the cable is a simple cable break or: the cable schema we see bellow
(1-4 2-5):
1---\ /---1
2-\ / \ /-2
3-/-\-/-\-3
4__/ / \ \4
5__ / \_5
6-6
7-7
8-8
kind regards
hi guys
i have created a plugin for jquery for asterisk ajax interfacement. the
interfacement work with ajam and on firefox work very well, the problem
is with IE :-(
an example:
the url is: asterisk/mxml
i want login on manager system and the string command is:
Hello,
i just installed asterisk 1.2.15. I got this error message. Somebody can
help me? Thank You.
Feb 27 11:47:43 NOTICE[17086] cdr.c: CDR simple logging enabled.
Feb 27 11:47:44 WARNING[17086] pbx.c: Already have an application 'Pickup'
Feb 27 11:47:44 WARNING[17086] loader.c:
Hi
I have cisco 7960 connected to asterisk ,using tftp xml config file,my
problem is it can receive any call but it cant call any extension.
Please can you send me ,how to solve this issue
Regards
Khaled Chehab
System Integration Engineer
Xplorium Offshore.
Sakiet Al Janzir
Postal
On Tue, 2007-02-27 at 11:52 +0200, Jonson Player wrote:
Hello,
i just installed asterisk 1.2.15. I got this error message. Somebody
can help me? Thank You.
Feb 27 11:47:43 NOTICE[17086] cdr.c: CDR simple logging enabled.
Feb 27 11:47:44 WARNING[17086] pbx.c: Already have an application
[EMAIL PROTECTED] wrote:
hi guys
i have created a plugin for jquery for asterisk ajax interfacement. the
interfacement work with ajam and on firefox work very well, the problem
is with IE :-(
an example:
the url is: asterisk/mxml
i want login on manager system and the string command
Hi:
I have a mgcp.conf and a mgcp_additional.conf which records the special
information about the extensions. And i found if i use ulaw in the general
context in mgcp.conf,then all the registered extensions can make both
outbound and inbound calls,the mgcp.conf is following:
[general]
port =
can you give a bit more info? I know that you need nat=never for example
- Original Message -
From: Khaled
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Cc: [EMAIL PROTECTED]
Sent: Tuesday, February 27, 2007 10:03 AM
Subject: [asterisk-users] Cisco 7960
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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
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 interpret
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
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)
CLI shows:
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
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 phone
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
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 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
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
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
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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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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[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
Philipp Kempgen wrote:
tcpflow -c tcp port 5038
s/5038/8088/ :-)
Regards,
Philipp
--
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Let's use IT to solve problems and not to create new ones.
Asterisk - http://www.das-asterisk-buch.de
Geschäftsführer:
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 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
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
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
-users
--
Thx
MAG
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Message: 4
Date: Tue, 27 Feb 2007 13:07:50 +0100
From: Giorgio
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
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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--
Message: 4
Date: Tue, 27 Feb 2007 13:07:50 +0100
From: Giorgio Incantalupo [EMAIL PROTECTED]
Subject: [asterisk-users] Forbidden - wrong password on authentication
for INVITE
To: Asterisk
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: ${GROUP_COUNT(1234})
I get this
Mike wrote:
Hi,
I was under the impression that Set(GROUP()=1234) incremented some
value associated with 1234.
So if I did the same thing twice, I'd get a group count of 2.
Ex:
exten = s,1,Set(GROUP()=1234)
exten = s,n,Set(GROUP()=1234)
exten = s,n,Noop(Used channels:
Hi,
Could someone double-check a behaviour I am seeing in 1.2 SVN HEAD
In sip.conf, create a type=friend entry with call-limit=1
1) Place an outbound call from the device
2) Place a call in to the device
sip show inuse is now something like:
* User name In use Limit
I have Asterisk setup with two PRI's one going to my telco and the
other going to a Norstar Meridian system. The Norstar Meridian is
sending a BTN number that needs to be passed to the Telco. Is there a
way to pass the BTN as a variable in the dial plan? Like
CallerID(num)? What is the
Actually it wasnt a straight paste. The straight cut and paste is:
exten = s,1,Set(GROUP()=${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 Set(group)
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,
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)
exten = s,n,Noop(Used
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
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
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
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(Group) to
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
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 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
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 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
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
- Original Message -
From: Doug Lytle [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, February 27, 2007 7:05 PM
Subject: Re: [asterisk-users] Polycom Firmware
Dovid B wrote:
Hi Guys,
A while back (several
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.
Found
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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Dovid B wrote:
- Original Message - From: Doug Lytle [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, February 27, 2007 7:05 PM
Subject: Re: [asterisk-users] Polycom Firmware
Dovid B wrote:
Hi Guys,
A
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
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
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
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
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
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:
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
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
What about the SIP leg?
- Mensaje Original -
De: Michelle Dupuis [EMAIL PROTECTED]
Para: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Enviados: martes 27 de febrero de 2007 16h'08 (GMT-0300)
America/Argentina/Buenos_Aires
Asunto: RE:
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 digit.
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
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 the
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
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
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
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,
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
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 -
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
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
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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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 are so
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
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.
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, char
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.
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
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 old
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
hi after a many manipulation i get OK/YELLOW signal what does mean?
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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..
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
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? -
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
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
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,
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 have tried
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
plan.
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
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
___
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
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
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
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
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 =
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
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 after
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