUNDi Lookup server, which in turn queries the other reg
>I use alphanumeric names as extensions in my Asterisk architecture,
>which are the username part of the e-mail of each person at my site.
>Because Asterisk was primarily built to use numeric extensions, I'm
>having some problems with people that have usernames with dots between
>letters, like
At 05:00 AM 11/9/2006, you wrote:
Several motherboard manufactures in the last 3-4 years have had
capacitor problems, some reached the point of leaking others began
to cause problems on the machine after they began to swell. Both
Dell and IBM have replaced systems I know of and had the onsite
Hello list,
I have prepared a couple of new tutorials you may find interesting:
- Installing an Asterisk 1.4 beta system - at http://astrecipes.net/?n=216
- Installing the Digium's Asterisk GUI for 1.4 - at
http://astrecipes.net/?n=217
It's nothing too complex, but you may find them interest
Hi,
I am trying to decide which BRI card to buy and am looking at the B410P. I
have the following questions and hope that you guys can give me some advice.
1) Anyone uses the B410P with Asterisk with good or bad comment? And how
does the B410P compared to the others like the AVM or the Beronet.
yusuf wrote:
Khaled wrote:
I installed libsrtp can any one help me how to ingrate it with
asterisk .to make SRTP
Regards
Hi,
I dont think SRTP is supported in Asterisk. There is some work to have
RTP over TCP, where be default its over UDP.
SRTP has nothing to do with the tra
Before I make any serious gaffes, is this an acceptable place to post
PHPAGI questions as well? I can't seem to find a dedicated mailing list
for it. If not, any suggestions?
Thanks,
Jay
___
--Bandwidth and Colocation provided by Easynews.com --
as
Doug,
This may help you out a little. It's a whitepaper that JR wrote on how
to get a DUNDi cluster working with two redundant primary servers that
handle all the DUNDi legwork. Read through it, you might get some
information you can use out of it.
http://txaug.net/storage/users/3/3/images/17/U
Hi All,
I have tried everything to get callerid to work reliably but to no avail.
I have configured zapata.conf as per documentation but still only get 50%
of callerid's through. As a test I called our system with my mobile a
number of times and only 50% get through. I do get warnings about
pol
Vicky wrote:
Anyone having problems with voxee since last few days or is it just me
? In peek hours i get LAGGED when i do a iax2 show peers or even 1000
ms latency . Most of time it is 20 ms or so but when i start sending
traffic to them latency increases to 1000 ms or even LAGGED ( also
sho
Hi all,
I use alphanumeric names as extensions in my Asterisk architecture,
which are the username part of the e-mail of each person at my site.
Because Asterisk was primarily built to use numeric extensions, I'm
having some problems with people that have usernames with dots between
letters,
Hi there!We have an old legacy norstar phone system m8x24-ds ( dr5 ) and a couple of m0x16. It has 5 external analog lines. It has no auto attendant, and no voicemail. So every incoming call is forwarded to a operator, she pick up the phone, talks to the caller and transfer the call to the right ex
That clarifies it!
First the stupid questions to eliminate the possibility of anything besides
the phones,
Have you connected a different make hardphone or softphone and confirmed
that works?
Have you tried a different IAX/SIP provider?
-Original Message-
From: Matt [mailto:[EMAIL PROT
Adam Mattina
Networking & Systems
Support
Layer 8 Group, Inc.
585.442.
[EMAIL PROTECTED]
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://l
On 11/9/06, mail-lists <[EMAIL PROTECTED]> wrote:
Erick Perez wrote:
> I can report that with asterisk 1.2.13, internal SIP calls work
> perfectly but (in my particular case) my asterisk box cannot recognize
> DTMF digits when it receives a call via our SIP provider. we are both
> using rfc2833 a
Just to report back in, the advice of the list was to not worry about it- they should work well. I took a DSL modem with a router on it and connected both phones (Grandstream GXP2k and 101)- they did not work. I found that I had to program in a STUN server. I also has to set it to use a random p
Am Donnerstag, den 09.11.2006, 12:19 -0300 schrieb Frederico Madeira:
> I'm registering 5 lines on my asterisk box from one voip provider.
> Lines;
>
> 4040.
> 4040.0001
> 4040.0002
> 4040.0003
> 4040.0004
>
> All lines is registered in 5060 port so when someone call to 4040.0001
> the call a
Sounds like the Sony/Toshiba battery
issue.
Contract goes to assemble company, but they
outsource component manufacturing to the
lowest bidder with no quality control.
-- -- Steven
http://www.glimasoutheast.org
"Bruce Reeves" <[EMAIL PROTECTED]>
wrote in message news:[EMAIL PROT
Does anyone have any information on how to use DUNDi precaching?
Mark Spencer made a post 2 years ago where he hinted it may be possible to
configure DUNDi such that you could centralise your DUNDi registration info by
using precaching, instead of having each DUNDi peer meshed with every other
Anyone having problems with voxee since last few days or is it just me ? In peek hours i get LAGGED when i do a iax2 show peers or even 1000 ms latency . Most of time it is 20 ms or so but when i start sending traffic to them latency increases to 1000 ms or even LAGGED ( also shows high in peak ti
On 11/9/06, Steve Totaro <[EMAIL PROTECTED]> wrote:
Martin Joseph wrote:
> On 2006-11-08 14:40:09 -0800, "Ken Williams"
> <[EMAIL PROTECTED]> said:
>
>>
>>
>> This is a multi-part message in MIME format.
>>
>> After about one weeks time I've gone from no VoIP to a completely
>> configured system
On Thu, Nov 09, 2006 at 04:05:25PM +0100, Christian wrote:
> hi,
> OK, her it goes. here is what happens when typing make for the second time.
[snip]
But the error you posted before was from 'make install', so a successful
run of make does not indicate any change.
--
Tzafrir Cohe
I'm registering 5 lines on my asterisk box from one voip provider.Lines;4040.4040.00014040.00024040.00034040.0004All lines is registered in 5060 port so when someone call to 4040.0001
the call arrive on asterisk but arrive to last number registered 4040.0004 becouse it is listening on same por
Erick Perez wrote:
I can report that with asterisk 1.2.13, internal SIP calls work
perfectly but (in my particular case) my asterisk box cannot recognize
DTMF digits when it receives a call via our SIP provider. we are both
using rfc2833 and I have tried relaxdtmf=yes/no
when i use an internal s
hi,
OK, her it goes. here is what happens when typing make for the second time.
make[1]: Entering directory `/root/zaptel-1.4.0-beta2/menuselect'
checking build system type... i686-pc-linux-gnu
checking host system type... i686-pc-linux-gnu
checking for gcc... gcc
checking for C compiler default ou
Thanks Leonardo,After change that parameter resolve the problem.Thans a lot.-- Frederico Madeira[EMAIL PROTECTED]
www.madeira.eng.br
2006/11/9, Leonardo Gomes Figueira <[EMAIL PROTECTED]>:
Frederico,Frederico Madeira escreveu:> 1. When users dial 2 on phone (alcatel) they don't received a dial tone
I can report that with asterisk 1.2.13, internal SIP calls work
perfectly but (in my particular case) my asterisk box cannot recognize
DTMF digits when it receives a call via our SIP provider. we are both
using rfc2833 and I have tried relaxdtmf=yes/no
when i use an internal sip extension and cal
Also, I am not using a zaptel timer. Could this possibly be causing
problems with DTMF??
I really don't know for certain but here's what I experienced: When
calling out asterisk gives the option to allow called numbers to
transfer by hitting the '#' by putting 'T' (or 't'?) as an option in th
What codec are you currently using for voice?
I have found that when nothing else works, playing with the gains on the
Zap channel helped. Usually lowering them.
I use rfc2833 for dtmf, alaw as codec.
Yes, a lowering could be a idea, but the problem is logged on any kind
of channels in my
pls post it complete, i can't see there your channels for TE110P 30 voice channels...Also do this:[default]exten=> _X.,1,Answer()exten=> _X.,2,Noop(This is debug, i'm receive from Alcatel:${EXTEN})
exten=> _X.,3,Wait()exten=> _X.,4,Playback(vm-goodbye)exten=> _X.,5,Hangupexten=> h,1,hanguppls post
Frederico,
Frederico Madeira escreveu:
> 1. When users dial 2 on phone (alcatel) they don't received a dial tone,
> only receive a ocuped tone;
>
> 2. When users make step one, in asterisk console i received this message:
>
> !! Unexpected Channel selection 3
> -- Extension '' in context 'de
Follow bellow:[trunkgroups][channels]language=ukcontext=defaultswitchtype=euroisdnsignalling=pri_netrxwink=300 usecallerid=yeshidecallerid=nocallwaiting=yesusecallingpres=yes
callwaitingcallerid=yesthreewaycalling=yestransfer=yescanpark=yescancallforward=yescallreturn=yesechocancel=yesechocancelwhe
*bump*
No suggestions at-all? Does anyone use this facility in a similar way
and NOT have problems?
Thanks,
Steve
On 11/3/06, Steve Davies <[EMAIL PROTECTED]> wrote:
Hi,
I have started using the call recording facilities in Asterisk 1.2
recently, and having worked out some of the foibles rega
Several motherboard manufactures in the last 3-4 years have had capacitor problems, some reached the point of leaking others began to cause problems on the machine after they began to swell. Both Dell and IBM have replaced systems I know of and had the onsite techs check for swollen or leaking capa
Frederico,Pls Post your zapata.conf, any ways pls read bellow:On 11/9/06, Frederico Madeira <[EMAIL PROTECTED]
> wrote:Hi guys,I have an Alcatel 4200 with a ISDN board pluged in the asterisk server with TE110P.
Input callsVOIP Proider ---> Asterisk ---> Alcatel
Output CallsVOIP Proider <--- Asteris
Hi guys,I have an Alcatel 4200 with a ISDN board pluged in the asterisk server with TE110P.Input callsVOIP Proider ---> Asterisk ---> Alcatel
Output CallsVOIP Proider <--- Asterisk <--- AlcatelIn alcatel phones, users should dial 2 for take a line tone and can dial. At this point start my problems:
Khaled wrote:
I installed libsrtp can any one help me how to ingrate it with asterisk
.to make SRTP
Regards
Hi,
I dont think SRTP is supported in Asterisk. There is some work to have RTP over TCP, where be
default its over UDP.
--
thanks,
yusuf
--
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I installed libsrtp can any one help me how to ingrate it
with asterisk .to make SRTP
Regards
*
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Hello,
Sorry for returning such an old topic but it looks like I found a
solution. I am using FC5 on an IBM x206 with TDM2400P and TE405P.
Using this general guide:
http://www-128.ibm.com/developerworks/library/l-hw2.html
and this hint
http://pastebin.ca/32678
I had put pastebin.ca stuff into /
Martin Joseph wrote:
On 2006-11-08 14:40:09 -0800, "Ken Williams"
<[EMAIL PROTECTED]> said:
This is a multi-part message in MIME format.
After about one weeks time I've gone from no VoIP to a completely
configured system for two of our offices to be able to page/communicate
interoffice as w
Hi Jorge,
I would also like to Asterisk on a Sun
Server with Solaris 10 as the OS if you do get any information on this I would
appreciate it if you could share it with me.
Thanks,
Akash
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew Joakimsen
Sen
On Thu, Nov 09, 2006 at 12:25:10AM +0100, Christian wrote:
> Hi all,
> I have now reinstalled my whole system because I had to change a few things
> wiht my drives. Here is what happens. I have done apt-get build-dep asterisk
> apt-get install linux-headers-2.6.17-2-686 which works just fine now.
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