Hi Florian
> Actually, I doubt the timing source will be required if you only use
> playback or background commands with the supplied gsm prompts. We run
> lots of machines without it.
>
> Timing sources are used for some cases of musiconhold, meetme and the
> likes, but not for regular stuff.
w
On Tue, Feb 27, 2007 at 08:01:41PM -0500, Jeronimo Romero wrote:
> Running Asterisk 1.2.9. I just installed a TE110P card and configured
> zaptel.conf & zapata.conf. The config files look right to me but I'm
> getting the following error when trying to start asterisk:
>
> Asterisk died with code 1
Hi Murf, Jason,
Steve Murphy wrote:
I have been testing asterisk 1.4 with a view to deploying it in my
organisation and I am experiencing jittery voice prompts from the voice
mail system. I get this jitter even if I try a simple "hello world" dial
plan.
What do you have installed, that will p
I see that you have signaling listed twice. That might be causing a problem.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Tuesday, February 27, 2007 6:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [ast
My asterisk install is showing the following every 1/2 second:
chan_sip.c:2739 auto_congest: Auto-congesting SIP/my.domain.net-0e118a90
There are lots of calls going through.
1. What can I do about this?
2. Is there a way to limit the number of calls (responding to invites with
no capacity
Hi Steven,
Thank you for your response. I had tried leaving endpoint phone number blank
but when I tried making an outside call, the Audiocodes seems doen't know where
to pass the call. So I need to assigned numbers. There is no problem for
incoming call aside from not being displayed the Calle
I have a high volume asterisk 1.40 installation and I ran a SIP SHOW
CHANNELS. (see partial output below). My questions are:
1. "wc-l" of the output shows 4000 lines. Does this mean 2000 active calls?
(2 channels per call)
2. The latter part of the output shows "unkn" for Form column. Why do
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetLanguage
There you can found how you can get the current language ( the same
used by playback ), so you can set a local variable to the current
language and use it instead of the blank value
Regards
On 2/26/07, kjcsb <[EMAIL PROTECTED
After switching to Asterisk 1.2.14 from 1.0.x, I encountered a very
strange problem. There is no sound with Playback() or Background()
commands.
Even though, Asterisk console shows the file is being played when I call
the extension (i.e. echo test), I can't hear anything.
My echo test exten
>Michelle Dupuis wrote:
> Isn't there a zap dummy (or something that uses the RTC) included in
> Asterisk 1.40 that creates the timing source? We don't install any external
> timing sources and we don't have choppyness problems on pure sip
> connections...
>
Yes, I have been looking into that af
Jeronimo Romero wrote:
Running Asterisk 1.2.9. I just installed a TE110P card and configured
zaptel.conf & zapata.conf. The config files look right to me but I'm
getting the following error when trying to start asterisk:
Asterisk died with code 1.
Automatically restarting Asterisk.
Does anyone
Hi:
This should be easy. I'm running 1.2.15.
When a caller calls someone's voice mail, it goes straight to a beep,
even though there is an unavail.wav file in that user's voice mail
directory.
Here is the relevant part of extensions.conf:
[internal]
exten => 2211,1,Dial(SIP/211,10)
exten => 221
Hello All,
For some reason my asterisk server is not registering a port number with
my VSPs. This is causing problems where people are not able to dial in
from any of my SIP or IAX VSPs.
I do have one VSP that has hard coded my IP and port so I can get
incoming calls but this still leaves
Running Asterisk 1.2.9. I just installed a TE110P card and configured
zaptel.conf & zapata.conf. The config files look right to me but I'm
getting the following error when trying to start asterisk:
Asterisk died with code 1.
Automatically restarting Asterisk.
Does anyone have any idea what is wro
Isn't there a zap dummy (or something that uses the RTC) included in
Asterisk 1.40 that creates the timing source? We don't install any external
timing sources and we don't have choppyness problems on pure sip
connections...
Jason - is this on a standard PC motherboard (or a mini device like Lin
Philipp Kempgen wrote:
John C. Wolosuk Jr. wrote:
Is there anyway to unset the extensions.conf definition of
writeprotect=yes while in the CLI interface (or by other mechanism) to
enable the dialplan save command? I accidentally overwrote my
extensions.conf but still have a running copy of
Philipp Kempgen wrote:
younss azzayani wrote:
hi after a many manipulation i get OK/YELLOW signal what does mean?
Don't manipulate. :-P
Regards,
Philipp
Start Asterisk and turn on the proper debugging.
Thanks,
Steve
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younss azzayani wrote on February 27, 2007 2:30 AM
> the cable is a simple cable break or: the cable schema we see bellow
1. If a piece of equipment such as the TE110P card is NOT seeing a T1
signal coming in, it will go into red alarm. That same piece of
equipment will then output on it's transmi
Steve Murphy wrote:
> On Wed, 2007-02-28 at 10:12 +1100, Jason Lewis wrote:
>> I have been testing asterisk 1.4 with a view to deploying it in my
>> organisation and I am experiencing jittery voice prompts from the voice
>> mail system. I get this jitter even if I try a simple "hello world" dial
>
On Wed, 2007-02-28 at 10:12 +1100, Jason Lewis wrote:
> Hi,
>
> I have been testing asterisk 1.4 with a view to deploying it in my
> organisation and I am experiencing jittery voice prompts from the voice
> mail system. I get this jitter even if I try a simple "hello world" dial
> plan.
>
> I hav
Anyone know any good reasons NOT to use the latest? I believe Bootrom
3.2.2 B and Firmware (SIP) 2.1.0. It's also the stuff the new IP 650's
are using.
I'm using 1.6.6 with no issues, besides the known call transfer thing.
I tried 2.X on a IP_601 and had trouble with the buddy-watch presence,
h
Anyone else experiencing a slow authentication command. I noticed this
command takes about 1.5 - 2 seconds of silence before it asked for password,
then another 2 sec of silence before it moves froward after that. Any ideas
I use Authentication regularly, no delay at all.
Post your dialplan ex
Hi,
I have been testing asterisk 1.4 with a view to deploying it in my
organisation and I am experiencing jittery voice prompts from the voice
mail system. I get this jitter even if I try a simple "hello world" dial
plan.
I have tried the release of 1.4 and also 1.4 svn and both display this
issu
younss azzayani wrote:
> hi after a many manipulation i get OK/YELLOW signal what does mean?
Don't manipulate. :-P
Regards,
Philipp
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
Asterisk? -> ht
I just struggled through the config on a Tenor AX. I'm not sure I can
help but I'll try. What do you need to do?
-Steve
FRANCISCO PEREZ-LANDAETA wrote:
Hi, just wondering if there is anyone that can help me configure my
quintum box to operate with asterisk. i have tried and made numerous
a
Hi, just wondering if there is anyone that can help me configure my quintum
box to operate with asterisk. i have tried and made numerous attemtps
configuring the tenor to work with [EMAIL PROTECTED] but have been unlucky.
anyone out there has a cheat sheet to configure this device.
thanks..
f
hi after a many manipulation i get OK/YELLOW signal what does mean?
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Hi Michelle,
actually, I didn't try it...
The server is a HP Proliant ML150T G3.
Currently I'm not in the condition to follow your suggestion, but I hope in the
near future to be able to give you a feedback.
Thanks!
Marco
> Have you tried starting Linux with irqpoll / noapic? Sounds like a
John C. Wolosuk Jr. wrote:
> Is there anyway to unset the extensions.conf definition of
> writeprotect=yes while in the CLI interface (or by other mechanism) to
> enable the dialplan save command? I accidentally overwrote my
> extensions.conf but still have a running copy of asterisk with the o
On Tue, 27 Feb 2007, Rob Schall wrote:
I wanted to try and see if I could get my Hawkings Net-Talk USB phone to
work with our asterisk setup via yakaphone. Has anyone ever tried this?
It sees the mic and speakers, but if we could get the keypad to talk
with yaka and in turn with asterisk, that w
On Tue, 27 Feb 2007, Michael Kamleitner wrote:
hi everybody,
I'm currently planning a small-sized web-applicaiton allowing users to
call-in via phone. the phonecalls should be recorded and processed further
by some custom scripts - sounds like asterisk is a perfect match for this
app.
however,
From: "Bala Neelakantan" <[EMAIL PROTECTED]>
Date: Tue, 27 Feb 2007 14:21:32 -0600
Looks like asterisk is receiving 202 while it is not expecting it.
/*! \brief Handle SIP response in dialogue */
/* XXX only called by handle_request */
static void handle_response(struct sip_pvt *p, int resp, cha
Is there anyway to unset the extensions.conf definition of
writeprotect=yes while in the CLI interface (or by other mechanism) to
enable the dialplan save command? I accidentally overwrote my
extensions.conf but still have a running copy of asterisk with the old
dial plan running in memory. whi
If re-invites are allowed then once both IAX endpoints are connected to
Asterisk and the call is active the server will attempt to step out of the
call. This actually works for both sip and IAX.
On 2/27/07, Joseph <[EMAIL PROTECTED] > wrote:
I find IAX connection with FWD very unreliable so I t
I find IAX connection with FWD very unreliable so I think I'll have to
roll out my own "SIP Express Router" as I want to communicate with few
SIP clients.
So I hope this the right solution.
I'm new to "SER" and to my understanding "SER is like a "road-map" it
tells the SIP Clients where they ar
Sip 1.5.2
Bootrom 3.1.3
Anyone know any good reasons NOT to use the latest? I believe Bootrom
3.2.2 B and Firmware (SIP) 2.1.0. It's also the stuff the new IP 650's
are using.
-Kenneth
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aster
T.38 pass-through should work fine on the SIP leg. (With Asterisk 1.40)
There are a few bugs but you can get past them.
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Octavarium
Sent: Tuesday, February 27, 2007 2:49 PM
To: Asterisk Users Mailing Lis
I am using 1.2.3
I get a 2 second pause before get the auth command from my primary dial plan
On 2/27/07, Matt <[EMAIL PROTECTED]> wrote:
What version of Asterisk are you running?
The 2 seconds of silence before it moves forward is probably because you
haven't set digittimeout and or you are n
Have you tried starting Linux with irqpoll / noapic? Sounds like a BIOS
bug..
MD
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marco
Parisotto
Sent: Tuesday, February 27, 2007 3:27 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] TE212P on FC6 - stack
Hi all
did anyone of you experience an error like "do_irq: stack
overflow" in configuring a TE212P on Fedora core 6? The server
immediately hangs, I don't know if this can be related to hardware
configuration or kernel incompatibility... This problem arises when I
try to configure the channels with
Hello all,
I added a record named pre_dst in the cdr table.
It has the same type as dst field.
And I used this line in the dialplan:
exten => _7.,1,Set(CDR(pre_dst)=${EXTEN:1})
When I call, 70123456, (7 is only to use the provider trunk),
I have this in the CLI:
Executing Set("SIP/foo-0816a490",
Looks like asterisk is receiving 202 while it is not expecting it.
/*! \brief Handle SIP response in dialogue */
/* XXX only called by handle_request */
static void handle_response(struct sip_pvt *p, int resp, char *rest, struct
sip_request *req, int ignore, int seqno)
Can you provide ethereal
What version of Asterisk are you running?
The 2 seconds of silence before it moves forward is probably because you
haven't set digittimeout and or you are not hitting # when you finish
entering your password.
What does your dialplan look like that is calling the authenticate command.
Please give
Yuan LIU wrote:
From: "Mojo with Horan & Company, LLC" <[EMAIL PROTECTED]>
Date: Tue, 27 Feb 2007 10:18:51 -0900
One thing I've noticed with SIP -> ZAP calls for quite some time is
that when asterisk is dialing n digits out the zap line, it dials n-1
digits, pauses for *TWO* seconds, and then
Mojo with Horan & Company, LLC wrote:
One thing I've noticed with SIP -> ZAP calls for quite some time is that
when asterisk is dialing n digits out the zap line, it dials n-1
digits, pauses for *TWO* seconds, and then sends the nth digit. Doesn't
matter how many numbers I want to send out th
On 2/15/07, Jordan Novak <[EMAIL PROTECTED]> wrote:
I have had a lot of complaints about the time it takes to setup a call. I
have timed it and it is almost five seconds before it even starts ringing.
The SIP device sends the request almost instantly but the channel is taking
a long time to pick
From: "Mojo with Horan & Company, LLC" <[EMAIL PROTECTED]>
Date: Tue, 27 Feb 2007 10:18:51 -0900
One thing I've noticed with SIP -> ZAP calls for quite some time is that
when asterisk is dialing n digits out the zap line, it dials n-1 digits,
pauses for *TWO* seconds, and then sends the nth di
What about the SIP leg?
- Mensaje Original -
De: "Michelle Dupuis" <[EMAIL PROTECTED]>
Para: "Asterisk Users Mailing List - Non-Commercial Discussion"
Enviados: martes 27 de febrero de 2007 16h'08 (GMT-0300)
America/Argentina/Buenos_Aires
Asunto: RE: [asterisk-users] H323-to-SIP proxy
hi dovid,
thx for replying, as I can see the chan_cellphone patch was done by you,
great! looks like this is exactly what I want. my goal is to connect a
normal consumer cellphone to the asterisk-server, allowing anyone else to
phone-in from their regular phone.
it would be even better if I coul
One thing I've noticed with SIP -> ZAP calls for quite some time is that
when asterisk is dialing n digits out the zap line, it dials n-1
digits, pauses for *TWO* seconds, and then sends the nth digit. Doesn't
matter how many numbers I want to send out the ZAP channel, this always
seems to ha
T.38 won't work over the H.323 leg of your call (even with Open H.323),
chan_h323 won't support it.
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Octavarium
Sent: Tuesday, February 27, 2007 12:51 PM
To: asterisk-users@lists.digium.com
Subject: [aste
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ICES
Lee Archer wrote:
I used mpg123 to stream air traffic control as a MOH class but I also
found it didn't always work with the shoutcast servers.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behal
Forrest Beck wrote:
I have Asterisk setup with two PRI's one going to my telco and the
other going to a Norstar Meridian system. The Norstar Meridian is
sending a BTN number that needs to be passed to the Telco. Is there a
way to pass the BTN as a variable in the dial plan? Like
CallerID(num)?
I wanted to try and see if I could get my Hawkings Net-Talk USB phone to
work with our asterisk setup via yakaphone. Has anyone ever tried this?
It sees the mic and speakers, but if we could get the keypad to talk
with yaka and in turn with asterisk, that would be really nice.
If there are any oth
Dovid B wrote:
Doug is this for the sip version or firmware ? As far as I know once
you go beyond a certain firmware version with polycom you cant go back.
Sip 1.5.2
Bootrom 3.1.3
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Temporary Safet
I need to receive a FAX call from a SIP device into my Asterisk box, then send
that FAX call to an H323 gateway and bridge the call, so Asterisk will be
acting as a Converter.
SIP device is a Grandstream HT496 so i can configure FAX Pass-through, but the
H323 gateway only supports T.38
BTW, i a
Dovid B wrote:
- Original Message - From: "Doug Lytle" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Tuesday, February 27, 2007 7:05 PM
Subject: Re: [asterisk-users] Polycom Firmware
Dovid B wrote:
Hi Guys,
A while back (several months ago
Hi im having this message in my console and dmesg.
rtc: lost some interrupts at 1024Hz
im not sure what this is.
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On Tue, 2007-02-27 at 19:14 +0200, Dovid B wrote:
> Doug is this for the sip version or firmware ? As far as I know once you go
> beyond a certain firmware version with polycom you cant go back.
>
> Dovid
We used bootrom version 2.6.1.
And yes, once you go to version 3.x, you cannot go back.
Fo
- Original Message -
From: "Doug Lytle" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Tuesday, February 27, 2007 7:05 PM
Subject: Re: [asterisk-users] Polycom Firmware
Dovid B wrote:
Hi Guys,
A while back (several months ago) I was having
What is the cellular connection for ? Are you using this for inbound or the
clients will call in in from thier cell phones ? If you need incoming (and or
ourgoing) lines you can get one from an ITSP. If you want to use your cell
phone you can use chan_cellphone. In order to use it you will need
Dovid B wrote:
Hi Guys,
A while back (several months ago) I was having issues with wmy
Polycom's and Asterisk. I was told to use a specific set of firmware
and sip version. I am unable to find that email. Anyone know which
ones work well with Asterisk ? (I believe it was 2.x )
I have yet to
Hi Guys,
A while back (several months ago) I was having issues with wmy Polycom's and
Asterisk. I was told to use a specific set of firmware and sip version. I am
unable to find that email. Anyone know which ones work well with Asterisk ? (I
believe it was 2.x )
Thanks,
Dovid__
hi everybody,
I'm currently planning a small-sized web-applicaiton allowing users to
call-in via phone. the phonecalls should be recorded and processed further
by some custom scripts - sounds like asterisk is a perfect match for this
app.
however, during prototyping I have no ISDN-connection wha
Dear Mike,
I had wanted to do something that is similar to your need as I wanted to be
able to add one active channel in multiple groups, it worked with The Ramon's
example in the link below which uses categories beside the set command, note
there are two examles depending on the asterisk version
Hi,
An observation on this feature, which I may have completely
misunderstood, so flame away if I am being dumb :)
Looking at the code, setting "limitonpeers=yes" causes all user and
peer calls to be ref-counted as if they are peer calls (assuming a
user and peer of the same name exist).
A side
Greetings Mike,
On Tue, 2007-02-27 at 11:28 -0500, Mike wrote:
> Ok, that sort of makes sense. But what I am doing is passing off a call
> into my Asterisk system to a cell phone. I want this to count as 2
> "channels". So, I am doing, in effect, this kind of algo:
>
> Answer the call
> Set(Gr
Ok, that sort of makes sense. But what I am doing is passing off a call
into my Asterisk system to a cell phone. I want this to count as 2
"channels". So, I am doing, in effect, this kind of algo:
Answer the call
Set(Group) to increment channel to 1
Play IVR, go into menus, etc.
Eventually go
Philipp Kempgen wrote:
Doug Lytle wrote:
Apart from that you assign the group 1234 twice to the *same*
channel. So GROUP_COUNT(1234) correctly reports only *1*
That would be it!
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Temporary Sa
Mike wrote:
Actually it wasn’t a straight paste. The straight cut and paste is:
exten => s,1,Set(GROUP()=${VAR})
exten => s,n,Set(GROUP()=${VAR})
exten => s,n,Noop(Used channels: ${GROUP_COUNT(${VAR})})
I've never tried using variables with GROUP(), but am guessing it's
permitted.
Try
Doug Lytle wrote:
> Mike wrote:
>> Hi,
>>
>> I was under the impression that Set(GROUP()=1234) incremented some
>> value associated with 1234.
>>
>> So if I did the same thing twice, I'd get a group count of 2.
>>
>> Ex:
>> exten => s,1,Set(GROUP()=1234)
>> exten => s,n,Set(GROUP()=1234)
>>
Hello,
I've enabled BT-200's SYSLOG logging, and I get some message whose meaning is
obscure to me. In particular, in a day I got the "Deletion of invalid timer"
message almost ten times from one phone, which has some call problems.
Can someone point me to a resource on BT200 error codes?
Thanks,
Actually it wasnt a straight paste. The straight cut and paste is:
exten => s,1,Set(GROUP()=${VAR})
exten => s,n,Set(GROUP()=${VAR})
exten => s,n,Noop(Used channels: ${GROUP_COUNT(${VAR})})
I believe that's good. But The group count is not "2", but "1". I thought
I'd be "2" since I called "S
I have Asterisk setup with two PRI's one going to my telco and the
other going to a Norstar Meridian system. The Norstar Meridian is
sending a BTN number that needs to be passed to the Telco. Is there a
way to pass the BTN as a variable in the dial plan? Like
CallerID(num)? What is the variabl
Hi,
Could someone double-check a behaviour I am seeing in 1.2 SVN HEAD
In sip.conf, create a type=friend entry with call-limit=1
1) Place an outbound call from the device
2) Place a call in to the device
"sip show inuse" is now something like:
* User name In use Limit
Mike wrote:
Hi,
I was under the impression that Set(GROUP()=1234) incremented some
value associated with 1234.
So if I did the same thing twice, I'd get a group count of 2.
Ex:
exten => s,1,Set(GROUP()=1234)
exten => s,n,Set(GROUP()=1234)
exten => s,n,Noop(Used channels: ${GROUP_COUNT(123
Hi,
I was under the impression that Set(GROUP()=1234) incremented some value
associated with 1234.
So if I did the same thing twice, I'd get a group count of 2.
Ex:
exten => s,1,Set(GROUP()=1234)
exten => s,n,Set(GROUP()=1234)
exten => s,n,Noop(Used channels: ${GROUP_COUNT(1234})
I get this
d addressee of this electronic message and
> its attachments, kindly delete it immediately from your system and
> notify the sender by electronic mail. You must not copy this message
> or attachment or disclose its content to any other person.
>
> Xplorium does not guarantee the integrity of this
Julian Lyndon-Smith wrote:
Given a choice, and a green-field site, would you
a) Have a separate network (switches etc) for your data and phone
b) Use the same network, but use VLAN's ??
What are the pro's and con's of each ?
TIA
Julian
___
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I've had a stable call center running since 2004, with only occasional
maintenance required. The only time it ever crashes is when I let it
fill up it's disks with call recordings.
I've had another system up and running since 2003 that hasn't hardly had
anything done to it.
Both of these sy
message
> or attachment or disclose its content to any other person.
>
> Xplorium does not guarantee the integrity of this electronic message
> and any of its attachments, or that they are free from computer
> viruses or other defects.
>
Hi,
I've found this doc helpful in configuring my iptables:
http://www.voip-info.org/wiki-Asterisk+firewall+rules
Following those settings, my devices register and function properly.
Alex
On 2/27/07, -- [ UxBoD ] -- <[EMAIL PROTECTED]> wrote:
I have this running on my Asterisk server, and ha
Lee Jenkins wrote:
kjcsb wrote:
The variable ${CONTEXT} stores the value of the current context.
However if we are in a macro that will be the name of the macro. How
do I access the name of the local channel's context.
For example:
[macro-test]
exten => s,n,NoOp(Context ${CONTEXT})
CLI sho
I have this running on my Asterisk server, and have opened up ports UDP/5060
and UDP/1-2 but for some reason when I try and connect too my SIP
extension it does not work. Are these the correct ports ?
--
--[ UxBoD ]--
// PGP Key: "curl -s http://www.splatnix.net/uxbod.asc | gpg --import
Given a choice, and a green-field site, would you
a) Have a separate network (switches etc) for your data and phone
b) Use the same network, but use VLAN's ??
What are the pro's and con's of each ?
TIA
Julian
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Philipp Kempgen wrote:
> tcpflow -c "tcp port 5038"
s/5038/8088/ :-)
Regards,
Philipp
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
Asterisk -> http://www.das-asterisk-buch.de
Geschäftsführer
[EMAIL PROTECTED] wrote:
>>> You probably need to do a GET, not HEAD, POST, PUT or something.
>
> The method is GET and with Firefox all work well
> i dont understand?
Somehow IE seems to send a different request. I'm not familiar
with jquery nor do I use IE so I can't tell if jquery or IE is
to
Anyone else experiencing a slow authentication command. I noticed this
command takes about 1.5 - 2 seconds of silence before it asked for password,
then another 2 sec of silence before it moves froward after that. Any ideas
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You probably need to do a GET, not HEAD, POST, PUT or something.
The method is GET and with Firefox all work well
i dont understand?
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Hello Users,
Good AfterNoon to all
I'm Mainly focused on OpenSER and Asterisk Integration.
I didn't Find any solution of My Question ?
Till now I'm doing only communicating OpenSER and Asterisk through SIP
Channel only.
User in Asterisk can Call to OpenSER and also vice-versa .
But My Ques
Dear All,
Please send the sip configuration for both phones along with a debug
from asterisk when you try to call from cisco to the eyebeam? also are
you trying to make them call peer to peer or not?
What I am suspecting is that there must be something mismatching when
the cisco phone tries to c
Hi Carlos,
Check out Asterisk LDAP authentication:
http://www.voip-info.org/wiki/index.php?page=Asterisk+LDAP
Greetz,
[EMAIL PROTECTED]
_
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Mohamed A.
Gombolaty
Gesendet: 27 February 2007 13:03
An: Asterisk Users Ma
Softphone Eyebeam v 1.5.2
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mohamed A.
Gombolaty
Sent: Tuesday, February 27, 2007 2:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco 7960
Dear Khaled,
What is the
Hi,
I'm using messagenet VoIP provider, I can make calls but I cannot
receive calls.
When I call my VoIP number the phone rings but when I pick up the call
drops and I get this message on Asterisk console:
*Forbidden - wrong password on authentication for INVITE to ...*
Is there anybody who kn
Dear Khaled,
What is the softphone u r using?
Thx
MAG
Khaled wrote:
> I am using firmware version pos3-07-500
> Kindly can you provide me with the basic configuration for cisco ip
> phone and asterisk config file
>
> *I have nat=never at my asterisk config file and nat enabled N0 at
> cisco p
Hi, i have a doubt about autentication in asterisk.
it's possible to integration the asterisk with the other server for
autentication, for example kerberos, ou other?
i want to implement asterisk in a department of university, but it's
necessary autentication by students, login and password for
Hello.
Take a look about function SIPPEER (asterisk -rx "show function
SIPPEER").
It helps how to use peer information.
Regards.
José Luis
El lun, 26-02-2007 a las 23:09 -0800, Yuan LIU escribió:
> >From: kjcsb <[EMAIL PROTECTED]>
> >Date: Mon, 26 Feb 2007 22:32:29 -0800 (PST)
> >
> > >> CL
On 2/27/07, Steve Davies <[EMAIL PROTECTED]> wrote:
Thanks for all of the pointers on this - I think merging the
"limitonpeers" change from trunk into 1.2.15 is my favourite option
right now. Or should I just take chan_sip.c from trunk? Would that be
fairly safe?
Err... What I meant was shall I
On 2/24/07, Pavel Jezek <[EMAIL PROTECTED]> wrote:
Brian Capouch wrote:
>
> But the included comments say, "The user part of a type=friend call
> will still be affected by the call limit"
>
> Those seem to be in conflict, but perhaps it's just my parser :-)
> Could someone clueful explain?
>
>
I am using firmware version pos3-07-500
Kindly can you provide me with the basic configuration for cisco ip phone
and asterisk config file
*I have nat=never at my asterisk config file and nat enabled N0 at cisco
phone
*I have an out bound proxy ip and port 5060 at cisco phone
*Voip control
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