On Tue, 22 Aug 2006, Joseph wrote:
I was thinking of using openVPN
No problem. We are using it without problems.
Armin
--
#Joseph
On Tue, 2006-08-22 at 22:06 -0500, Brandon Galbraith wrote:
MPLS is a VPN, but it doesn't use encryption in most cases.
-brandon
On 8/22/06, Paul
Hi Douglas, Tomislav, Peter and anyone else following this.
Thank you for the comments regarding Asterisk Jobs. It is
unfortunate about the amount of jobs on the site. However,
we are well aware that the site is not the most user friendly site
in the world (yes, we admit it too) and we are
Thanks, but my problem is that I need to transfer a call, while the called
party is ringing. I cannot wait that the called answer to call.
Thanks again
Giordano
-Messaggio originale-
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Brodie Macleod
Inviato: martedì 22
Hi,
Can the column width for commands run in the Asterisk CLI be increased?
Currently when I run 'show channels' I can't see the whole channels id/name
as its to long for the columns width and is cut off. I need to grab a list of
active channels, which is currently not do able.
Thanks
Shaun
On Wed, Aug 23, 2006 at 03:41:22PM +1000, Warrick Zedi wrote:
Tzafrir,
When last did you look at AsterFax? What do you believe is required to
set it up? In what way are there wheel reinventings in either HylaFax
or AsterFax?
Tzafrir Cohen wrote:
On Wed, Aug 23, 2006 at 08:28:51AM
*blush* yes!
whois wes said:
your timeout is set to 1/2 a second (500 milliseconds).
change timeout to 5000
On 8/22/06, Eric [EMAIL PROTECTED] wrote:
I am prototyping with the Manager interface and pasting the
following into the telnet session.
Action: Originate
Channel:
Joseph wrote:
Is anybody making calls over VPN? If so what is the penalty as
encryption is involved.
I was planning to use VPN to register Sipura units to my local asterisk
this way I don't have to deal with NAT issues.
vpn's work just fine as long as the vpn end-points have enough
Hi Doug,We run with SOAP on both the client/server side, i.e. conventional AGI making SOAP request to SOAP server running on Apache. I'm sure FastAGI would be quicker but wasn't around and other than the overhead of making the SOAP request is very lightweight on the Asterisk side.
SimonOn 8/22/06,
How would I set up a call between two extensions which are both pstn
phones (and not peer devices)?
Thanks
Eric
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Hi All
When a user wishes to 'break out' they prefix there call with a 9.
I know how to remove the 9 and then dial the remaining number and this
is all working fine.
I now need to remove the 9 but then prefix another number onto the phone
number before dialing now but am unsure how to do
We've done this with OpenVPN and it works fine. I'd recommend that the VPN server is not on the same box as Asterisk. Stick it on a firewall/gateway box giving access to the network containing the Asterisk boxes behind it. This way the Asterisk box(es) is seeing normal unencrypted traffic and the
Were getting messages auto_congest:
Auto-congesting . I cant find much information about
what this means, and Ive had a look at the source code but that didnt
help me much. Could anyone point me to a description of auto congestion?
Thanks
David
Yes, I agree. And one more thing: With some encrypt setups of openvpn the
data path is 'on hold' when openvpn recreates/renegotiates a new encryption.
This means that you have a short interrupt (some milliseconds) when openvpn
server
a) establishes a new connection
b) re-creates the encrypt-key
In article [EMAIL PROTECTED], Eric [EMAIL PROTECTED] wrote:
How would I set up a call between two extensions which are both pstn
phones (and not peer devices)?
Use the Local channel. For example, to make a call to 1234567890, and when
it answers, connect it to an outgoing call to 1357924680, do
VPN's as General rule work very well compared to NAT Translation. I have two remote offices with 5 Grandstream GXP2000 SIP Phones in each both with 1Mbit SDSL. I use Juniper Netscreen 5GT's to handle the interoffice VPN's using Route Based VPN. One thing I have noticed however is you cannot
I've been asked to look into this for security reasons:
During a call, a credit / debit card payment may be required. At the
moment, the card details are given over the phone. This is not ideal for
obvious reasons.
I was wondering if it would be possible
a) for the agent to ask the caller
This is actually working as designed. You need to use type=peer in order for
call-limit to work properly, which in turn is what allows hints to work
properly.
- Brad
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Gagnon
Sent: Tuesday, August
If you already have the IP in a file, why don't you set it up so the
file itself says: externip=xx.xx.xx.xx and then do a #include in
sip.conf for the /etc/myip file? I believe you'll have to do a sip
reload either way (which can obviously be part of your cron job) if
you're not already, but
Try show channels concise
--
--
Steven
http://www.glimasoutheast.org
Shaun Hofer [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
Hi,
Can the column width for commands run in the Asterisk CLI be increased?
Currently when I run 'show channels' I can't see the whole channels
Try running apache as the asterisk user instead of
"apache"
My assumption is that "apache" or your apache user
does not have access to the voicemail folders.
-- -- Steven
http://www.glimasoutheast.org
"Sergio R. D'Ippolito" [EMAIL PROTECTED]
wrote in message news:[EMAIL
If memory serves correctly, most of the above has been raised as issues in the
past and the suggested work around has been to run a dns caching server on the
asterisk box.
FWIW, I always use IP addresses instead of dns names. But, I don't have to
deal with dynamic ip changes of any device
Dear dan
Thanks for your help,
I am using Web-MeetMe_v2.0.0.gz ,I copied app_cbmysql.c to /usr/src/asterisk/apps/
,can you please tell me how to include it at the Makefile
Regards
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Dan Austin
Sent:
I am using Asterisk-Stat http://www.areski.net/asterisk-stat-v2/about.php
If that is not what you are looking for, then maybe
its source will help you.
-- -- Steven
http://www.glimasoutheast.org
"Christopher Aloi" [EMAIL PROTECTED] wrote in
message news:[EMAIL PROTECTED]...Hello
Scott Pinhorne wrote:
Hi All
When a user wishes to 'break out' they prefix there call with a 9.
I know how to remove the 9 and then dial the remaining number and this
is all working fine.
I now need to remove the 9 but then prefix another number onto the phone
number before dialing now
Hi All
I have 2 phones registered to an asterisk server. The phones are sat
behind a NAT.
If I have the asterisk sat inline on the call after setting it up (with
transfer option specified as an example) the call works fine.
If I take out all options so the asterisk should bridge the call
Hey everybody,
I've set up an extension that allows users to send a call directly to
voice mail. Yesterday, someone accidentally sent a call to an extension
that didn't exist and the call was dropped. I found the option to check
if a mailbox exists and it works fine, but I get the following
On Wednesday 23 August 2006 06:56, Watkins, Bradley wrote:
This is actually working as designed. You need to use type=peer in order
for call-limit to work properly, which in turn is what allows hints to work
properly.
I'm sorry, but type=friend is *SUPPOSED* to equal both user and peer. If
On Wednesday 23 August 2006 07:07, Remco Barendse wrote:
I am aware that it could mean serious delays for a call to be completed if
the dns lookup was done for every call but surely it should be possible
for * to keep re-trying to resolve an ip address for previous failed
entries let's say
It's not a bug. When you use type=friend, it will create a user object
*and* a peer object. This will make call-limit not function, thereby
breaking hints. There is no reason to use friend anyway. It does not
gain you any functionality, and in fact breaks some.
- Brad
-Original
When I do a sip show peers I see that some phones have lost their registration
or is no longer reachable. When this occurs I would like the system to send
someone an email that the extension is no longer reachable.
Roger Workman
Business Development
Upperclassman/Universal Holdings LLC
Voice:
I'm using asterisk 1.2.10
David Gagnon wrote:
Are you having this problem with the trunk?
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Lucas Alvarez
Envoyé : 22 août 2006 18:23
À : Asterisk Developers Mailing List; Asterisk Users Mailing List -
Yeah, use Asterisk-Addons and configure the CDR to go into a MySQL
database. Once, there, it's really easy to use PHP or Perl to create a
custom web-page that shows whatever you want to see. I've got one set
up to search for a specific period of time, or for a specific extension.
Have you done a show channels to see if Asterisk thinks that SIP
Device is in use? I experienced this problem once after doing a
Blind-Transfer from a Cisco 7940 SIP Phone. The transferred call had
long since been disconnected, but the Cisco phone thought it still had
control of the call, so
I have tested Redfones boxes. Tried
two of them and was able to re-create some issues. I did not have PRI lines but
a 24 channel em wink line so not sure if PRI is affected as well. I found
that over time we had issues with hanging zap channels. Asterisk reported
everything was just fine
As stated in the original post, when I entter the IP with an editor
directly into sip.conf calls work just fine but I am looking for a way
to have that done _automatically_.
The Asterisk - Future of Telephony book says it is possible for Asterisk
to access a Linux environment variable
On Wednesday 23 August 2006 08:48, Watkins, Bradley wrote:
It's not a bug. When you use type=friend, it will create a user object
*and* a peer object. This will make call-limit not function, thereby
breaking hints. There is no reason to use friend anyway. It does not
gain you any
Still no answers huh?
I've asked a couple of time how to do this, and by the lack of answers, I'm
guessing there is no way.
The workaround unfortunately is to create an entry for each IP address in
the range (I hope you don't have to open up a whole C class)
-Original Message-
From:
Hello
Wouldn't the correct way of handling call limits, be using the Call Group
Applications available in Asterisk?
Regards
Jon
-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Andrew Kohlsmith
Sendt: 23. august 2006 15:30
Til:
running v1.2.10 svn checkout...
When I listen to the VM options, it says 'press 3 for advanced options',
but after pressing '3', there is nothing there with the exception of
pressing '*' to return to the main menu.
Have I missed a config option, sound file, or is the advanced option not
I'm needing some pointers from anyone who has been able to get Cisco routers to recognize the iax protocol and perform QOS on it. Or if there is a better way to get my iax traffic prioritized by the router.
-- BruceNortex Networks
___
--Bandwidth and
Andrew Kohlsmith wrote:
snip
This is broken behaviour.
I don't know why we have the distinction of users and peers in the first place. A single entry with something along the line of calltype taking incoming, outgoing or both would be far clearer and eliminate all this inconsistency.
On 8/23/06, Alistair Cunningham [EMAIL PROTECTED] wrote:
Does anyone have an opinion of:
1. Comcast Cable
2. Bellsouth DSL
for residential internet and VoIP service? I'm particularly interested
in reports on:
1. VoIP voice quality.
2. Any NAT or firewall problems with SIP.
3. How long they
Does anyone have an opinion of:
1. Comcast Cable
2. Bellsouth DSL
for residential internet and VoIP service? I'm particularly interested
in reports on:
1. VoIP voice quality.
2. Any NAT or firewall problems with SIP.
3. How long they take to install the service from date of order.
4. How
Rich Adamson wrote:
running v1.2.10 svn checkout...
When I listen to the VM options, it says 'press 3 for advanced
options', but after pressing '3', there is nothing there with the
exception of pressing '*' to return to the main menu.
Rich,
If you don't have the dialout option enabled in
On 8/23/06, Bruce Reeves [EMAIL PROTECTED] wrote:
I'm needing some pointers from anyone who has been able to get Cisco routers
to recognize the iax protocol and perform QOS on it. Or if there is a better
way to get my iax traffic prioritized by the router.
Can't you just setup a policy class
Brad,
It works with friend. I'm using this config since 1 year. I dunno why it
didn't work for Andrew.
David
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Watkins,
Bradley
Envoyé : 23 août 2006 08:48
À : Asterisk Users Mailing List - Non-Commercial
On 1.2.10, presence is working very well using friend. The state is
refreshing successfully. There is probably antoher problem with your
installation cause I'm using hint with friend since 1 years in all my
production system.
David
-Message d'origine-
De : [EMAIL PROTECTED]
Woah there... Relax, man. I will concur that there are some
inconsistencies and things are not exactly how they should be. I'm
mostly just pointing out that, for various reasons that I'm not
particularly well-equipped to discuss (oej would be able to regale you
with the necessary history if you
Bruce,
this might be able to help give you some hints or a place to start:
http://www.voip-info.org/wiki/view/QoS+Cisco
Hope that helps
\R
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Since you're using the variables to decide what to do next
(VMBOXEXISTSSTATUS), you can go ahead and set priorityjumping=no in the
general section of extensions.conf, unless you're using the n+101
priority jumping elsewhere.
On Wed, 2006-08-23 at 08:39 -0400, Doug Lytle wrote:
Hey everybody,
Agreed that with a other IAX and SIP that have registration information and
secrets that works.
The problem is when you have a provider that just sends you a SIP call and
the only way to identify it is by IP address. In those cases (if I
understand correctly) we need a host line don't we? (Or at
The easiest way is to register for free dynamic DNS
service at www.dyndns.com. Then use externhost=
instead of externip= in sip.conf . If you are using a
Linksys router like the WRT54G, it already has a
dyndns client which will update the dyndns servers
with your ip address everytime it changes.
I may have to eat my words, then. This is the case with trunk, and I can't
recall the last time I built a 1.2.x system. I could have sworn that behavior
didn't change, but I've been wrong before.
- Brad
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf
Bruce Reeves wrote:
I'm needing some pointers from anyone who has been able to get Cisco
routers
to recognize the iax protocol and perform QOS on it. Or if there is a
better
way to get my iax traffic prioritized by the router.
I just spent some time doing this myself. If your routers
Bruce Reeves wrote:
I'm needing some pointers from anyone who has been able to get Cisco
routers to recognize the iax protocol and perform QOS on it. Or if there
is a better way to get my iax traffic prioritized by the router.
You can either match on udp/4569, or, match on TOS header bits.
I'm thinking I used deny and permit statements on broadvoice.com way
back when, and the configs/sip.conf.sample suggests its still valid for
v1.2.10 code.
You might take another look at that for sip.
Benjamin Lawetz wrote:
Agreed that with a other IAX and SIP that have registration
This bridges the call on the phone and not the switch unless I am mistaken
On 8/22/06, Brodie Macleod [EMAIL PROTECTED] wrote:
Although I'm not using this firmware, attended transfers on these phones are
done like this (while talking to the person you want to transfer):
1. Press one of
I have used bellsouth dsl and comcast cable. In my experience they both
have there problems, but at least in my area I have consistently always
gotten anywhere from 2x to 3x more bandwith and reliable rates. but
thats just my 2 cents.
Mog
___
In my case it was not a class c, but just 4 separate addresses, one each
in NY, Seattle, Miami and London on the Level 3 network. I ended up
creating separate entries for each, in and out, and for the outbound
route, put all 4 in the order of their ping times. That is working nicely.
W
I'm in the process of upgrading an asterisk to 1.2.10 and started by
upgrading libpri-1.2.3 (make make install) and zaptel (make make
install).
Was about to install asterisk, but doing a ls I get the following error:
ls: relocation error: /lib/libpthread.so.0: symbol _h_errno, version
GLIBC_2.0
I have a Polycom 501 that works great from behind simple
firewalls, like Dlink, etc however behind a Cisco PIX Firewall I see the
register messages for the extensions on the Asterisk CLI but when I do a sip
show peers I see:
702/702
x.x.x.x D N
54297 UNREACHABLE
701/701
x.x.x.x D N
Also the phone can dial out from behind the
PIXbut obviously not receive calls.
Bill
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Bill Gibbs
Sent: Wednesday, August 23, 2006
11:53 AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject:
What is the process to get an IP phone registered to Asterisk? I bought an
Asterisk with a GUI and it has templates for devices such as sipura, cisco, and
xten but I am using a Fanvil IP phone. How do I load the template for my IP
phone into astrisk so that it can work?
Thanks
Wyatt
Well, you could just press the transfer button when the line starts to ring
instead of waiting for someone to answer.
-Brodie
On Wednesday 23 August 2006 02:07 am, Giordano Grandis wrote:
Thanks, but my problem is that I need to transfer a call, while the called
party is ringing. I cannot
Hi
I use a PIX 515 and had a similar problem when I started.
I turned on the fixup for SIP (as well as having nat in sip entry) and
it seems to do the trick for me.
Good Luck
SP
Bill Gibbs wrote:
Also the phone can dial out from behind the PIX…but obviously not
receive calls.
Bill
The archives should contain these details, but here they
are again-
Nearline29-
change this line: app_test.so
app_forkcdr.soTo
this:
app_test.so app_forkcdr.so app_cbmysql.so
Near line 88 add thes lines (just above this line " look:
look.c"
app_cbmysql.o: app_cbmysql.c
$(CC) -pipe
Hello,
Does anyone out there have experience or settings they can share to
help connect Asterisk to an Avaya Definity system over H.323?
If so we need your help! Please email me directly.
Many thanks,
Matt King
Managing Director, Orderly Software Ltd.
If you are running a new version of PIX sw (6.3.4 or 6.3.5), then leave
fixup on and set nat=no. The PIX is the only firewall that I have
seen that truly does nat correctly. It nat's both the source and dest
inside the packet. You can even do reinvite with multiple phones behind
a PIX and
I use Speakeasy.net and have been satisfied for a good 4 years now...
mogorman wrote:
I have used bellsouth dsl and comcast cable. In my experience they both
have there problems, but at least in my area I have consistently always
gotten anywhere from 2x to 3x more bandwith and reliable rates.
We're having a problem with calls coming in from our TE110P (an EM wink
T1) through to our queues and then when someone picks up the calls goes
dead or silent. They are becoming known as ghost calls in our
organization. It's seems to only have cropped up in the last couple
weeks though we had
We are running the default asterisk package on Ubuntu Dapper. Our
connection to the PSTN is over an IAX trunk with our provider. We are
getting really bad call quality on calls over the IAX trunk--voice
seems to be garbled or out of order and often completely breaks up.
But on internal calls
Hi all,
Just having a strange situation with no clues how to solve.
I have an Asterisk/TRIXBOX located in US and an IAX extn running on PA168V ATAin another country. All my configs seems to be on 4569 but i see my extn connected at a different port like 13569.
How can i make it to register at
Hello,
Ok, few bad words about A200.
Our company is based in Lithuania.
Our company used SPA-3000, but because of echo problems we are not using
them anymore.
Now we are trying our luck with Sangoma A200 but the following problem
occurred on few systems we installed. When calling person
Hello,
Ok, few bad words about A200.
Our company is based in Lithuania.
Our company used SPA-3000, but because of echo problems we are not using
them anymore.
Now we are trying our luck with Sangoma A200 but the following problem
occurred on few systems we installed. When calling person
There's been some (futile?) effort a while back attempting to get a
Bluetooth capable phone integrated into asterisk as a channel. The
idea, of course, was to make it possible to have asterisk utilize a
cellular connection for backup, calls on free nights/weekends, or free
in-network minutes.
We are running the default asterisk package on Ubuntu Dapper (which
has the advanced timing options used by ztdummy). Our connection to
the PSTN is over an IAX trunk with our provider. We are getting
really bad call quality on calls over the IAX trunk--voice seems to be
garbled or out of order
I have a customer that has returned two cards, a TE210P and a TE110P
because they are no longer working. Both cards were connected to an
3COM NBX system but not to the same one. On the TE210P only the port
that was connected to the NBX failed, the other works perfectly.
The
That's a very nice idea Greg. I'm not sure that my Asterisk 1.2 has the
externhost= function but it would solve my problem.
I have a dyndns.org account already that reports my externip.
Larry
Greg Delgado wrote:
The easiest way is to register for free dynamic DNS
service at
Hi,
Does anyone know how to solve this issue.
I have Asterisk box on public IP and three clients connected to it.
Unfortunately they are behind NAT (simple one-to-one). Those three clients can
make outgoing calls hassle free, but when I try to make a call between them
something is not
Glad I could help. I agree, these mailing lists are a life saver. I
personally have only been using Asterisk for about 5 months now, in fact
I have never even delt with any PBX's before (complete newbie) but
everyone here is very helpful and I am picking up a lot.
Kevin
David Cook wrote:
Aaron Daniel wrote:
Since you're using the variables to decide what to do next
(VMBOXEXISTSSTATUS), you can go ahead and set priorityjumping=no in the
Thank you very much, this took care of it.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little
andrutto wrote:
Hi,
Does anyone know how to solve this issue.
I have Asterisk box on public IP and three clients connected to it. Unfortunately they
are behind NAT (simple one-to-one). Those three clients can make outgoing calls hassle
free, but when I try to make a call between them
HiI am trying to set up * box with the ISDN hfc-s cards. One in NT mode and two in TE. I am using thebristuff-0.3.0-PRE-1r.tar.gz
. The installation went well, but soon after the zaphfc was loaded I started to receive these message in kernlog:Aug 23 21:00:08 asterisk kernel: zaphfc: empty HDLC
Strange?!?
These three phones are using g726 (this codec is configured in sip.conf and in
SIP ATA as well).
--
Zostan Dziewczyna Lata! http://link.interia.pl/f1997
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-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
incent Delporte wrote:
Hello
I'm having a problem with the Linksys 3102: With incoming PSTN
calls, I can hear the caller through the X-Ten softphone, but he can't
hear me. The problem is worse with Sjphone and the GrandStream 100
hardphone,
-signature
Size: 189 bytes
Desc: This is a digitally signed message part
Url :
http://lists.digium.com/pipermail/asterisk-users/attachments/20060823/486aabc3/attachment-0001.pgp
--
__
Do You Yahoo!?
Tired of spam
I have a test application, what it does is just connect to the
asterisk manager, and listen for events. I also set the connection
to receive on user, call and agent events.
I Noticed that everytime the queue is empty and a caller joins in,
asterisk tends to throw too many
queuememberstatus
On Wed, Aug 23, 2006 at 09:35:17PM +0200, Andrew Nowrot wrote:
Hi
I am trying to set up * box with the ISDN hfc-s cards. One in NT mode and
two in TE. I am using the bristuff-0.3.0-PRE-1r.tar.gz . The installation
went well, but soon after the zaphfc was loaded I started to receive these
[EMAIL PROTECTED] wrote:
Looking for a way to hard reset a ADIT 600 just purchased used. But it
seems to have a master password already set. We've tried the front reset
but maybe we don't have the right sequence of boot order. Any help would be
much appreciated? - Jim
Jim,
Did you
On Tue, 22 Aug 2006 16:55:37 +0200, Niklas Larsson wrote:
I'm using AMI to initiate a call, first calling the agent and when
he picks up, the call is placed to the customer. The prob is if the
user rejects the call (or they don't have cw...), the call is still
placed to the customer...
I
Can
anyone tell me where this is coming from? I cant seem to find any information
on it anywhere. I dont believe Im using special tones
anywhere. Any ideas?
Aug 23
14:30:34 WARNING[15443]: chan_zap.c:5492 ss_thread: Unable to start special
tone on 15
_
Kevin
Hey
guys,
I'm getting the
following message when I start asterisk:
Aug 23 13:42:40 WARNING[29258] loader.c:
/usr/lib/asterisk/modules/res_config_mysql.so: undefined symbol:
__pure_virtual
Aug 23 13:42:40 WARNING[29258] loader.c: Loading
module res_config_mysql.so failed!
I don't know how
Thank you Greg and RR.
externhost=myhost.dyndns.org works perfectly so figuring out how to
access a shell variable from within the CLI is no longer necessary -
although it would be nice to know!
externhost works in 1.20 onwards.
Thanks for finding the solution.
Larry
Greg Delgado wrote:
Larry Alkoff wrote:
As stated in the original post, when I entter the IP with an editor
directly into sip.conf calls work just fine but I am looking for a way
to have that done _automatically_.
The Asterisk - Future of Telephony book says it is possible for Asterisk
to access a Linux
Everybody,
What is the proper usage of NoCDR()? I keep getting the following
warning about lacks end:
Aug 23 16:34:32 WARNING[23822]: cdr.c:443 ast_cdr_free: CDR on channel
'Local/[EMAIL PROTECTED],1' not posted
Aug 23 16:34:32 WARNING[23822]: cdr.c:445 ast_cdr_free: CDR on channel
John Marvin wrote:
Larry Alkoff wrote:
As stated in the original post, when I entter the IP with an editor
directly into sip.conf calls work just fine but I am looking for a way
to have that done _automatically_.
The Asterisk - Future of Telephony book says it is possible for
Asterisk to
HiThanks for your reply.I will check it first thing in a morning and of course will let you know about results.Cheers
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Hi
Can anyone help with my following problem connecting asterisk to a new
provisioned isdn30e line.
At long last I have had BT install our new isdn 30 (I421) line for our
asterisk server after 3 months of waiting.
Before this we have used asterisk with a tdm400 and some analogue lines.
I
You can find cheap gsm (+/- 150$) gateways too, although the cheap ones
will require a additional pstn card. (expensive ones could do sip)
Zoa.
Jay Milk wrote:
There's been some (futile?) effort a while back attempting to get a
Bluetooth capable phone integrated into asterisk as a channel.
Dear All,
I need to buid an IVR that could make a request to a data
base (oracle) in a remote host.
The idea is that an user dial a extension with 2 options and one of
them ask for a data (in the case a date). This data is the field that the data
base needs to find the information
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