Non ho visto che c'era la nuova versione visto che la home dava come
stabile la RC8o
--
I havent seen new versione because junghanns's home report RC8o as
stable version
Dear trixter
Our software AstBill is now in use/beeing implemented by many smaal service providers and a few very large. It is Open Source.
I love to work with you on this and if any features are missing we be happy to implement it.
Are Casilla --
http://astartelecom.com - Independent VOIP
Can anyone please let me know if they have got UK CallerID working using a
X101P?
While you're waiting for a live answer, there are several threads on
this list you could search for.
Try this
http://www.google.com/search?hs=7Adhl=enq=asterisk+uk+calleridbtnG=Search
Dear Ralf
We have a few large installations that are using Asterisk and SER managed by our Open Source software AstBill.
It is working exelent. Basically Asterisk is handeling the PSTN and Voicemail part.
The authentication in Asterisk is done using ANI/CLI.
This setup is not very well
On Tue, Oct 25, 2005 at 06:01:55AM +0100, Paul Duffy wrote:
Hi All
Can anyone please let me know if they have got UK CallerID working using a
X101P?
If so please can you let me know which version of asterisk, did you apply
the UK CID patch, what are your settings in zapata.conf,
On Tue, 2005-10-25 at 08:21 +0100, Are wrote:
Dear trixter
Our software AstBill is now in use/beeing implemented by many smaal
service providers and a few very large. It is Open Source.
I love to work with you on this and if any features are missing we be
happy to implement it.
I didnt
On Tue, 2005-10-25 at 08:27 +0100, Are wrote:
The authentication in Asterisk is done using ANI/CLI.
Same way as broadvoice, wonder if using that setup if I set my caller id
to someone else will it cause the INVITE that broadvoice does
(broadvoice will invite the person registered as that
Hi,
I gave it a quick try (audio only):
- set Public SIPaddress, SIP registrar, SIP-proxy etc. to the IP of the
asterisk
- set DEFAULT_AUTHENTIC... to 'asterisk'
- removed all STUN entries etc.
- provided user name and pwd according to configured SIP friend
- SIP communicator registers with
On Thu, Oct 20, 2005 at 10:45:38AM +0200, Giordano Grandis wrote:
Hi all,
i'm looking for an utility that let me trace an ISDN trunk (or all ISDN
traffic) on HFC PCI card.
ZapHFC is zaptel. You can basically use all the tools availble for
Zaptel. What exactly do you want to trace?
--
Perhaps this question should be directed to Cisco support, but since
these guys made me nuts (please check that your cable is plugged in
correctly etc.), I thought I'd ask here.
We bought a Cisco 7905G phone, which boasts to have PoE (Power over
Ethernet) support.
We have a Netgear FS108P
The Norwegian Asterisk user's group is meeting on Tuesday next week. A
full one-day seminar in several tracks covering Asterisk is arranged in
Oslo.
See http://www.asterisk.no for the agenda.
I will attend the meeting and enjoy listening to people's experience of
Asterisk and various
The asterisk users group in Sacramento, California is going to have its
first meeting a week from friday, and I would be interested in talking
to anyone that is on this list that would think about going.
If you are in Sacramento please email me off list. I have a place
holder site for now for
Tomasz Chmielewski ha scritto:
We bought a Cisco 7905G phone, which boasts to have PoE (Power over
Ethernet) support.
the 7905 can be powered using pre-standard inline power.
So it /doesn't do 802.3af/
I searched the voip wiki - http://www.voip-info.org/wiki-Cisco+POE -
and found a
Hi,
Since we're doing this...
There is now a New Zealand Asterisk Users Group set up.
There is a wiki and mailing list at http://astug.org.nz both are sparse at the
moment and could do with some input.
If you're in New Zealand (or not) and interested in Asterisk then sign up and
get
Sergio Chersovani ha scritto:
No way to power up the phone is the the switch can be forced to send
power in any case.
I meant that the phone can power up with a custom poe injector that does
not care about 802.3af
Sergio
___
--Bandwidth and
Sergio Chersovani schrieb:
Sergio Chersovani ha scritto:
No way to power up the phone is the the switch can be forced to send
power in any case.
I meant that the phone can power up with a custom poe injector that does
not care about 802.3af
does poe injector = poe switch (is poe switch
Tomasz Chmielewski ha scritto:
if so, it means my switch is not dumb enough or what?
yes. And the cisco pre-standard poe has reverse pinouts.
I guess your switch does not send power because it doesn't see that the
cisco phone wants power.
I dunno if netgear can force the power injection.
Hi,
is there an Agentlogout procedure opposite of the one we get with Agentlogin ?
I tried simply having another agent log from the same extension, but when I try
Show agents
10 (Alessio) available at '[EMAIL PROTECTED]' (musiconhold is
'default')
51 (Giuliano) available
Hello Tomasz,
I got the 7905 working with an Dell POE switch without any
modifications of cables, the 7960 also works on the Dell switch but
you have to modify the cable.
I also tried the Netgear FS108p and it does not work with the 7905,
7912 and 7960 as I have tested. Even with modified cables
Hello,
Is it possible to connect 2 (SIP) phones via the dialplan. Sort of
like dragging 2 phones to each other in Flash operator panel.
The thing is I need an action in the dialplan that will connect 2
phones to each other as a reaction to an event without any
intervention from one of the 2
rulle mus schrieb:
Hello Tomasz,
I got the 7905 working with an Dell POE switch without any
modifications of cables, the 7960 also works on the Dell switch but
you have to modify the cable.
I also tried the Netgear FS108p and it does not work with the 7905,
7912 and 7960 as I have tested. Even
Hi there,
can anybody tell me how can i distinguish an national from an
international call. The CallerID on the channel doesn't have any leading
'0' or '00' so that it is possible that i cannot be sure what type of
call i have.
i have tried to include 'nationalprefix' and
Hi all,
I am trying to use iaxmodem
I defined iax extension (591) and started iaxmodem.
iaxmodem registers with asterisk (is on the same box)
when a fax calls this extension, i get
Registration completed successfully.
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW
Hi Alessio,
The opposite of logging in with Agentlogin is simply hanging up the phone!
:-)
If you use AgentCallBack, you can instead logoff explicitly.
You vcan also log off users manually from the console.
Hope this helps
l.
On Tue, 25 Oct 2005 12:00:07 +0200, Alessio Focardi
[EMAIL
On Tue, 2005-10-25 at 12:29 +0200, [EMAIL PROTECTED] wrote:
Hi all,
I am trying to use iaxmodem
Great. It appears to solve a real problem in a very cost effective
mannor and needs people playing with it to find any bugs that havent
been found yet.
Timestamp: 1ms SCall: 30757
Dell 3424P, has poe, Qos,and Vlan
Mus
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To UNSUBSCRIBE or update options
On Tue, 2005-10-25 at 00:52 -0400, Leif Madsen wrote:
Now, as someone has also pointed out, using quotes around the string
is probably better form as it should handle spaces and such.
In expressions only. Set() command is broken in this area (1.2beta and CVS HEAD). To clear,
Ok, I am sorry, I didn't understand the slinear codec option
now I added the slinear codec into the 591 definition
[591]
username=591
type=friend
secret=password
record_out=On-Demand
record_in=On-Demand
qualify=no
notransfer=yes
host=dynamic
disallow=all
context=from-internal
callerid=Fax Frame
i want to put sip peer registration command register =
in my database . anybody have any idea about it how to do this
fahd
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Asterisk-Users@lists.digium.com
On Tue, 2005-10-25 at 12:57 +0200, [EMAIL PROTECTED] wrote:
so it seems I am having trouble with device file, I think:
Setting device = '/dev/ttyIAX'
actually I don't have any '/dev/ttyIAX'
Who should have created it ? How can I create it ?
thanks in advance,Andrea
At least it
On Tue, 2005-10-25 at 04:11 -0700, trixter aka Bret McDanel wrote:
On Tue, 2005-10-25 at 12:57 +0200, [EMAIL PROTECTED] wrote:
so it seems I am having trouble with device file, I think:
Setting device = '/dev/ttyIAX'
actually I don't have any '/dev/ttyIAX'
Who should have created
Hello,
Does anybody know when I can find firmware for Swissvoice Vizufon
(CIP-5500)? Google isn't sayng anything...
I want to update firmware because when I call somebody, Asterisk says:
WARNING[22728]: chan_sip.c:4826 check_auth: Stale nonce received from
'sip:[EMAIL PROTECTED]'
sip.conf
Now I have the file
/dev # dir ttyIAX
lrwxrwxrwx1 root root 10 Oct 25 13:52 ttyIAX -
/dev/pts/1
The file is now created when I start iasxmodem, and deleted when I quit the
app.
the result is the same, it is not able to write to the device
I am root, iaxmodem is running as
On Tue, 2005-10-25 at 14:00 +0200, [EMAIL PROTECTED] wrote:
Now I have the file
/dev # dir ttyIAX
lrwxrwxrwx1 root root 10 Oct 25 13:52 ttyIAX -
/dev/pts/1
That should be about right, there are various differences in systems but
generally that should work for linux.
Has anyone had any experience of setting up their * server as a pppoe
server such that devices would link to the server running * using pppoe
and then do SIP over the PPP interface. I sounds simple and workable for
specific handsets / IAD's that support pppoe.
Stuart
You could check these links. I'm trying to do the sip peer registration
like this but I get some error about username / auth name mismatch.
I think I do something wrong in the MySQL table.
I hope it works for you and if it works I would like to hear it from you.
Good luck
asterisk01:/dev # dir /dev/pts/
total 93
drwxr-xr-x2 root root0 Oct 13 20:51 .
drwxr-xr-x 30 root root94840 Oct 25 13:52 ..
crw--w1 root tty 136, 0 Oct 17 13:17 0
crw--w1 root tty 136, 1 Oct 25 13:52 1
crw--w1 root
I am sorry, I didn't mention another problem.
I WAS NOT able to compile the spandsp lib shipped with iaxmodem.
./configure is OK
makeinstead returns
asterisk01:/usr/src/iaxmodem-0.0.5/lib/spandsp # make
Making all in src
make[1]: Entering directory
On Tue, 2005-10-25 at 14:29 +0200, [EMAIL PROTECTED] wrote:
Timestamp: 3ms SCall: 04700 DCall: 1 [192.168.1.10:4569]
Unable to pass the full buffer onto the device file. 2015 bytes of 2048
written.Unable to pass the full buffer onto the device file. -1 bytes of 1
written.Oct 25
Hello:
We are using Asterisk as a voicemail and media server. Call processing
is done by a different box running SER. I am experiencing a problem when
trying to implement call park on Asterisk. The call is transferred to
the parking
lot OK but parkandannounce wants to dial the calling party
Probably you are right
I installed hylafax and configured it to use iaxmodem, but I didn't start
it
Now I will research how to start hylafax, and I will try again
Andrea
trixter aka Bret
Hi,
Does anyone know what the legalness is of unblocking a blocked call?
For instance, when someone blocks their number it comes into our
system with the block flag (across PRI). It is then passed on to the
ATA as blocked. Is it legal for me to set the flag back to unblock
the call? (I
On Tue, 2005-10-25 at 14:36 +0200, [EMAIL PROTECTED] wrote:
I am sorry, I didn't mention another problem.
I WAS NOT able to compile the spandsp lib shipped with iaxmodem.
I am unsure what comes with iaxmodem, I do know that the iaxmodem
project has fixed some spandsp problems and those
Hi,
I.e. I want to be able to turn on and off the MWI light independant of any
voicemail function in asterisk.
Is this at all possible?
(We have Polycom IP300 phones, and the MWI light works fine with
voicemail and SendText()).
/Ola
___
--Bandwidth
Thank you. After some reboots and repeated testing, I've refined my
observations. The no-audio problem is gone (no explanation). Through
further experimentation I've been able to observe a few consistent
things about behavior in its current condition...
The main problem seems to be related
Stuart Hirst wrote:
Has anyone had any experience of setting up their * server as a pppoe
server such that devices would link to the server running * using
pppoe and then do SIP over the PPP interface. I sounds simple and
workable for specific handsets / IAD's that support pppoe.
Stuart
I realize that on my box was not installed vgetty and agetty
( i think that they are demanded from hylafax to get data from the ttyIAX
device)
I have added them, reconfigured re-make and re-installed hylafax and
restarted it.
The problem, now, is about egetty, which actually dos not exists.
On Mon, Oct 24, 2005 at 06:42:02PM -0200, [EMAIL PROTECTED] wrote:
Yes, for sure, it works. With TE110P from Digium using E1/ISDN/Pri
signalling.
By heart, I remember the following:
1. Configure Siemens E1 port as station and Asterisk as Pri_Net (or
Central Office).
2. At Siemens,
Hello!
As I
know, the "register" is a variable of [general] section in
sip.conf.
You
can't use it in database, ie you can't add new registers without
reload
the
asterisk..
I am
right?
Regards.
Jsalas.
-Mensaje original-De: tijmen van den brink
[mailto:[EMAIL
Benjamin Lawetz wrote:
Since I can't do that, what I've settled on is heartbeat + mon.
Heartbeat will monitor for a system level failure and switch to the backup
machine if neccesary; and mon will watch the asterisk (or any
other) service and restart it and/or alert me if it
Dear Juan
I think you are right. you can't add new registers without
reload
asterisk. and the register can't be put in the REALTIME database.
But there is an alternative to put the sip.conf file in in the
database. This is a bit different from the REALTIME engine. This is
just a database
I'll see what I can do about improving the error messages involved with
the write error, but that's not going to help your problem here.
Until the IAXmodem documentation says otherwise, you *must* install and
use the spandsp version that ships with IAXmodem. The only exception to
this of
Hello there,
For some reason, Festival() works fine when I call from PSTN (via an IAX
connection that I've got from Voice Pulse), but does not produce any
sound when I call from my X-Lite SIP phone. However, if I use text2wave
instead of Festival(), both my PSTN and my X-Lite connections
On Mon, October 24, 2005 9:00 am, Tom Rymes wrote:
I would like to be able to edit the pager notification e-mail. ...
Looks like this new feature is already checked into CVS.
--
Paul Dugas, Computer Engineer Dugas Enterprises, LLC
[EMAIL PROTECTED] phone: 404-932-1355 522 Black
Good Question.
We have tested it with any combination we can think about and it is
working safely. There is no way (we know about) that you can pass toll
free calls. :-)
Basically SER is configured to only accept clients that have the same
callerid as account numbers so SER refuse to pass the
Title: AudioCodes - TP260
Hi All,
Does anyone have any experience with using Asterisk with AudioCodes TP260 SIP board? If yes, please let me know if you have had any problems.
Regards,
Chard Johnston
___
--Bandwidth and Colocation sponsored by
Juan Salas wrote:
Hello!
As I know, the register is a variable of [general] section in sip.conf.
You can't use it in database, ie you can't add new registers without reload
the asterisk..
You can have a static config in a database, but you will still have to
reload.
/O
Thanks for the answer.
The problem went away starting faxgetty, I am sorry, I didn't carefully
read the README
Now I have another problem, which probably is exactly what Lee said, a
spandsp version error.
Now I am trying with the spandsp-0.0.3pre4 version.
Andrea
Got a test setup with CCM 4.1 and Asterisk running kind of successfully. Been trying out using * fro the VM system. I can make calls from the CCM side across to * and answer them using a copy of Xten Lite. If I allow a call to head to voicemail, I can't hear any prompts from the system. If I watch
Hi Pablo!
I understood your problem. It is related to Siemens PBX.
With this topology, Asterisk is acting as a PSTN Central Office (a Public
Central). What you asking is something like this:
Asterisk acting as Central Office - HiPath - Public Central Office
That is: the SIP devices connected
I am using a A104 Sangoma card. We are runningasterisk cvs head on our
production box.After wanpipe configuration I am receiving the below
mentioned error.
pri show span looks good as below.
pri show span 1
Primary D-channel: 24
Status: Provisioned, In Alarm, Down, Active
Switchtype: National
Hi all
First of all excuse me if i make such a big post, hope
also to write in the right place.
I need to connect my linux/asterisk (10.0.0.252) box
to a Nortel PBX (192.168.1.10) with h323
I'd like to allow some phones to register via sip to
asterisk and
with these to the Nortel PBX wich gives
Have you tried Asterisk 1.2beta1? does it work under that release?
We have been using an a104u with PRIs on 1.2b1 for about 6 weeks now
with no problems.
MATT---
On 10/25/05, Sharon [EMAIL PROTECTED] wrote:
I am using a A104 Sangoma card. We are runningasterisk cvs head on our
production
On 10/25/05, Sergey Okhapkin [EMAIL PROTECTED] wrote:
On Tue, 2005-10-25 at 00:52 -0400, Leif Madsen wrote:
Now, as someone has also pointed out, using quotes around the string
is probably better form as it should handle spaces and such.
In expressions only. Set() command is broken in this
Taking in the account poor Asterisk documentation, it's a bug. The bug
can be called as a feature, only when it is documented:-)
On Tue, 2005-10-25 at 10:47 -0400, Leif Madsen wrote:
Set(CALLERID(name)=)
will set the name part of callerid to guess what?-) Yes, to a string
containing 2
Have you tried contacting Sangoma technical support? They are likely the best equipped to support the card and the alarm you're receiving.
On 10/25/05, Sharon [EMAIL PROTECTED] wrote:
I am using a A104 Sangoma card. We are runningasterisk cvs head on ourproduction box.After wanpipe configuration
sangoma tech's cldn't help with that error.
Also it did work with asterisk 1.0.3 version wanpipe-beta9-2.3.3
now i'm using wanpipe-beta15-2.3.3 with asterisk cvs head
probably someone cldanswer this
when wanpipex.conf files are created do they depend on the number of
spans or number of cards.
For hylafax to answer a call, you need to use faxgetty.. Add this 2
lines to your /etc/inittab and run init q to force a reload:
IAX:2345:respawn:/usr/local/bin/iaxmodem ttyIAX
modem:2345:respawn:/usr/sbin/faxgetty ttyIAX
Change the paths according to your system.
Julian J. M.
On 10/25/05,
The number of spans. If you've got a quad card, you can actually configure 4 different wanpipe interfaces on that card.
On 10/25/05, Sharon [EMAIL PROTECTED] wrote:
sangoma tech's cldn't help with that error.Also it did work with asterisk 1.0.3 version wanpipe-beta9-2.3.3
now i'm using
On Tue, Oct 25, 2005 at 12:31:41PM -0200, [EMAIL PROTECTED] wrote:
Hi Pablo!
ok. i do all the changes but now i have this error
-- Channel 0/1, span 1 got hangup
Oct 25 11:46:40 WARNING[3639]: app_dial.c:416 wait_for_answer: Unable to
forward voice
-- Hungup 'Zap/1-1'
== No one is
I succesfully compiled the spandsp-0.0.3pre4 version, but nothing changed
no chance to compile the spandsp package boundled with iaxmodem
Andrea
[EMAIL PROTECTED]
thank you for your answer
I discovered that two lines, and now i removed the error in writing to the
device (the device was not freed by faxgetty)
now the problem is spandsp, If I call from a fax machine it rings 5-6 times
and then it goes away (remote hungap)
Unfortunately I am not able to
[EMAIL PROTECTED] wrote:
now the problem is spandsp, If I call from a fax machine it rings 5-6 times
and then it goes away (remote hungap)
What your HylaFAX modem config file saying for RingsBeforeAnswer?
Lee.
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--Bandwidth and Colocation
Hello all,
forgive me if this is a simple question, but does bridging a SIP channel and an IAX channel that use the same codec (say, alaw) involve transcoding? i'm trying to figure out what kind of hardware i'll need, and i'm going to be using SIP endpoints and IAX trunking to move the audio along
You could use a call file. This would be achieved like the following:
exten = s,n,System(bash_file SIP/110 112)
where bash_file is a script you make that drops a .call file into
asterisk's outgoing directory. bash_file could contain something like
this -- (from memory, research before you
Hi Pablo,
I really cannot forward the extension.conf due company rules. I am sorry.
However, you are in the right path. If you can dial Hi-Path's extensions
from Asterisk, you have 95% of the configuration done.
All you need to do is:
. enable on Hi-Path inter-trunk traffic. That is, traffic
Hi
I have just a quick question on the README for the chan-capi-cm-0.6
relating to ECT.
In the first example case -
i.e.
exten = s,1,capicommand(ect|${MYHOLDVAR})
how is the destination number specified ? Is it implied somewhere ?
snippet from README ..
ECT:
Explicit call
I have read that the digium t1 cards disable echo automatically if fax is
detected, but I am assuming that this is hardware EC.
I have 2 TE110P cards that, I believe, do not have HW EC.
So, if I am using SW EC, does the EC still get cancelled on a fax call?
If not, is there a way to control
Hi all
First of all excuse me if i make such a big post, hope
also to write in the right place.
I need to connect my linux/asterisk (10.0.0.252) box
to a Nortel PBX (192.168.1.10) with h323
I'd like to allow some phones to register via sip to
asterisk and
with these to the Nortel PBX wich gives
We use tp-260 boards for ss7/sigtran.. they seem to behave similarly to
mp2000 or tp1610 series boards which we have used with both mgcp and sip
protocols.. their stuff seems to work rather well .. at least for us but
YMMV.
Chard Johnston wrote:
Hi All,
Does anyone have any experience with
On Tuesday 25 October 2005 13:07, Steven wrote:
I have read that the digium t1 cards disable echo automatically if fax is
detected, but I am assuming that this is hardware EC.
You assume incorrectly. The zaptel software echo canceller also is disabled
upon fax tone detection. This is on all
Hi
Have you got SER up and running
If so then get asterisk up and running
Then make sure ser can route to asterisk , search in google for routing
to voicemail from ser, lots of people do that
Now the call will be in asterisk, you will need to allow ser to pass
calls, and vice versa ser needs
Hi Matt,
Thanks for the feedback.
Regards,
Chard.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Hess
Sent: Tuesday, October 25, 2005 1:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] AudioCodes -
Someone has any idea about this issue? Thank you very much.
Alexandre Leclerc a écrit :
Hi all,
as it is obivious at the bottom of this screen dump, when I'm recieving
a fax from PSTN in the PBX, it fails to send it to extension 100 which
is a fx_oks on my digium card. But I can call
On 10/25/05, Sergey Okhapkin [EMAIL PROTECTED] wrote:
Taking in the account poor Asterisk documentation, it's a bug. The bug
can be called as a feature, only when it is documented:-)
Poor Asterisk documentation? Ouch. Have you checked out
http://www.asteriskdocs.org? We've just recently
Hi to all,
i am trying to complie the openh323 for Asterisk. I have installed
everything needed but when i try to do a make to asterisk-oh323-0.7.3 i get
the following message:
variable `oh323_tech' has initializer but incomplete type
Any ideas?
Thank you,
Budoka.
Ok, I tried something slightly different.
I modified the existing the udp.monitor (or was it the tcp.monitor) of mon
and basically sending a sniffed SIP Registration packet which I send to
the asterisk server. If I don't receive an answer within a set time. The
monitor sends an error.
It tells
The comment below makes me wonder could ttyIAX be configured to answer
mgetty?
I have made the mgetty talk to ttyIAX however, as soon as a ring comes
into th eextension , mgetty shuts down... so I cannot keep the signal up.
I tried to use the pppd daemon directly with ttyIAX and it said that the
[EMAIL PROTECTED] wrote:
The comment below makes me wonder could ttyIAX be configured to answer
mgetty?
Although I've not tried mgetty with IAXmodem, the intent was to make
this possible (for faxing), yes.
I have made the mgetty talk to ttyIAX however, as soon as a ring comes
into th
Hi All:
I have special set up to be done. See anyone can help me some ideas.
Two Asterisk servers, server A trunks to PSTN, server B works as call
routing engine.
All sip phones are registered in server B.
I have scenario like following:
1. A call comes to server A, server A sends the call
mgetty dump:
10/25 11:23:17 IAX tio_get_rs232_lines: TIOCMGET failed: Invalid argument
10/25 11:23:17 # data dev=ttyIAX, pid=15158, caller='none',
conn='DIRECT', name='', cmd='/bin/login', user='Fedora Core release 3
(Heidelberg)'
--
10/25 11:23:36 IAX mgetty: experimental test release
I have downloaded SJPhone - and well.. it does connect to my system,
however popping audio is heard when i dial my music on hold extension...
the quality is really really bad..
i have a WLAN-SDIO utility, the signal strength is at 11mb/s. however. is
that sufficient? The codecs for sjphone are
On Tue, 2005-10-25 at 11:15 -0700, Tielin Xu wrote:
How does server B receive the message from server A?
Many thanks for your help.
Nintendo eh? The Redmond office? Thats near where I live.
So let me make sure I understand the problem. Server A needs to get
information from Server B about
Tielin Xu wrote:
Hi All:
I have special set up to be done. See anyone can help me some ideas.
Two Asterisk servers, server A trunks to PSTN, server B works as call
routing engine.
All sip phones are registered in server B.
I have scenario like following:
1. A call comes to server A, server A
A SIP registration as a monitor is not a bad idea at all. The
registration process is not too terribly complex, and I think I could
write a perl script that could attempt registration when supplied with a
host, username, and password.
No promises, but if I can put something together I'll
Great.
All of the references I read mentioned the card specifically, not zaptel or
asterisk.
Thanks for the info.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past.
---- --- - - - -- - - -- - - - --- -
As a secondary point, I'm looking at buying a Imate Jas Jar running
windows mobile 5.0 to replace my treo.
Does anyone have any thoughts on Windows mobile 5.0 specific softphones.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On
Well what I was thinking of doing in the future was to have a cron job drop
a call file that would call another asterisk server that would auto-answer
and either generate some kind of network answer to MON or connect another
call to the first asterisk. Allows you to test your PRIs at a certain
On Tue, October 25, 2005 20:27, [EMAIL PROTECTED] said:
I have downloaded SJPhone - and well.. it does connect to my system,
however popping audio is heard when i dial my music on hold extension...
the quality is really really bad..
i have a WLAN-SDIO utility, the signal strength is at
On Tue, October 25, 2005 20:43, Dean Collins said:
As a secondary point, I'm looking at buying a Imate Jas Jar running
windows mobile 5.0 to replace my treo.
Does anyone have any thoughts on Windows mobile 5.0 specific softphones.
Cheers,
Dean
LOL! If you wait a bit longer you can buy a
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