On Wed, Aug 23, 2006 at 11:03:23PM -0700, Steve Edwards wrote:
> On Thu, 24 Aug 2006, Jeremy McNamara wrote:
>
> >Rushowr wrote:
> >>Hey all, I have an interesting issue that just recently started when I
> >>grabbed a copy of the trunk about a week ago and compiled it. Ever since
> >>that compile,
Hello everyone,
I have released AstLinux 0.4.3:
http://sourceforge.net/projects/astlinux/
For all of those that have been waiting to switch to 0.4.x, this is
your chance. The few remaining problems with uclibc have been fixed
(i.e. voicemail timezones and voicemail -> email via MSMT
On Thu, 24 Aug 2006, Jeremy McNamara wrote:
Rushowr wrote:
Hey all, I have an interesting issue that just recently started when I
grabbed a copy of the trunk about a week ago and compiled it. Ever since
that compile, if I start Asterisk (disconnected terminal, using
safe_asterisk to launch) and
HiThe newest bristuff didn't change anything. Still the same. I was wondering if this is happening only to me or not. Does anyone has the same problem? Maybe I am messing something when loading the modules.
Does anyone have any other tips.Andrew
___
--Ban
On 08/24/06 09:02 El Flynn said the following:
Just wondering -- has anyone used the SIP phone feature on the Nokia
E60/61/70 phones? We're trying to see if this would be an OK phone to
get for the company, particularly since we're already running Asterisk.
SIP works well with asterisk, with
We are using some E61 and E70's with asterisk. Only problem we have at this
moment is that we are unable to use a password for the authentication. I
haven't found out yet why this isn't working. They are working good, but I
would like to see some small things changed in future firmware versions
(li
Douglas Garstang wrote:
It doesn't matter where you turn in Asterisk, there's gotcha's. For example, you can't put the hint stuff into realtime, and there's no inherint way to comment extensions.
It doesn't seem like Asterisk is good enough for you Doug.
Switch to one of the competitors' p
Hi,
have you looked here
http://www.newlc.com/Using-SIP-with-Nokia-Series60-and.html
thanks
atik
On 8/24/06, El Flynn <[EMAIL PROTECTED]> wrote:
Hi list,
Just wondering -- has anyone used the SIP phone feature on the Nokia E60/61/70
phones? We're trying to see if this would be an OK phone t
Rushowr wrote:
Hey all, I have an interesting issue that just recently started when I
grabbed a copy of the trunk about a week ago and compiled it. Ever since
that compile, if I start Asterisk (disconnected terminal, using
safe_asterisk to launch) and then continue on about my work with it, when
Hey all, I have an interesting issue that just recently started when I
grabbed a copy of the trunk about a week ago and compiled it. Ever since
that compile, if I start Asterisk (disconnected terminal, using
safe_asterisk to launch) and then continue on about my work with it, when I
disconnect my S
Try changing the configuration on your PAP2 linksys, more precisly the part
where is the NAT parameters, try changing the options from NO to YES.
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de andrutto
Enviado el: Miércoles, 23 de Agosto de 2006 03:41 p.m.
I could fix it.
The problem was
permissions on the directory /var/spool/asterisk/voicemail.
Thanks
De:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
En nombre de Steven
Enviado el: Miércoles, 23 de
Agosto de 2006 08:01 a.m.
Para:
asterisk-users@lists.digium.com
Asunto:
Giorgio Incantalupo wrote:
Hi,
I have an asterisk box with a sangoma a102 (two PRI ports).
Is is possible to connect port A to port B in order to use port B as a
simulation of a telco PRI line?
If yes, is there a special cable needed? How can I configure the card
and zaptel.conf?
Yes. You'll
Hi list,
Just wondering -- has anyone used the SIP phone feature on the Nokia E60/61/70
phones? We're trying to see if this would be an OK phone to get for the company,
particularly since we're already running Asterisk.
Not asking for a review of the phone, but rather how well the built-in SI
On 8/23/06, Infobox Peru <[EMAIL PROTECTED]> wrote:
maybe you could make it with PHP and its driver for Oracle.
For PHP have a look here : http://phpagi.sourceforge.net/
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailin
Hello all,
I'm using Asterisk h323 (default/NuPhone) with some success with
SJPhone. I say some success because while I'm able to receive audio
from Asterisk, I seem unable to send audio to it...
Any suggestions? Anybody managed to get this to work?
Thanks,
Matt King
I upgraded to 1.2.11 and now I see two behaviors different than the
existent in 1.2.10:
1.- I get 183 Session Progress instead of 180 Ringing.
2.- If I have three extensions, A, B and C. A using codec X, B using
also codec X and C using codec Y. If C dials to B and A tries to pick
up the call (usi
Tzafrir Cohen wrote:
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 2
maybe you could make it with PHP and its driver for Oracle.
Daniel Pizarro
www.infobox-peru.com-- Forwarded message --From: Moises Silva <
[EMAIL PROTECTED]>Date: 23-ago-2006 17:12Subject: Re: [asterisk-users] About IVR and OracleTo: Asterisk Users Mailing List - Non-Commercial Dis
Hi List,
I have an A200 with echo can. 2-FXO and 2 FXS.
Today I went and upgraded asterisk, zaptel and libpri. I ran the sangoma util to
patch asterisk. When I started up asterisk ZAP1 worked like a charm. However
ZAP2 has been acting up. I only get one way audio on it. The person that I call
Hi There,
Is there anybody who installed asterisk on freebsd
4.11 release ?
I was not succesful. please guide me.
I updated the ports and I installed the lib using
ports but when I try to install zaptel it says cannot
load it for release before than 5
I couldn't install the asterisk from ports als
I don't think you can use the template of another brand with your Fanvil.
You must configure the phone manually. First time I ear about FAnvil IP
phone so I cannot help you
David
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de
[EMAIL PROTECTED]
Envoyé : 2
I tested the trunk two days ago and I agree that the presence feature is
broken. I don't know if it's an error or if it's volunteer. I will post a
bug report on the tracker and we will see.
One thing is sure; hints are working well on 1.2.10 and not in the trunk. Is
this because they did some chan
Sure is possible. Look into google 'asterisk agi fastagi'.
Regards
On 8/23/06, Javier Lara Sanchez <[EMAIL PROTECTED]> wrote:
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 a
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
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.
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
HiThanks for your reply.I will check it first thing in a morning and of course will let you know about results.Cheers
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http:
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 acc
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
'Local/[
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 environm
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:
T
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 ho
Can
anyone tell me where this is coming from? I can’t seem to find any information
on it anywhere. I don’t believe I’m 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
_
Kevi
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...
>
>
[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 ever
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
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 event
---
> A non-text attachment was scrubbed...
> Name: not available
> Type: application/pgp-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
-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
> hardph
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
___
--Bandwidt
HiI 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 message in kernlog:Aug 23 21:00:08 asterisk kernel: zaphfc: empty HDLC fra
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 somethi
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 Te
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:
Tha
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 right.
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 www.dyndns.com.
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 car
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 a
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.
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 hangups
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 hangups
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 ATA in 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
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 bet
We're having a problem with calls coming in from our TE110P (an E&M 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 ha
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.
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
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.
__
The archives should contain these details, but here they
are again-
Near line 29-
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_cbmys
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
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 wa
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
__
Also the phone can dial out from behind the
PIX…but 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: [ast
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 UNR
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_
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 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
___
--Bandwi
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'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 informatio
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
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 already
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 Of
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.
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 l
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,
>
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
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update
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
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]
[mailto:[EMAIL
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 Di
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
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 t
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 the
Andrew Kohlsmith wrote:
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.
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 Colo
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
tota
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: asterisk-users@lists.di
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: [EMA
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 functi
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 containi
I have tested Redfone’s boxes. Tried
two of them and was able to re-create some issues. I did not have PRI lines but
a 24 channel e&m 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
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
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, s
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.
Christopher
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 -
N
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: 3
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 Message--
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
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
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
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 an
1 - 100 of 123 matches
Mail list logo