Hi everybody,
can anybody explain one thing: say we have 2 SIP phones(or H323) and one
Asterisk Box on one local network. The phone1 calls phone 2 via Asterisk and
phon3 answers: is the real conversation streaming thru the * box, or it's
going straigth from one phone to the other?
Regards and
Hello Mauro,
Tuesday, December 27, 2005, 9:26:54 AM, you wrote:
MZ Hi everybody,
MZ can anybody explain one thing: say we have 2 SIP phones(or H323) and one
MZ Asterisk Box on one local network. The phone1 calls phone 2 via Asterisk and
MZ phon3 answers: is the real conversation streaming thru
On Tue, December 27, 2005 9:26, Mauro Zanin said:
Hi everybody,
can anybody explain one thing: say we have 2 SIP phones(or H323) and one
Asterisk Box on one local network. The phone1 calls phone 2 via Asterisk
and
phon3 answers: is the real conversation streaming thru the * box, or it's
Hello Robert,
I have this following setting on my WRT54GS:
# RTP ports
iptables -t nat -A PREROUTING -i $WAN -m udp -p udp --dport 1:2 -j DNAT
--to-destination $ASTERISK_IP
iptables -A FORWARD -i $WAN -o $DMZ -m udp -p udp --dport 1:2 -d
$ASTERISK_IP -j ACCEPT
# IAX port
Hello
I have a Vood vrg 121 (mgcp adapter) that I'm trying to register to my Asterisk but it doesnt work at all.
I have no earlier experience using mgcp devices I have just been using sip phones so dont be to hard ;-)
1) Do I have to do anything special to activate mgcp functionality in * or is
Elene Kinsky wrote:
We have 2 GXP-2000 dead during automatic firmware upgrade. Devices now
send out only one ARP packet for default gateway resolution during
boot and nothing more!
We've contact Grandstream support, but they cannot help. Now we want
to send devices to Grandstream for repair
On 12/26/05 08:28 Andrew Kohlsmith said the following:
There are two problems with this: 1. the A104 can have each span's sync
independent of the others, unlike the Digium cards. 2. With both spans
trying to sync to each other you can run into interesting clock situations
you may want to
Hello,
I am trying to develop a simple but fast application/daemon to take SIP
invites, convert them into ENUM queries, send those queries to an ENUM server
(likely residing on the same hardware as the daemon), get back an ENUM response
and convert that to a SIP 302 (or other 300 level)
It helped, a lot!
Thank you
Dov
- Original Message -
From: Faris Raouf [EMAIL PROTECTED]
To: Dov Bigio [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-CommercialDiscussion asterisk-users@lists.digium.com
Sent: Saturday, December 24, 2005 4:17 PM
Subject: Re: [Asterisk-Users]
Hello all,
Is it possible to change what filename automon (*1) files get, and if so,
how?
I checked the wiki, but only found info about filenames for normal
monitoring. Does the same work for automon?
TIA!
--
F Peeters
PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch
2 Sweex
Steve Underwood wrote:
We have 2 GXP-2000 dead during automatic
firmware upgrade. Devices now send out only one ARP packet for default
gateway resolution during boot and nothing more!
We've contact Grandstream support, but they cannot help. Now we want to
send devices to Grandstream
On 12/27/05, Francesco Peeters (Asterisk) [EMAIL PROTECTED] wrote:
Hello all,
Is it possible to change what filename automon (*1) files get, and if so,
how?
I checked the wiki, but only found info about filenames for normal
monitoring. Does the same work for automon?
Not really. The only
On Tuesday 27 December 2005 05:25, Dinesh Nair wrote:
what would the equivalent be for the digium cards ? would something like
the following work ?
span=1,0,0
span=2,1,0
span=3,2,0
span=4,0,0
(note that span's 1 and 4 are set as PRI NET)
What is each span connected to? Remember that
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I have 1.0.67 and for some weird reason the phones froze ... not sure
yet why.
NTP server (clock on your phone) does it work for you? I have working
NTP server, but phone allways shows wrong time.
--
Tomislav Parcina
[EMAIL
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] says...
Hi Tomislav,
If you want to do recording and are worried about high processor load
when keeping asterisk in the media path with SIP, you might check out
http://www.oreka.org which is an open source voip recorder that can
run on a
I can't find how to force an asterisk server to stay in the middle
between two asterisk clients, the iax2 reinvite pulls the call out of
the cdr, which is no good ...
suppose A calls B for 10 minutes
clientA --- server ---clientB
in the server cdr I see an A-B call of some seconds
and if I
On Tue, December 27, 2005 13:49, BJ Weschke said:
On 12/27/05, Francesco Peeters (Asterisk) [EMAIL PROTECTED]
wrote:
Hello all,
Is it possible to change what filename automon (*1) files get, and if
so,
how?
I checked the wiki, but only found info about filenames for normal
monitoring.
On 14:03, Tue 27 Dec 05, Tomislav Parcina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I have 1.0.67 and for some weird reason the phones froze ... not sure
yet why.
NTP server (clock on your phone) does it work for you? I have working
NTP server, but phone allways
BJ Weschke wrote:
Maybe one of the Digium folks can confirm, but no, I don't think it's
possible to upgrade the firmware on a TDM400P. I think you'd need to
exchange the card with Digium for a later version.
The only Digium boards with field-upgradeable firmware are the TE4XXP
and TE2XXP
;extensions for dan and adam
;dan - since people already know dan as extension 3, we keep
that for compatibility
exten = 3,1,GoTo(Pleximenu|103|1)
exten = 103,1,GoTo(default|103|1)
;adam
exten = 104,1,GoTo(default|104|1)
The bottom of the dialplan
I have 13 Polycom IP301's where the clock keeps resetting to a +5
offset. I can change the config file to show -5, change it to -5 on the
phone and after an hour or so the phone will update itself back to +5.
Anyone have any ideas? The other 70+ phones are not exhibiting this
behavior.
BJ Weschke wrote:
Maybe one of the Digium folks can confirm, but no, I don't think
it's
possible to upgrade the firmware on a TDM400P. I think you'd need to
exchange the card with Digium for a later version.
The only Digium boards with field-upgradeable firmware are the TE4XXP
and
On 12/27/05, Adam Moffett [EMAIL PROTECTED] wrote:
;extensions for dan and adam
;dan - since people already know dan as extension 3, we keep
that for compatibility
exten = 3,1,GoTo(Pleximenu|103|1)
exten = 103,1,GoTo(default|103|1)
;adam
exten
Hi all,
Im curious if anyone has tried installing Asterisk on
a Virtual Private Server from a web hosting company? I am a web hosting
reseller with Jodohost.com, so I can have as many Linux VPSs as I want,
and I thought I might try it. Im just curious if anyone else has
tried this
I can't find how to force an asterisk server to stay in the middle
between two asterisk clients, the iax2 reinvite pulls the call out of
the cdr, which is no good ...
The trick is to use some Dial options that forces * to stay in the
path, like t,T,h,H,w or W
See
On Saturday 24 December 2005 14:42, Zeeshan wrote:
Hi,
When I call to my asterisk server, voice prompts play ok but when it
goes to music on hold, sound is all broken. Why is that, is there some
ports which Music on hold uses which are not configured properly, or
there is some other reason.
Hi all,
I have just installed Asterisk 1.2.1 on my server and I'm having a problem with the X100 Zap channel.
The channel works for a while when I boot up the server and then degrades to an garbled dial tone and speach.
Also this problem will appear when I reload the Asterisk config files. My
Hi,
I am using Asterisk 1.2.1 and followed instructions
on http://www.asteriskguru.com/tutorials/spandsp.htmlto
install faxing capability on my server.
I get the following error messages...
Asterisk Dynamic Loader Starting: ==
Parsing '/etc/asterisk/modules.conf': Found[app_rxfax.so]Dec
Simone Cittadini ha scritto:
I can't find how to force an asterisk server to stay in the middle
between two asterisk clients, the iax2 reinvite pulls the call out
of the cdr, which is no good ...
suppose A calls B for 10 minutes
clientA --- server ---clientB
in the server cdr I see an A-B
Depends if you have reinvite on or off. On yes off no. At least that
is what I have read...you can verify with a network sniff on the
Asterisk server...use
tcpdump -ln host ip.off.asterisk.server and not tcp port ssh (telnet or
whatever protocol you are connecting to the astersisk server
On Tuesday 27 December 2005 09:18, Time Bandit wrote:
The trick is to use some Dial options that forces * to stay in the
path, like t,T,h,H,w or W
See http://www.voip-info.org/wiki-Asterisk+cmd+Dial for an explanation
of those options.
Why not just put 'notransfer=yes' in the appropriate
Steve Totaro wrote:
Field-upgradeable? Does that mean that I can do it myself? That would
be great since some systems are in production and sending the board to
Digium takes time.
The 2nd gen firmware has field-upgradeability. The 1st gen firmware does
not, unfortunately. There is not
Hi list..
Using asterisk 1.0.10 and the cdr_mysql addon to write CDR records to a
MySQL table. That part works great. The issue is that I also need the
Master.csv text CDR log and thusly have the cdr_csv.so module loaded.
The problem is, after 10-15 mins of activity, it just.. stops writing.
Time Bandit wrote:
The trick is to use some Dial options that forces * to stay in the
path, like t,T,h,H,w or W
See http://www.voip-info.org/wiki-Asterisk+cmd+Dial for an explanation
of those options.
Or set 'notransfer=yes' for at least one of the IAX2 peers/users involved.
In article [EMAIL PROTECTED]
aachen.de, [EMAIL PROTECTED] says...
It is not only re-invite that determines what happens to your media path,
there are also Dial() arguments like t,T,w,W (and possibly some more)
that can force it go through Asterisk. The same applies to codec
settings, i.e.
Why not just put 'notransfer=yes' in the appropriate iax.conf user/peer entry?
Oups, answered too fast. That is what happens when I try to answer a
technical question before finishing my first coffee.
Thanks for the correction
___
--Bandwidth and
On 12/27/05, Dov Bigio [EMAIL PROTECTED] wrote:
Hi,
I am using Asterisk 1.2.1 and followed instructions on
http://www.asteriskguru.com/tutorials/spandsp.html to
install faxing capability on my server.
I get the following error messages...
Asterisk Dynamic Loader Starting:
== Parsing
We've contact Grandstream support, but they cannot help. Now
we want to send devices to Grandstream for repair but they on
longer reply mail!
This is where a good reseller is worth their weight in gold. Unless you're
buying massive quantities of the things (in which case a failure of 2 is
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
My understanding is that canreinvite only redirects the media path.
Signaling and media are separate with SIP (which is what makes it so
nice by the way).
Yes, and dtmf can be sent with the sound. And if that is the case, then
media
Dov Bigio wrote:
I am using Asterisk 1.2.1 and followed instructions on
http://www.asteriskguru.com/tutorials/spandsp.html to install faxing
capability on my server.
what platform are you running on? (wich distro?)
Does the make of the app_txfax and app_rxfax work out well?
no podria decirte, porqe tengo problemas con los scripts de email2fax y
Asterfax...espero resolverlos pronto y verificar el correcto envio de
faxes...On 12/21/05, Jorge Cisneros [EMAIL PROTECTED] wrote:
gracias Felix por el tip, ya lo hice y si funciono todo bien. tengo
otro problema no puedo
On 12/21/05, Joao Correia [EMAIL PROTECTED] wrote:
Hello,
Making calls works fine on a Beronet 1 port card connected to an ISDN line PTP.
I cant seam to receive any calls. Asterisk says it cannot match
extension. The funny is that I tested this configuration on a ptmp and
it worked.
Any tips ?
I'm trying to set up call transfer and automon options. They work fine
with ZAP lines (analog telephone) and with Grandstream Budgetone 102. I
have problem with Cisco 7905 and 7940. I think that problem is with dtmf
signalization.
This is my configuration in 7940
dtmf_inband: 1
Tyler,
Can you upgrade to 1.2???
Alex
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tyler
Sent: Tuesday, December 27, 2005 9:40 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] CDR_CSV stops writing, help!
Hi list..
;extensions for dan and adam
;dan - since people already know dan as extension 3, we keep
that for compatibility
exten = 3,1,GoTo(Pleximenu|103|1)
exten = 103,1,GoTo(default|103|1)
;adam
exten = 104,1,GoTo(default|104|1)
The
I'm trying to look for documentation on how the TDD/TTY interfaces with
the user. From the looks of it, fskmodem talks directly to a channel.
Does it matter what type of channel it connects to? SIP/IAX/Zap?
Secondly, how does one interface with it on the asterisk side?
Obviously there is no
How to make one touch record on asterisk 1.2.1 use mixmonitor app ?
In res_features.c line line 469:
monitor_app = pbx_findapp(Monitor)
How to make pbx_findapp return mixmonitor ?
T
___
--Bandwidth and Colocation provided by Easynews.com --
Hi Jason. It seems your doing things right whatever that means. I
think the problem is more hardware related. Sure you have line in the
FXO?? have you tried dialing directly from some IP Phone?? I have
several applications that relay on automatic call generation with
Asterisk Manager and a PHP
Many of you current
cid_rewrite (v1.0.0) users probably noticed that your 411lookup is broken,
thanks to another change by the 411.com folks. So we fixed it :) The
latest changes include:
1. Adapt to new
411.com format
2. Improved address
conversion and extraction from reverse lookup
On 12/27/05, Vikas [EMAIL PROTECTED] wrote:
How to make one touch record on asterisk 1.2.1 use mixmonitor app ?
In res_features.c line line 469:
monitor_app = pbx_findapp(Monitor)
How to make pbx_findapp return mixmonitor ?
You'd need to make a few more changes to res_features.c than just
Hi BJ, Kristof,
It worked!
I am using the version at
http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.2pre21c/asterisk-1.2.x/.
I think I had bad symlinks on /usr/local/lib and by reading the tutorial on
AsteriskGuru I found that... (The previously installed version of spandsp
has been
When I reload IAX, I get the following messages on the console:
asterisk_test*CLI iax2 reload
== Parsing '/etc/asterisk/iax.conf': Found
Dec 27 16:56:28 NOTICE[23015]: chan_iax2.c:8618 set_config: Ignoring
bindport on reload
Dec 27 16:56:28 NOTICE[23015]: chan_iax2.c:8658 set_config: Ignoring
And buy your phones from a reputable dealer who will provide you with
support. Grandstream's policy (and sipura, snom, polycom, etc.) is to
provide warrantee service through their resellers. We have never had them
reject a properly documented RMA. -Mike
Michael Crown
Managing Partner
Script Head wrote:
As this isn't a part of *, has anyone accompilished a whisper mode in
yet? What I am looking for is an ability for to say something while
monitoring a channel and the agent being able to hear what I say while
the called party is not.
ScriptHead
Add it
to/etc/ld.so.conf
-Original Message-From: Kanishka Somaratne
[mailto:[EMAIL PROTECTED]Sent: Monday, December 26, 2005 9:51
PMTo: asterisk-users@lists.digium.comSubject:
[Asterisk-Users] LD_LIBRARY_PATH
HiI set the LD_LIBRARY_PATH and when i reboot i have to set it
Hi all, I have rolled out a few Zultys ZIP2 phones, and they seem to work
fine, except when trying to check voicemail. If we go into comedian mail, we
are prompted for a extension #, then a password. The ext # transmits
properly, but the password is not being heard by asterisk. The CLI output
says
Hi, i have a cisco 7912G with SIP firmware, its connect to the asterisk
through nat. The only problems is in the voice mailasterisk not
detect the tones, therefore i cant access to my voice mail extension.
Thanks in advance.
Diego.
___
It seems that Asterisk gives priority to extensions in the extensions.conf file
over what's access in the db via the switch statement. For example, if you have
an entry in extensions.conf and realtime for the same extension, Asterisk won't
look in the db.
Anyone know if there's a way to
Dan Elder wrote:
Hi all, I have rolled out a few Zultys ZIP2 phones, and they seem to work fine,
except when trying to check voicemail. If we go into comedian mail, we are
prompted for a extension #, then a password. The ext # transmits properly, but
the password is not being heard by
On 12/27/05, Douglas Garstang [EMAIL PROTECTED] wrote:
It seems that Asterisk gives priority to extensions in the extensions.conf
file over what's access in the db via the switch statement. For example, if
you have an entry in extensions.conf and realtime for the same extension,
Asterisk
By config file do you by chance happen to mean dhcpd.conf? I'm pretty
sure there are settings like this in ipmid.cfg or sip.cfg, but it could
be that the dhcpd.conf one is conflicting. In mine I have
option time-offset -32400;
Again, I'm not totally sure there is a time offset in the phone
Thanks, but static isn't an option. Users will have the ability to make changes
to their dialplan via a web portal. Doing a 'reload' every few seconds/minutes
is even less viable especially when you consider that a reload deletes all the
SIP subscriptions.
-Original Message-
From: BJ
I acquired a Blackberry 7100T over Christmas. I had heard it will work
with * and that is what I want to do with it. But I think it needs a SIM
card to make it work. If this is true how do I go about getting a SIM
card for it and how to set it up? Thanks for any help you can offer.
Bob
Douglas Garstang wrote:
It seems that Asterisk gives priority to extensions in the extensions.conf file over what's access in the db via the switch statement. For example, if you have an entry in extensions.conf and realtime for the same extension, Asterisk won't look in the db.
This is true
To get a sim card you need service from T-Mobile. Any cell phone, land line,
sat phone, etc will work with Asterisk depending on what you mean work with
*. You can set up an phone number as a custom extension. If you mean you
want to setup the blackberry as a sip phone that communicates with your
Victor Alvarez schrieb:
Hi,
I'm afraid I don't know how to use the command Transfer.
I am also interested how the command Transfer should be used.
I am aware of the possibility to add the option t or T to dial, so #33
transfers the call to extension 33.
Is there any use of this command
On Fri, 23 Dec 2005, stéphane plichon wrote:
Armin Schindler wrote:
Please create a verbose log of level 5 with 'capi debug'...
Armin
debug for incoming call:
...
CAPI INFO 0x3302: Protocol error layer 2
...
CAPI INFO 0x3302: Protocol error layer 2
==
when my Cisco IAD send a call to my Asterisk gateway the gateway treats
it as if I don't have a peer statement in sip.conf, when I do. Here are
the first two packets, notice the Found no matching peer or user for
'192.168.7.250:50437' on the second packet. Any one seen this before,
or have a
Take a look at the following page (you might be able to change the priority):
http://www.voip-info.org/wiki-Asterisk+config+extensions.conf+sorting
On 12/27/05, Douglas Garstang [EMAIL PROTECTED] wrote:
It seems that Asterisk gives priority to extensions in the extensions.conf
file over what's
On Sun, 25 Dec 2005, Michael J. Tubby G8TIC wrote:
On Sat, 24 Dec 2005, Michael J. Tubby G8TIC wrote:
I changed the dial-string to include flags 'ob' as you mentioned
(below)
and now I get the following when I dial a BT phone number
- dial number, get:
Proceeding (in 100)
Hello!
This is actually less a question than some information, if anyone else
struggles with the same issue.
I am located in the UK and use a Sipura-3000 adapter to connect to a BT
line (via fxo port). One problem I had was that disconnect supervision
didn't work:
Some caller phones me
I use:
# Enable_VAD (1-enabled, 0-disabled)
enable_vad: 0
dtmf_inband: 1
dtmf_outofband: never
dtmf_avt_payload: 101
and it works well for me. Sometimes going through a callmanager I have
to set outofband to avt to get dialtone sent though.
On Tue, 2005-12-27 at 16:05 +0100, Tomislav Parcina
On Saturday 24 December 2005 16:40, Kevin P. Fleming wrote:
Interestingly, some systems I manage also began exhibiting this behavior
in the past ten days or so. I have been working with the telco and they
too show the Calling Number being received as expected over the PRI, but
yet the far end
Hi Nahid!
Why do you want to do this? What about the ENUM resolvers inside
asterisk? There is the old EnumLookup application, and the new the
ENUMLOOKUP function (with plenty of features). Have you tried them?
If you do not want to use asterisk's internal ENUM resolvers, you could
also use
I wish I could on this box. It is in the plans, but we can't do that
just yet. Is this a known issue with the cdr_csv module in the 1.0
branch ??
tf.
On Tue, 2005-12-27 at 10:25, Alexander Lopez wrote:
Tyler,
Can you upgrade to 1.2???
Alex
-Original Message-
From:
Doug,
You might also check out the
wiki. There is a great deal of information regarding the connection of
Asterisk to legacy systems. I wrote the one on connecting
to an NEC NEAX 2400. Heres the main wiki page:
http://www.voip-info.org/wiki/index.php?page=Asterisk
Heres the page for
You do not need the BES server.. It is nice for total wireless
syncronization but not need for it to work.
The BB will work in three ways:
BES server, Married with Exchange server or Lotus notes.
Internet only, you are given an address like
[EMAIL PROTECTED], you them forward your emails to
I have a polycom 501, for some reason asterisk always shows
the round trip time to it as being significantly higher than the 2
grandstreams, all 3 are on the same lan.
Grandstream 40/40 192.168.16.40 D
255.255.255.255 5060 OK (4 ms)
Grandstream 31/31 192.168.16.31 D
255.255.255.255
Hello folks !
TDM2400 with E for echocan module is ok for me, replacing my old passive
cards.
No more echo issues now. I had many before to switch to this wonderfull card
!
Perfect for my use...
Here is an Asterisk SVN-branch-1.2-r7608M, in an old PII-400 MHz
Linux version 2.6.12-1-686 (gcc
Yes, I tied something like this. I included the database context first (the one
that has the realtime switch) followed by the context that has the extensions
locally. I shut the database down and Asterisk returns fast busy when dialling
the number. Doesn't appear to work.
[OffNet]
#include
Actually, who says this is supposed to work anyways? When Asterisk fails to
connect to the database when querying a number, does it have the logic to then
fail over and try the same number in contexts that follow? If it doesn't, then
there's no point.
-Original Message-
From: Douglas
If you do have the Blackberry Enterprise Server, there are options available
to send out a policy to the devices that contains SIP server and account
information. I have seen no other way to access those settings nor do I have
any clue how they would function if I tried to set it to use my
Hello,
Has any one been able to recveive a call from asterisk to msn ( not windows messenger by registering on asterisk) but on regular as hotmail id.
Please contact me even if there is a charge for it.
Rehan
-- Rehan Ahmed AllahWalahttp://www.SuperTec.com - Tommrow's Technology,
I think I found what is munging up the peer lookup:
This call from another Asterisk box starts:
-- SIP read from 192.168.69.254:5060:
The peer lookup that fail reads:
-- SIP read from 192.168.7.250:52141:
Asterisk seem to be thrown off by the fact that the return port is not
5060, and fails
I'm looking for a ay
to track when an agent logs inand logs out. Best if it could be put in a
mysql db but a text file will be ok for now..
Any help would
be great !
Thanks
___
--Bandwidth and Colocation provided by Easynews.com --
In short, does Asterisk have any database redundancy???
Is there any way to specific more than one db host in res_mysql.conf? If you
specify dbhost with a hostname, and use round-robin dns, does Asterisk read
this file only on startup or on every db connect attempt? If it fails to get a
I connect to Asterisk via SSH all the times. Did not notice about console
messages about module loading.
Thanks
- Original Message -
From: BJ Weschke [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, December
Hello
Can some one point me to more info on how to register a SIP PEER or a user on asterisk, say a FWD account on Real time database.
Thank You
Rehan-- Rehan Ahmed AllahWalahttp://www.SuperTec.com - Tommrow's Technology, Today.http://www.didx.net - DID Number Exchange and Peering Service.
Hi everyone,
I just upgraded my Asterisk box from 1.0 to 1.2, immediately after the
upgrade my Polycom Soundpoint 501 stop working. All outbound calls from
my phone show NO ANSWER in the CDR, the call connects but disconnects
after 60 second. Inbound calls to this phone work perfectly and all of
On Tue, 2005-12-27 at 19:27 +0100, Armin Schindler wrote:
It looks like the call is signaled on both ports !?
On another installation in France I'm also getting this, but with 2
Fritz! cards, the call is signalled on both cards.
--
Dave Cotton [EMAIL PROTECTED]
Hello,
can anybody tell me, if it is possible to play a soundfile to a caller
BEFORE having picked up? Will the call be billed for the caller on PSTN?
Best regards,
Arik
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users
How would you play a file to a line that hasn't been picked up? You have to
pick up the line in order to do anything with it.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Arik Funke
Sent: Tuesday, December 27, 2005 1:33 PM
To:
On Tue, 27 Dec 2005, Dave Cotton wrote:
On Tue, 2005-12-27 at 19:27 +0100, Armin Schindler wrote:
It looks like the call is signaled on both ports !?
On another installation in France I'm also getting this, but with 2
Fritz! cards, the call is signalled on both cards.
Is this some
On 12/27/05, Ross C [EMAIL PROTECTED] wrote:
I'm curious if anyone has tried installing Asterisk on a Virtual Private
Server from a web hosting company? I am a web hosting reseller with
Jodohost.com, so I can have as many Linux VPS's as I want, and I thought I
might try it. I'm just curious
If you have a PRI you can use app_playback with the NOANSWER option,
check show application playback in the CLI
On 12/27/05, Arik Funke [EMAIL PROTECTED] wrote:
Hello,
can anybody tell me, if it is possible to play a soundfile to a caller
BEFORE having picked up? Will the call be billed for
Kerry, the OP wanted to know if it's possible to do so so that billing
doesn't start. If you call a toll free number from overseas then you
will hear a recording telling you something like this: you will be
charged long distance charges if you continue this call.
You shouldn't be charged if you
Use dmesg
On 12/27/05, Franz Wu [EMAIL PROTECTED] wrote:
I connect to Asterisk via SSH all the times. Did not notice about console
messages about module loading.
Thanks
- Original Message -
From: BJ Weschke [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
[EMAIL PROTECTED] writes:
Bart,
We have has similar issues with BroadVoice in the past. From what I
understand they had problems with DTMF depending on which proxy you register
to. This is a bug that related to their session border controllers which
should have been resolved.
... snip
Don,
The previous question I believe was what linux are you using?
By the way, I would like to know that too, just I was trying to make this
work for weeks with no success.
Thanks,
Carlos Alperin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dov
Which firmware version are you using on your spa3000?
Peter Hoppe wrote:
|| Hello!
||
|| This is actually less a question than some information, if anyone else
|| struggles with the same issue.
||
|| I am located in the UK and use a Sipura-3000 adapter to connect to a BT
|| line (via fxo
1 - 100 of 129 matches
Mail list logo
