Hi
I noticed that my system has three sets of data regarding telephony
behaviour in different parts of the world:
1. libtonezone , part of zaptel, and the data is from the source file
zaptel/zonedata.c . Zaptel seems to use it for generating some tones.
2. /etc/asterisk/indications.conf .
Antoine Fressancourt wrote:
I will sum up the results of my investigations :
- When canreinvite is set to yes, I manage to make a video call
between the 2 parties, when I emit a DTMF signal, it triggers the
playback of a sound clip correctly, but I can't playback a video clip.
What's the
[EMAIL PROTECTED] wrote:
The issues we would like to resolve are the following:
1) We can ping our originating SIP providers. However, that doesn't guaratee us that we
can receive calls from them. In several occasions, some of our SIP providers have had
routing (SIP) problems and when we dial
Using tdm400. While transfering a call from outside to another
extensions, while this outside call is waiting with music, the
another extension call hangs up suddenly, and the call is back to the
outside call suddenly.
Wathcing logs:
Jan 15 13:32:44 DEBUG[30148] res_musiconhold.c: Read 462
I implemented something on these lines for unattended transfer. Basically
what I did was storing the call-id in an inheritable diaplan variable and
then starting a new mixmonitor on the transferred extension.
Hope this helps,
l.
On Mon, 15 Jan 2007 22:45:49 +0100, Jay Moore [EMAIL
Hi
Someone complained in #asterisk-gui that the Zaptel hardware detection
did not work. It seems that zapscan.bin did not work properly, as no
/etc/asterisk/zapscan.conf was generated.
Solution: detect zaptel hardware with genzaptelconf, and generate
zapscan.conf from it.
genzaptelconf
grep
2007/1/16, Leo Ann Boon [EMAIL PROTECTED]:
Antoine Fressancourt wrote:
I will sum up the results of my investigations :
- When canreinvite is set to yes, I manage to make a video call
between the 2 parties, when I emit a DTMF signal, it triggers the
playback of a sound clip correctly, but I
On 15 Jan 2007, at 06:01, Tomer Horn wrote:
Hello,
I am looking to purchase a new quad-band cellphone and I'm looking
for one with WiFi and enough CPU power for stable SIP calls. I was
wondering if anyone here can share his experience and recommend on
a good cellphone. Any chance there
Hello Gents,
following on this discussion, anyone particularly have one view or the
other about 1.4 and the voicemail and meetme enhancements (supposedly)
it has? We're not in production yet, I've tested 1.2 up until 1.2.13
in the lab as well as 1.4b3, since none of them got a real hammering
Its
There is no problem with the CPU utilization, it is around 40%, I will not
be able to try this without the VPN, maybe I should try another VPN solution
like OpenSwan, or PPTP.
Why do you think that IAX will make a difference than SIP?
On 1/16/07, Gordon Henderson [EMAIL PROTECTED] wrote:
On
Hello,
After upgrading to 1.4 my ENUMLOOKUP returns nothing. Even with new
format.
I've tried commands:
SET(foo=${ENUMLOOKUP(+13015611020loligo.com)});
${ENUMLOOKUP(+13015611020,ALL,c,,e164.org)};
Could I turn debug of ENUMLOOKUP on?
Thanks.
Michael
Hi everyone!
I'm wondering if anyone on the list had the opportunity to work with an NTT
INS1500 ISDN PRI service before.
You see, in Japan, if you receive a call that was just forwarded by another
number, the call presentation not only includes the caller (ANI) and your
number (DNIS), it will
Hi
i have a similar dialplan:
exten = 99,1,Gotoif(?2:3)
exten = 99,2,Meetme(100)
exten = 99,3,Meetme(100|options)
i'd like to do something like:
exten = 99,1,Gotoif(?2:4)
exten = 99,2,Meetme(100)
exten = 99,4, ... exit ...
exten = 99,3,Meetme(100|options)
How can i exit the dialplan?
Hello everybody,
I have a small problem: I've set language=it in iax.conf, but MeetMe
conferences still play en files. I see from the CLI that the Playback
app is called with language=en parameter.
From the sources of app_meetme I see that it takes the language from
the channel, so I think this
Hello,
I would like the IP phones to not accespt SIP requests (like INVITE) from any
device other than its proxy. Snom phones ignore this while Cisco Polycom
accepts the call. Any idea what to do to disable it?
Thanks! __Yehavi:
exten = 99,4,Hangup
?
nik600 wrote:
Hi
i have a similar dialplan:
exten = 99,1,Gotoif(?2:3)
exten = 99,2,Meetme(100)
exten = 99,3,Meetme(100|options)
i'd like to do something like:
exten = 99,1,Gotoif(?2:4)
exten = 99,2,Meetme(100)
exten = 99,4, ... exit ...
exten =
IAX allows trunking. With 16 channels open between 2 asterisk servers
that should recover over 400kbs compared to SIP.
I believe OpenVPN can be forced into a no-encryption test mode. If you
can do that temporarily it is easier to judge the vpn overhead.
O.Kamal wrote:
There is no problem with
On Jan 11, 2007, at 11:08 PM, Jarek Jarzebowski wrote:
Dnia 31-12-2006 o 18:05:00 Jarek Jarzebowski [EMAIL PROTECTED]
napisał(a):
Dnia 31-12-2006 o 17:39:10 Tzafrir Cohen
[EMAIL PROTECTED] napisał(a):
On Sun, Dec 31, 2006 at 05:08:26PM +0100, Jarek Jarzebowski wrote:
Dnia 31-12-2006 o
On Monday 15 January 2007 6:21 pm, Anselm Martin Hoffmeister wrote:
could you verify or negate that adding the T option makes it work?
That or transfer=no in iax.conf for hte user/peer entries involved. I never
thought of IAX2 transfers here, for some reason I thought that Asterisk was
I have a soundpoint 501 phone that has locked up twice now. You can make
a call but when a call is sent to it, it responds with sip busy
messages. You get the same message when the phone is in do not disturb.
I reset to defaults the first time and it worked for a week or so and
then stopped. The
Good Day List,
I'm having some issues with echo cancel on my Asterisk box, and have
done
extensive reading and have gained some useful pointers from this list
but have a couple of hopefully fairly simple questions.
The Asterisk box is connected via 20 FXO ports on a TDM2400 with the
Hardware
Pode.
On 16 Jan 2007, at 10:21 , Tim Panton wrote:
On 15 Jan 2007, at 06:01, Tomer Horn wrote:
Hello,
I am looking to purchase a new quad-band cellphone and I'm looking
for one with WiFi and enough CPU power for stable SIP calls. I was
wondering if anyone here can share his experience
currently using 1.2.14 and zaptel 1.2.12
i'm using mfc/r2 so i can't move to 1.4 with sip jitter control and
improved jitter control in zaptel 1.4.
my problem is excessive jitter using linksys spa942.
when i set canreinvite=no, which forces rtp to pass through *, quality
is horrible.
On 1/15/07, Douglas Garstang [EMAIL PROTECTED] wrote:
-Original Message-
From: BJ Weschke [mailto:[EMAIL PROTECTED]
Sent: Monday, January 15, 2007 3:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Queue cmd option 'i'
On 1/15/07,
ron,
i recently fixed the poor quality with our spa942 using canreinvite=yes
have you found out the problem with your spa941? i can't get around the
problem of poor quality audio when the rtp pass through *
[EMAIL PROTECTED] wrote:
Hello,
This is not exactly an Asterisk question, but I was
Hello Asterisk,
Can anyone help or put some light on, how can I configure Asterisk to
work with SS7?
What do I need, in terms of Hardware and Software?
Regards,
Neel
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing
I have read that an IAX trunk requires a timing device. What wasn't
clear to me was whether it is like TDM ie 1 timing device for the trunk,
or if each end requires a timing device. I have a zaptel card in one
server; do I have to have one in the second server in order to do an IAX
trunk?
I set
If the other server doesnt have any hardware device that can act as timer.
then just compile zaptel and modprobe ztdummy .. This kernel module should
act as timing source i think . ( it works with meetme ) .
On 16/01/07, Andy Hester [EMAIL PROTECTED] wrote:
I have read that an IAX trunk
I have call center PCs that switch between an IBEAM SIP softphone and a NEBU
IAX softphone (for reasons
that aren't germane here). The SIP softphones work fine, but the IAX
softphones get a fast busy unless I give
them an IAX trunk to use, instead of the PRI trunks that all the other phones
its notransfer=yes in iax.conf not transfer=no :)
On 16/01/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Monday 15 January 2007 6:21 pm, Anselm Martin Hoffmeister wrote:
could you verify or negate that adding the T option makes it work?
That or transfer=no in iax.conf for hte user/peer
On Tuesday 16 January 2007 10:09 am, Vicky wrote:
its notransfer=yes in iax.conf not transfer=no :)
Ahh yes. force consistency in the CLI where it doesn't necessarily belong,
but use idiotic variable names in the config files. :-)
-A.
___
I don't know if I understand you correctly but you could place a Goto or a
Hangup there:
exten = 99,1,Gotoif(?2:4)
exten = 99,2,Meetme(100)
exten = 99,3,Goto or Hangup
exten = 99,4,Meetme(100|options)
-Original Message-
From: nik600 [mailto:[EMAIL PROTECTED]
Sent: Tuesday, January
I can only say that we use 1.4 in production (150 sip phones, sangoma
and te410p cards, 75 agents 25+ queues). We do have a segfault now and
then, but as I mentioned in a previous post, I think that the causes of
that have been fixed in the 1.4 svn branch already.
Julian.
RR wrote:
Hello
notransfer has been deprecated in 1.4 in favor of transfer
ast_log(LOG_NOTICE, The option 'notransfer' is deprecated in favor of
'transfer' which has options 'yes', 'no', and 'mediaonly'\n);
- Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Tuesday 16 January 2007 10:09 am, Vicky wrote:
its
On Tuesday 16 January 2007 11:58 am, Jason Parker wrote:
notransfer has been deprecated in 1.4 in favor of transfer
ast_log(LOG_NOTICE, The option 'notransfer' is deprecated in favor of
'transfer' which has options 'yes', 'no', and 'mediaonly'\n);
Sure, make an ass out of me, or rather Vicky
You need a timing device on both ends.
Zoa
Vicky wrote:
If the other server doesnt have any hardware device that can act as
timer. then just compile zaptel and modprobe ztdummy .. This kernel
module should act as timing source i think . ( it works with meetme ) .
On 16/01/07, *Andy Hester*
We have been running an Asterisk box with 1.2.9.1 on it since August in
a call center environment. We use the Asterisk box as an IVR and then
pass the calls on to a Nortel Option 11C. Today we found in our long
distance bill two calls that lasted a VERY long time. One was 58 hours
and another was
Hey All,
I am attempting to get the RXFax app working and having a hell of a time
of it. I am hoping that some of you fine folks can help me out.
I have installed Asterisk v1.2.14, SpanDSP v0.0.2pre26 and app_rxfax.
All compiled and installed fine.
When I attempt to call the
Savoy, Kevin - Williston, ND wrote:
We have been running an Asterisk box with 1.2.9.1 on it since August in
a call center environment. We use the Asterisk box as an IVR and then
pass the calls on to a Nortel Option 11C. Today we found in our long
distance bill two calls that lasted a VERY long
I upgraded my Asterisk configuration to 1.4.0 yesterday, when I was adding a
TDM400P to have two PSTN connections to my analog phone lines. Adding the
phone lines was a success, however, I now notice the MP3Player audio sounds
horrible (incomprehensible). The only changes that I have made to the
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Zoa
Sent: Tuesday, January 16, 2007 11:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] IAX Trunk timing
You need a timing device on
Hey All,
Nevermind this question. I figured out that my problem was that I
needed to downgrade my libtiff library to v3.7.1. My OS had installed
3.8.2 during system install and apparently spandsp doesn't like that
version.
It's all working perfectly now. Thanks in any case!
Darren
On 16 Jan 2007, at 13:46, Patrick W. Foster wrote:
I am hopeful that
someone will recognize the issue and give me a pointer on where to
look for the problem.
- Registered IAX2 '4414' (AUTHENTICATED) at 192.168.1.102:4569
-- Accepting AUTHENTICATED call from 192.168.1.102:
On 12:06, Tue 16 Jan 07, Andy Hester wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Zoa
Sent: Tuesday, January 16, 2007 11:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
Adam Sharples wrote:
I want to tune to echo canceller, but am unsure if any of the options
provided have any effect on the hardware module. Do the settings such
as
echocancel and echotraining in Zapata.conf affect the hardware module?
No. Hardware echo cancelers on Digium cards are
Though I haven't really tried it out myself, one option I've seen would be
to use the same set of A100-series cards (the same one also being used for
T1/E1/J1) from Sangoma to handle the physical port and their commercial SS7
gateway software for the out-of-band signalling. However, the SS7
its notransfer=yes in iax.conf not transfer=no :)
this is getting close!
however, it takes about SEVEN seconds after the called party hangs up
before the next priority is executed - same as with the T option.
as contrast to h option, when called party hits asterisk, the next
priority is almost
You should:
Set(TIMEOUT(absolute)=14400)
When the call is received - this will set the maximum limit of a call and
asterisk will force hang-up when the limit is reached.
14400 seconds = 4 hours, which for our purposes is longer then any call we
expect. Even if you double-it or set it to several
I have Teliax trunk set to ulaw and g729 and I have a modem/fax extension
from a sipura forced to ulaw. When the call goes out through Teliax IAX
trunk, asterisk transcodes to g729. Is there a way to tell asterisk not to
transcode calls from/to a specific extension?
I'm running asterisk 1.2.4
Greetings,
The District of North Vancouver, a municipal government in BC, Canada,
is hosting a Digium instructed Asterisk Bootcamp at our training center
from February 5th-9th, 2007. Primarily arranged to provide training to
some of our staff, there is space available for others to avail of
Hi,
Someone knows an Open Source solution that can handle Outbound IVR for
asterisk?. What I'm looking it a pre-preprogrammed a telephone call that reach
a Person and start making an Interview over the telephone.
Specifically I want to call all my customers exactly one hour after the service
I had the same problem last year, at the time for some reason Timeout
instruction wouldn't trigger, so, just to be sure not to have to pay
for another longdistance call, I did the following, (following
someone's advise in here)
/usr/sbin/asterisk -rx show channels concise |awk -F : '($11
On Tuesday 16 January 2007 2:31 pm, chester c young wrote:
however, it takes about SEVEN seconds after the called party hangs up
before the next priority is executed - same as with the T option.
What kind of last leg are these calls? to POTS (even CAS T1) or PRI?
as contrast to h option,
Am Dienstag, den 16.01.2007, 12:01 -0800 schrieb Alejandro Duplat:
Hi,
Someone knows an Open Source solution that can handle Outbound IVR for
asterisk?. What I'm looking it a pre-preprogrammed a telephone call that
reach a Person and start making an Interview over the telephone.
I sent this yesterday, but saw zero traffic, so I think it got lost in
the ether, so I'm sending again.
I'm having an issue using Asterisk 1.2.14 and an Audiocodes Mediant 1000
ISDN gateway. For the most part, everything is working except for
attended transfers. When I do an attended transfer,
Hi,
We are using MozIAX as a softphone with a USB headset and are making
outbound calls using IAX with ulaw encoding to our voip provider.
We're running asterisk 1.4
Users are complaining that the ring tone generated by asterisk is much
louder than the voice call once connected. They are having
Gary Mensenares wrote:
Though I haven't really tried it out myself, one option I've seen would be
to use the same set of A100-series cards (the same one also being used for
T1/E1/J1) from Sangoma to handle the physical port and their commercial SS7
gateway software for the out-of-band
Hi,
Having moved from asterisk-1.2.8 to 1.2.14, i've noticed that startup is
much slower. In other words, if I say asterisk -R they type stop now,
it takes on the order of 7 seconds instead of 1 second. The old asterisk
startup printed out something like 650 lines, whereas the new one prints out
Hi,
I'm trying to configure Asterisk and DISA.
Asterisk is working, but I cannot have DISA dialing out.
This is a snippet of my extensions.conf:
[internal]
exten = 1003,1,DISA(no-password|outgoing2)
[outgoing2]
exten = 1003,1,Playback(beep.gsm)
exten = 1005,1,Playback(beep.gsm)
My
Hi,
I've got an Asterisk setup including a TDM2400 for analog trunks
extensions plus two IP501s an IP601 (all firmware 1.6.7 as supplied).
The initial buddy / hint setup was fairly straightforward, but I have a
strange problem in that some extensions don't show any status indication.
Asterisk
On 16 Jan 2007, at 19:56, Victor Perez wrote:
I have Teliax trunk set to ulaw and g729 and I have a modem/fax
extension from a sipura forced to ulaw. When the call goes out
through Teliax IAX trunk, asterisk transcodes to g729. Is there a
way to tell asterisk not to transcode calls
On 16 Jan 2007, at 20:33, Russell Horn wrote:
Hi,
We are using MozIAX as a softphone with a USB headset and are making
outbound calls using IAX with ulaw encoding to our voip provider.
We're running asterisk 1.4
Users are complaining that the ring tone generated by asterisk is much
louder
I have some Audiocodes units which appear to be running Linux,
according to the unit's own System Log
kern.warn Linux version 2.4.21openrg-rmk1 #2 Wed Aug 30 17:05:29 IDT 2006
However my contact at Audiocodes claims otherwise
On 12/4/06, Yaniv Nizan [EMAIL PROTECTED] wrote:
I doubt that
I'm aware of Cingular being GSM. We're standardizing on Sprint since
Cingular is less than optimal around here.
Even with LNP, knowing the NPA-NXX would nail probably 90%+ of our people.
The ones that are on LNP could be added as 10 digit LCR.
From a technical standpoint, can * handle over
Hello Asterisk,
Please can anyone explain what is the difference between TE405P and TE410P?
According to the data sheet, the difference I see is the PCI voltage.
TE405P use only 5.0 volt PCI slot and TE410P use only 3.3 volt PCI.
Regards,
Nitesh
___
On Tue, 16 Jan 2007, Eric Germann wrote:
I'm aware of Cingular being GSM. We're standardizing on Sprint since
Cingular is less than optimal around here.
Where is here? Planet Earth?
Down here (San Diego), Cingular advertises they have the fewest dropped
calls. I think they pulled a Clinton
2007/1/11, Ale [EMAIL PROTECTED]:
Ciao,
Enrico Pasqualotto wrote:
Is better ultramonkey, dundi or SER proxy in front of * server?
You can also consider Hartbeat + rsync, or simply pfsync + rsync ;)
The problem with Asterisk HA, is mainly the lost of calls when
failover occurs. This is
KS == Savoy, Kevin - Williston, ND [EMAIL PROTECTED] writes:
KS We have been running an Asterisk box with 1.2.9.1 on it since
KS August in a call center environment. We use the Asterisk box as an
KS IVR and then pass the calls on to a Nortel Option 11C. Today we
KS found in our long distance
Hi Nitesh -
Please can anyone explain what is the difference between TE405P and TE410P?
According to the data sheet, the difference I see is the PCI voltage.
TE405P use only 5.0 volt PCI slot and TE410P use only 3.3 volt PCI.
That is the only difference.
- Noah
Andrew Joakimsen wrote:
2) What does one go about doing to correct GPL violations? Perhaps
someone has a generic legal letter that can be used in these
situations?
Only a copyright holder whose code is being used outside the terms of
the GPL can pursue action against the violator.
Are all of the sip phones in the same context?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Robert Jenkins
Sent: Tuesday, January 16, 2007 1:44 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject:
the answer sucks, but is apparently correct.
imho Andrew Kohlsmith is The Man, although there was someone in Germany
who emailed about the T option which actually works about as well -
please email me. Andrew Kohlsmith please email me. Will pay paypal
if that's ok.
--- Andrew Kohlsmith
Andrew Joakimsen wrote:
I have some Audiocodes units which appear to be running Linux,
according to the unit's own System Log
kern.warn Linux version 2.4.21openrg-rmk1 #2 Wed Aug 30 17:05:29 IDT 2006
Googling turns up:
http://www.jungo.com/openrg/openrg.html
OpenRG is a Linux based device
Hi,
Yes, there are just the three Polycoms (200 - 202), the rest of the system
is analog. The Polycoms always 'see' each other, the problem is with them
seeing some Zap channels.
Although the 501s don't have the display of the 601 plus sidecar, from
Asterisks point of view the 'watchers' count
Am Mittwoch, den 17.01.2007, 07:38 +0800 schrieb Leo Ann Boon:
Andrew Joakimsen wrote:
I have some Audiocodes units which appear to be running Linux,
according to the unit's own System Log
kern.warn Linux version 2.4.21openrg-rmk1 #2 Wed Aug 30 17:05:29 IDT 2006
Googling turns up:
On Mon, 2007-01-15 at 15:26 -0600, David Gomillion wrote:
I don't think you can do that. Here's why: on the Polycom's, the
Transfer button doesn't reappear until the transferree picks up the
phone. Unless something changed in the firmware recently. But, if you're
completing it before the 3rd party
There is nothing in the GPL that prohibits you from selling the software
(RedHat Software). There is also nothing stops a sales person from denying
it.
They must provide a copy of the GPL and they must give you the source code
and related modifications if you ask (not sure if you have). There are
Am Dienstag, den 16.01.2007, 15:04 -0800 schrieb chester c young:
the answer sucks, but is apparently correct.
If your application involves the caller (e.g. an employee of your
company) to rate the call he just did, or to enter any data to a mysql
database over the phone right after the call,
You are trying to subscribe to a non SIP channel?
Not sure that can be done...never tried.
-Original Message-
From: Robert Jenkins [mailto:[EMAIL PROTECTED]
Sent: Tuesday, January 16, 2007 4:43 PM
To: Damon Estep; 'Asterisk Users Mailing List - Non-Commercial
Discussion'
Subject: RE:
Hi guys,
I did an upgrade on one asterisk from 1.2.14 to 1.4.0, after this, all
calls originated from PBX trunked with asterisk through TE110 board i
receive this message:
[Jan 16 21:19:42] ERROR[2453]: chan_zap.c:8142 zt_pri_error: !!
Unexpected Channel selection 3
the call was completed and
show channels will display all calls including Duration and BridgeTo. You
can check the BridgeTo column to determine if one call leg is still attached
to the other. If that fails, you can also check the duration for hung calls.
To automate, there are a number of approaches. I personally suggest
I need a method of limiting the duration of calls when RTP media does
NOT travel through Asterisk.
I know that the Dial() command limit option L requires Asterisk to
carry the media, but what about Set(TIMEOUT(absolute)=XX)?
Are there any other apps/options that might work for this?
Thanks!
All,
Just came across the prompt #3 from inside the top menu of VM in latest stable.
Allison does not announce the prompt, but if you know it is there, you can
press 3 successfully follow the prompts from there to send your message to
other users on the system. But, of course, obviously, I am
Hi,
How do you folks handle really large queues (350+ simultaneous
callers) in your Asterisk PBXes?
We're going to be bringing in around 16 PRIs' worth of inbound
callers, doing skills-based routing, and queuing them up for
approximately 200 agents.
What's the best way to handle all of these
Hi guys,
Look my example:
pabx*CLI sip show channels
Peer User/ANRCall ID Seq (Tx/Rx) Form Hold
Last Message
64.71.xx.xx322121226ee03b46000 00103/15992 unkn No (d) Rx:
BYE
64.71.xx.xx0113941735 57344d766af 00103/0 unkn No
$25 to openvpn.org - thanks to Anselm Martin Hoffmeister
--- Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote:
Am Dienstag, den 16.01.2007, 15:04 -0800 schrieb chester c young:
the answer sucks, but is apparently correct.
If your application involves the caller (e.g. an employee of your
Hi there
The dnsmgr in Aterisk 1.4.0 seems not to work. I enabled DNS lookups in
dnsmgr.conf but after reloading the conf files * never refreshes DNS lookups.
Any ideas how to debug this issue?
Thanks in advance
Housi Mueller
-
Don't pick
Hi All,
I'm using asterisk 1.2.9.1 and mysql 3.23, asterisk add-ons 1.2.3.
All seems to work normally with realtime voicemail, reads vmbox
parameters from the db fine. When I try to change the password,
asterisk operates normally, enter new password ok, re-enter new
password ok, password has
On Tue, 16 Jan 2007, Christopher Snell wrote:
Idea #1: Use servers with (2) Digium 4-port PRI cards, running
Asterisk, as media gateways. From here, send calls to 2 or more
Asterisk queue servers. For each incoming call, run an AGI on the
media gateways that determines which queue server is
I too seem to have the same problem, dont know about poor quality
but its certainly not loud enough, I have to put my mouth to the
microphone, otherwise the other end reports they cannot hear me. This
does however seem to do a good job to cancel out the background noise
On 11/10/06, Ron Winograd
Hi List,
any good comparison between TDM404B and TDM2401B .
i'm not very happy with TDM404B voice quality, low volume and sometimes echo.
I was wondering if any of you guys have good experience with TDM2401B.
thanks!___
--Bandwidth and Colocation
Hi All,
I'm having some troubles with my Dell 860 and TE110P card. Using
Asterisk 1.2.14, Zaptel 1.2.12 and Libpri 1.2.4. I'm getting digital
noise, like a half ring almost and other jitter. Here's the kicker it's
only on the outside part of the call. Ie. if I rang you, you would here
it but I
Andrew Joakimsen wrote:
I too seem to have the same problem, dont know about poor quality
but its certainly not loud enough, I have to put my mouth to the
microphone, otherwise the other end reports they cannot hear me. This
does however seem to do a good job to cancel out the background noise
On 2007-01-14 22:01:44 -0800, Tomer Horn [EMAIL PROTECTED] said:
Hello,
I am looking to purchase a new quad-band cellphone and I'm looking for
one with WiFi and enough CPU power for stable SIP calls. I was
wondering if anyone here can share his experience and recommend on a
good cellphone.
Guys, anybody has seen or is using some kind of softphone on any square
screen device with WM5? Ive tried sjlabs one and xten for pocket pc and they
do work on Wm5 but they are designed for standard screens, anybody using
anything on square ones?
___
Hello friends,
I am trying to install asterisk 1.4. I am configuring it as follows:-
./configure --prefix=/home/vivek/downloads/install/asterisk/
But still while running 'make install', it tries to install it in
/var/lib/asterisk/ and stops because of failing permissions.
I have provided it
Hi,
I'm trying to use the SIPAddHeader application to add a header containing to
semicolon separated strings like this:
exten = 12, 1, SIPAddHeader(X-TestHeader:a=test1;b=test2)
But in the resulting INVITE message only the first part
(X-TestHeader:a=test1) is added. Setting into quotation mark
I've been trying the SJPhone with no luck. Where did you download the
Xten version from?
On 1/16/07, Anton Krall [EMAIL PROTECTED] wrote:
Guys, anybody has seen or is using some kind of softphone on any square
screen device with WM5? Ive tried sjlabs one and xten for pocket pc and they
do work
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