I posted this bug yesterday:
http://bugs.digium.com/view.php?id=10058
but really was hoping that one of you would be willing to try something
simple for me and reply back with your results.
Basically - I have run into a problem where Asterisk RFC2833 DTMF does not
seem to be compatible with lar
Jason,
I am at least having similar issues with rfc2833 DTMF:
http://bugs.digium.com/view.php?id=10058
On 6/20/07, Jason Ma <[EMAIL PROTECTED]> wrote:
Hi buddies,
I encountered DTMF issue when I tried to place call from x-lite to a
sip conference serice,here is the diagram.
X-lite>Asteris
He can not have the same username/secret. In trixbox - your ring group idea
is probably best...
On 6/28/07, Ryan Stille <[EMAIL PROTECTED]> wrote:
I'll start by saying I'm a trixbox user, and a new one at that, so
hopefully you can respond to me on those terms.
I have a user who works from ho
On Wed, Mar 29, 2006 at 2:12 PM, Adam Robins wrote:
>
> I am switching from IAX2 to SIP for my inter-Asterisk transport due to
> assorted quality issues following the 1.2.4 upgrade.
>
> On the server that SENDS the call, I have the following in SIP.CONF:
>
> [192.168.1.2_OB]
> type=peer
> fromuser
if I find anything I will post it
here...
On Fri, Mar 6, 2009 at 2:13 PM, Adam Robins wrote:
> no
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *tracinet
> *Sent:* Friday, March 06, 2009 2:08 PM
> *To:*
nd
username=customera-out
secret=
host=192.168.0.11
accountcode=customera
amaflags=billing
dtmfmode=auto
[customerb-out]
context=customerb
type=friend
username=customerb-out
secret=
host=192.168.0.11
accountcode=customerb
amaflags=billing
dtmfmode=auto
On Fri, Mar 6, 2009 at 2:48 P
quot; for the most
> recent versions of Asterisk.
>
> JT
>
>
> On Mar 6, 2009, at 11:29 AM, tracinet wrote:
>
> > That stinks... We are migrating to SIP from IAX2 at the moment and
> > running into the same exact problem. No way to control the
> > destination
isk instances
> which could then be used by the dialplan to split your calls apart
> into different contexts or behaviors.
>
> See function "SIP_HEADER" and application "SIPAddHeader" for the most
> recent versions of Asterisk.
>
> JT
>
>
> On Mar 6, 2
in those headers between Asterisk instances
> > which could then be used by the dialplan to split your calls apart
> > into different contexts or behaviors.
> >
> > See function "SIP_HEADER" and application "SIPAddHeader" for the most
> > r
This may sound odd, but I have the opposite problem. I am trying
to disable call waiting by only allowing 1 call to come into the
SPA-941, but I am getting 2 calls per line key. If have 1
extension set up on all 4 line keys, the phone handles 8 incoming
calls. I have a few customers that like th
Phone quality is pretty good with asterisk, however, customizing the
phone is a total pain in the rear. Did not have much luck with it
(mostly because it was too cumbersome and I ran out of time). We
decided to not go with the Mitels and are sticking to the Linksys
SPA-941 phones. For the $$ you
We had to stop offering the GXP-2000 due to all the same issues
mentioned above. Really not for business use. Have had good
results with Linksys SPA-941.On 2/19/06, mustardman29 <[EMAIL PROTECTED]> wrote:
I have both as well,I mostly agree with you about the display. The buttons are ok but not g
far as that is
concerned. Due to NDA, I can not discuss pricing here, but I have
talked to my rep and he has agreed to offer excellent pricing terms to
the asterisk users here. Just mention "tracinet" when you call
him and tell him you are an asterisk user. Here is his contact
info:
Scot
I am also seeing this in my logs.On 11/30/05, Aaron Daniel <[EMAIL PROTECTED]> wrote:
We just upgraded our current asterisk cluster to the release version ofAsterisk 1.2.0. Strange enough, out of the 11000+ calls, only 720 (andcounting) have a disposition of FAILED in the cdr's. These 720+ have
on
Been working on testing asterisk 1.2 before upgrading our production
systems from 1.0.x and have found a few issues. The one I am
working on
now involves DTMF failure with the following setup:
*Linksys SPA-941* ---SIP---> *asteriskA(v.1.2)* IAX2> *asteriskB(v.1.2)* SIP-> *Global
no change in behavior.
What still doesn't make sense to me is that why would this not work
with asterisk 1.2 yet still work when used with asterisk 1.0.x?
On 12/5/05, tracinet <[EMAIL PROTECTED]> wrote:
Been working on testing asterisk 1.2 before upgrading our production
systems from 1.0
This can be done. Just make sure you specify the DTMF mode you are using per SIP account. for example:
[sipuser]
type=friend
secret=blahblah
qualify=yes
nat=yes
disallow=all
allow=ulaw
dtmfmode=rfc2833
etc.
etc.
[sipprovider]
type=friend
secret=blahblah
qualify=yes
nat=yes
disallow=all
all
What does your /etc/asterisk/cdr_mysql.conf look like?
On 12/12/05, Juanjo Portela <[EMAIL PROTECTED]> wrote:
Dear Tomislav & Colleagues,
I read your post in the Astersik-list about you can not compile the cdr_mysql on Asterisk 1.2
Well, I have a similar problem.
I compiled the cdr_mysql module,
I believe you are missing 2 variables in your conf file:
table=cdr
(the table your cdrs should be stored)
sock=/var/lib/mysql/mysql.sock
(the location to your mysql.sock)
try something like this:
[global]
hostname=localhost
dbname=dbasterisk
table=cdr
password=dbpassword
user=dbuser
sock=/var/
Oswald,
I have had the same issues with Sipura devices since moving to Asterisk
1.2 as well. We use rfc2833 exclusively in our network and the
Sipura devices just stopped working with regard to DTMF. After
MANY packet captures comparing Sipura devices which did not work to
Cisco devices that did
Instead of hostname=localhost, it would be hostname=IP address of MySQL server.On 12/12/05,
Innocent Evil <[EMAIL PROTECTED]> wrote:
I was also following this thread.
Would anybody please tell, what would be configuration file if mysql is a different machine than asterisk box?
Thanks,
--You
I actually opened a bug report on this earlier this month:
http://bugs.digium.com/view.php?id=5918
I have tried a new SVN version from a few days ago and it still showed as FAILED for me in the following scenario:
incoming call from PSTN ---SIP---> asterisk ---IAX2---> asterisk ---SIP---> SIP Ph
I should note that in the following scenario:
incoming call from PSTN ---SIP---> asteriskA ---IAX2---> asteriskB ---SIP---> SIP Phone
The call log does show disposition ANSWERED on asteriskA, but FAILED on asteriskB.
On 12/15/05, tracinet <[EMAIL PROTECTED]> wrote:
I actual
least sees, and italways shows a code of 16...Aarontracinet wrote:
> I should note that in the following scenario:>> incoming call from PSTN ---SIP---> asteriskA ---IAX2---> asteriskB> ---SIP---> SIP Phone>> The call log does show disposition ANSWERED on asteriskA, but
I don't know about you, but my option 4 says "change password" yet when
I press it, it does give me the option to remove the temp
greeting. A bit confusing - I agree.On 12/12/05, Matt <[EMAIL PROTECTED]> wrote:
Thank you that will do it.Wow that was slightly not intuitive :)On 12/12/05, Steve Blai
roOn 12/21/05, Kevin P. Fleming <[EMAIL PROTECTED]> wrote:
tracinet wrote:> I don't know about you, but my option 4 says "change password" yet when I> press it, it does give me the option to remove the temp greeting. A bit> confusing - I agree.
That means you
Had the same issue with Fedora Core 1. The way we overcame the
issue was to use a 3Ware SATA controller. The 3Ware card was
detected as a SCSI controller and the drives work perfectly. If
that is not an option, you may want to check if the BIOS will let you
run the SATA drives in legacy mode whi
Nothing wrong at all - this is the Merry Christmas thread. Feel free to start a Happy Kwanza thread :)
Merry Christmas to all !!On 12/23/05, Mark Phillips <[EMAIL PROTECTED]> wrote:
What's wrong with us that celebrate Kwanza?Mark, G7LTT/KC2ENIRandolph, NJhttp://www.g7ltt.comDmitry Ivanov wrote:>
Below is what I have in my notes regarding CentOS. I am not sure exactly where I originally found this - could have been right from this mailing list actually!
"Patch CentOS
spinlock.h file before installing zaptel
Rebuilding Zaptel - Every
time there is a kernel update with yum (which is the ca
Out of curiosity - are you running an smp kernel or a uniproc kernel? I am doing some benchmarking as well on a similar system (dual XEON 3Ghz with 4GB RAM and SATA drives in mirrors) and am seeing the uniproc kernel performing better under CentOS
4.3 (2.6.9-34.0.2.EL) when testing general server
Spent quite a bit of time troubleshooting this and figured it would save someone a lot of time if this was documented
(thanks to drumkilla, file and Juggie for their assistance on this as well).
Been using Broadvox DIDs to receive incoming calls for over a year now with an older version of asteris
Sounds like they are providing a Vonage-style service that is tied into
the phones. Not sure they will sell them "unlocked". Looks
cool though.On 11/22/05, Lenny Tropiano / asterisk.org Mailing list <[EMAIL PROTECTED]
> wrote:Another IP phone possibility for Asterisk.No, not the SPA941 (from the
Have you tried the "soft hangup" command?On 11/24/05, Paradise Dove <[EMAIL PROTECTED]> wrote:
hi,how can i hangup such calls without restarting asterisk?the Zap channel on this case is busy for more than 7 hours
some logs are followed.thanks,Paradise Dove-Nov 23 16:59:49 NOTICE
We have been using MBX for some time and they are great to work with. We supplied them our requirements including the links to the digium site for the various cards we wanted to use and they came up with configuration that works great. Jon Frank is our sales rep and he is VERY attentive and quick
What are your zttest results? zttest can be run from
/usr/src/zaptel/ directory (run ./zttest from there). Do you have
Digium hardware or ztdummy?
Pedro
http://www.TRACI.netOn 3/18/06, Rana Dutt <[EMAIL PROTECTED]> wrote:
We have two Linksys 942 phones which
sound great when they call each other
Any progress on this? Would love to be able to detect DND on this phone.
- PedroOn 2/14/06, Josh Dady <[EMAIL PROTECTED]> wrote:
(now that I've remembered which address is subscribed to this list)Does anyone with one of these phones have any sort of presenceworking? I'm looking to monitor the DN
Unfortunately, Linksys is reserving the provisioning tools/info to their official resellers. The idea is that you pay your Linksys reseller to provision your phones (does not make ANY sense to me all). As a service provider, we should be able to buy these phones and have access to the bulk provis
I just issue a 'service op_panel reload' at the command line and it reloads the configuration without stopping the service :)On 5/7/06, Tzafrir Cohen <
[EMAIL PROTECTED]> wrote:On Sun, May 07, 2006 at 08:44:41AM -0400, Doug Lytle wrote:
> Wilson Pickett wrote:> >>> No, you have to kill the op_serve
What you *could* do is record one greeting as the unavailable message and another as the busy message and during the day, just play the unavailable one and at night play the busy one...
On 11/15/06, C F <[EMAIL PROTECTED]> wrote:
On 11/15/06, Wildheart <[EMAIL PROTECTED]> wrote:> Hi,>>I want to
39 matches
Mail list logo
