Hi,
I have installed the Asterisk 1.4 package which support snmp
operation for Asterisk MIB. The Asterisk server is running smoothly
.Fine . But I can't perform any snmp operation in the Asterisk MIB.
Please let me know if any other files need to be configured so that I
can perform the
Hi All,
I'd like to know if anyone has figured out a way to be able to have
users logon/logoff manually from Cisco 79xx phones (with SIP firmware
loaded)?
Scenario is, user walks into office, sits at a random desk, and logs
onto the phone. The system would need to log them off of the last
Dean,
As you know Asterisk is primarily for telephony (yes we have fun with it
controlling our light's, rebooting servers etc). Video conferencing is a
completely different game. IMHO it does not make sense to build video
conferencing for asterisk since lots of people that need video
Hi,
I'm trying to connect to Telewest/Virgin Media with a TE110P using
asterisk 1.4.13/zaptel 1.4.6. No matter what I try, my span always
appears as
PRI span 1/0: Provisioned, Down, Active
My zapata.conf is currently
---
[channels]
echocancel=yes
:) well i was expecting this type of reply and have been preparing my mind
to do what you said. But need a little help. i have a lttle idea about how
remote provisioning works, we have to send the data using xml. and we can
make an application which can do it in any language. But i dont know how
I can look at adding a server-filter parameter to astmanproxy.users
(no promises on timescale though!) as I wrote the per-user filtering
in the first place.
My problem with astmanproxy at the moment is that I don't get any
responses from the maintainer (Dave at popvox?). I have a couple of
Most of thread snipped.
On 10/24/07, marcotasto [EMAIL PROTECTED] wrote:
Some days ago I've sent to David Rowe a little patch that preserves the echo
cancel
status between calls.
Surely this is only appropriate where you have a local analogue device
that is unchanging - If you retained the
Steve Davies wrote:
Most of thread snipped.
On 10/24/07, marcotasto [EMAIL PROTECTED] wrote:
Some days ago I've sent to David Rowe a little patch that preserves the echo
cancel
status between calls.
Surely this is only appropriate where you have a local analogue device
that is
You guys are scaring me! I'm building a 2950 with SAS RAID 5 on the new PERC
and it will have 2 TE420P's. I hope it works or my bacon will fry.
On 10/25/07, Joseph Begumisa [EMAIL PROTECTED] wrote:
Has anyone had any compatibility issues with a TE110P card installed
on a Dell Poweredge
Utterly untested, but here goes with the server-filtering parameter...
The attached patch should apply to version the 1.21 tarball cleanly,
and includes all my other changes which haven't made it into the main
astmanproxy code.
Please do feed-back on whether this works (it compiles :-) ).
Hi
I have a asterisk with many phones (type=friend)
When I issue the command sip reload some of the phones become unreachable
and they come back just after.
I guess that the sip.conf file is too big and asterisk takes too much time
reloading the entire file.
Is there a way to avoid this
I am trying to use Asterisk as the voicemail system of the TELCO where I work.
I wanna test with 2 mail boxes ( and later with a better machine/server I
hope try with 7 ).
How do I include in voicemail.conf the file with the mail boxes?, In a big
system like this,is better use text
Hello,
Quoting Digium Support:
The TE110P has been discontinued and replaced in our product lineup with
the TE120P, which features many overall improvements and does not suffer
from the HDLC Abort/Bad FCS problems that the TE110P did.
Better switch to TE120P,
On 10/25/07, David Kennedy [EMAIL
Um, yeah, the part you suggest is obvious.
What about having two different SIP accounts for each phone, I guess I
need to do one for inbound and one for outbound? Different extensions
or whatever.
Thanks,
Steve
Klaverstyn, David C wrote:
The way I would accomplish this is to have 2 Asterisk
Chris Bagnall wrote:
Our testing has yielded pretty good results. We had 10 simultaneous
calls with ulaw and quality was very good. We are pure VOIP also.
How many VMs were you running at the time, and what load were they under?
We've setups running between 3 and 5 VMs on a single box
On 10/24/07, Alan Lord [EMAIL PROTECTED] wrote:
And anyone who has echo problems with x100p or other analogue cards
should really give this a try. Most of the experiences I have read about
have been very positive. Mine also :-)
Any in case anyone's wondering if it's too CPU intensive for
Hi, thanks for the quick reply
I've literally just got off the phone with a Telewest engineer - after
being told the line was ok yesterday, I've been told it wasn't
actually turned on. D'oh!
If need be I do have a spare TE120P in a box in my desk drawer, so I
can send that to Manchester to be
On Thursday 25 October 2007 15:36:44 Admin DeryTelecom wrote:
Hi
I have a asterisk with many phones (type=friend)
When I issue the command sip reload some of the phones become unreachable
and they come back just after.
I guess that the sip.conf file is too big and asterisk takes too much
Will All inbound calls go to SIP phones on the conversion server?
If so, you just need a .X extension that forwards all inbound calls to the
second server.
But, thinking about this, it would appear that the second conversion server
will need most of your dialplan, and as such may still
be
Any one have experience with this CISCO Wireless IP phone running with
Asterisk??
It doesn't support SIP protocol I believe, so I need to know if the
skinny channel can work with the 7921.
Thanks for help.
___
--Bandwidth and Colocation Provided by
I'm pretty sure, just like Voicepulse service, it won't work, but its
worth a shot, no? Its free...
http://sourceforge.net/projects/vgps/
On 10/24/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
Hi all,
I need a fully developed web based remote provisioning system. I cant find
anything reliable on
Hi Guys,
I Have this peers on my sip.conf
[provider-302333-3000]
type=friend
context=provider
secret=xpto
username=302000
host=sip.provider.com
fromuser=302000
insecure=very
canreinvite=no
[provider-30-3001]
type=friend
context=provider
secret=xpto
username=303001
I have a SIP voice server which I want to place an Asterisk server in
front of to handle call routing.
At the moment I can call the apps on the server fine, but it cannot
transfer to another extension via asterisk.
Even attempting a call from the voice server to an asterisk extension
On 10/25/07, Adrian Marsh [EMAIL PROTECTED] wrote:
I'd like to know if anyone has figured out a way to be able to have
users logon/logoff manually from Cisco 79xx phones (with SIP firmware
loaded)?
Scenario is, user walks into office, sits at a random desk, and logs
onto the phone. The
Hi Steve.
What I did was to allocate one EC instance the first time a channel asks for it
and reuse the same memory area for the same channel every time a new call is
coming.
The memory is then freed when the channel is unloaded.
I've done this with my TDM400P in mind and I don't know what
Frederico Madeira wrote:
I Have in my sip.conf two extension 3000 and 3001.
I have this rule in my extensions.conf
exten= _X.,1,Dial(SIP/[EMAIL PROTECTED](num)},60,Tt)
exten= _X.,2,Hangup
exten= _X.,1,Dial(SIP/[EMAIL PROTECTED](num)},60,Tt)
exten= _X.,2,Hangup
And every calls
Hello listers,
I went to pull some CDR's from my PBX, and noticed they were a bit
light. I also noticed output on the console about CDR's not being
posted. I am currently running 1.4.13, and in looking at the change
log, this was a change in behavior as part of mantis 10659. Personally,
Philipp
This didn't wotk.
Let's suppose that my sip extension 3000 want to call to (302).123.3211
I need a rule in extensions.conf to match with this number, right ?
So, I can't use rules that you advice.
My problem is only for outbound calls.
--
Frederico Madeira
[EMAIL PROTECTED]
On Thu, 25 Oct 2007, Frederico Madeira wrote:
Philipp
This didn't wotk.
Let's suppose that my sip extension 3000 want to call to (302).123.3211
I need a rule in extensions.conf to match with this number, right ?
So, I can't use rules that you advice.
My problem is only for outbound
Let's suppose that my sip extension 3000 want to call to (302).123.3211
I need a rule in extensions.conf to match with this number, right ?
Let me see if I have this correct. You want to use the
provider-302333-3000 for any call going out from 3000 and
provider-30-3001 for any call going
Someone posted to the wrong list, thought I would help out.
Thanks,
Steve
---BeginMessage---
I have heard that they have been problems with Realtime Queus in
Asterisk. What are the problems? Do they still exist? Do they exist
in the most current version of 1.2? Has anyone been able to use
Hi, How install chan_unicall.so in Asterisknow??
Thanks!
Cristian.___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Michael,
way cool.
Works in WINE also :)
db
On Wed, 2007-10-24 at 23:09 -0400, Michael Munger wrote:
Not sure if one exists, but someone had asked me for this a while ago.
Here it is! My Polycom Provisioning Tool. Notice the version is 0.0.1.
Just a concept program (but it works well).
I understand that Asterisk 1.4 should support T.38 pass-through, but I
need Asterisk (or something on the Asterisk box) to act as a T.38
endpoint. Judging from the unclaimed $12,000USD bounty, it doesn't
appear that Asterisk itself can do this.
Joseph Begumisa wrote:
Has anyone had any compatibility issues with a TE110P card installed on a
Dell Poweredge 1950? I noted the following error on the LCD display of the
Dell Poweredge 1950:
E1711 PCI PErr Slot 1 E171F PCIE Fatal Error B0 D4 F0.
The Dell hardware owners
Steve Totaro wrote:
Joseph Begumisa wrote:
Has anyone had any compatibility issues with a TE110P card installed
on a Dell Poweredge 1950? I noted the following error on the LCD
display of the Dell Poweredge 1950:
E1711 PCI PErr Slot 1 E171F PCIE Fatal Error B0 D4 F0.
The Dell
Jerry Geis wrote:
I have a box with a TE210P. Things work for a while then stop when
making call files.
I get NOANSWER as the return code (right away).
I am running asterisk 1.2.12.1, libpri 1.2.3 and zap 1.2.9.1
When I try to update to newer zaptel the machine locks when loading the
On Thursday 25 October 2007 07:40:06 Pepo wrote:
I am trying to use Asterisk as the voicemail system of the TELCO where I
work. I wanna test with 2 mail boxes ( and later with a better
machine/server I hope try with 7 ).
How do I include in voicemail.conf the file with the mail
Steve Totaro wrote:
Calling Digium. Post your /var/log/messages and /var/log/asterisk/full
(just anything that looks relevant).
Try a Sangoma card.
Or better yet, give us an opportunity to fix it. Sangoma cards have
problems too and I'm sure they have been going through a trial of fire
On Thursday 25 October 2007 10:36:02 Brett Crapser wrote:
[outbound]
exten= _X.,1,GotoIf([${CALLERID(num)} == 3000]?path0|1)
exten= _X.,1,GotoIf([${CALLERID(num)} = 3000]?path0,${EXTEN},1)
exten= _X.,2,GotoIf([${CALLERID(num)} == 3001]?path1|1)
exten= _X.,2,GotoIf([${CALLERID(num)} =
Brian Hutchinson wrote:
You guys are scaring me! I'm building a 2950 with SAS RAID 5 on the new PERC
and it will have 2 TE420P's. I hope it works or my bacon will fry.
You shouldn't see any problems with those boards. The 2950 is a common
environment. If I remember correctly, there used to
Rony Ron wrote:
Hello,
Quoting Digium Support:
The TE110P has been discontinued and replaced in our product lineup with
the TE120P, which features many overall improvements and does not suffer
from the HDLC Abort/Bad FCS problems that the TE110P did.
Although this is true ( :-) ) I think
On Thu, Oct 25, 2007 at 11:31:37AM -0500, Tilghman Lesher wrote:
On Thursday 25 October 2007 07:40:06 Pepo wrote:
I am trying to use Asterisk as the voicemail system of the TELCO where I
work. I wanna test with 2 mail boxes ( and later with a better
machine/server I hope try with 7
Hello
When using LookupCIDName, Asterisk 1.4 says that it's
deprecated, and we should use ${DB(cidname/${CALLERID(num)})}
instead, but I don't know how to use it:
;DEPRECATED exten = s,1,LookupCIDName
;ERROR
exten = s,1,${DB(cidname/${CALLERID(num)})}
I guess I should use this as a
Hi
I posted earlier about having issues connecting to Telewest's ISDN,
only to find out later Telewest had forgotten to turn it on -
hopefully I'm not having a similar silly problem.
My PRI span is now up and operational, but when I try to send a call
out over it, I just get congestion tones.
I'd like to grab the SIP response code that comes back from an INVITE. The
HANGUPCAUSE gives the converted ISDN cause code. Anyone know of a way to get
the SIP response code instead?
Doug.
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the
On Thu, 25 Oct 2007 18:46:19 +0200, Vincent
[EMAIL PROTECTED] wrote:
I guess I should use this as a parameter to a function, but which one?
Never mind, I found how to use it:
exten = s,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
___
Hi
While I have fixed the problem from this post, I do have another
problem, and you have asked for a debug output here, so I'll go
against my better instinct and reply here :)
-- Making new call for cr 32774
-- Requested transfer capability: 0x00 - SPEECH
[ 00 01 0e 06 08 02 00 06 05 04
David Kennedy wrote:
Hi
I posted earlier about having issues connecting to Telewest's ISDN,
only to find out later Telewest had forgotten to turn it on -
hopefully I'm not having a similar silly problem.
My PRI span is now up and operational, but when I try to send a call
out over it, I
David Kennedy wrote:
Hi
I posted earlier about having issues connecting to Telewest's ISDN,
only to find out later Telewest had forgotten to turn it on -
hopefully I'm not having a similar silly problem.
My PRI span is now up and operational, but when I try to send a call
out over it, I
Hi folks,
We have a problem here where users are calling a remote PBX and need to
use # and * to navigate it. We were using the Tt options in Dial() so
that we could later perhaps take advantage of this feature.
Features.conf's sections are fully commented out, so I wasn't expecting
the options
Matthew Fredrickson wrote:
Steve Totaro wrote:
Calling Digium. Post your /var/log/messages and /var/log/asterisk/full
(just anything that looks relevant).
Try a Sangoma card.
Or better yet, give us an opportunity to fix it. Sangoma cards have
problems too and I'm sure they
David Kennedy wrote:
Hi
While I have fixed the problem from this post, I do have another
problem, and you have asked for a debug output here, so I'll go
against my better instinct and reply here :)
I just looked through your debug and can't see any obvious problems.
It's likely you'll need
The thing I'm trying to get an answer on now is getting Dell or rPath to
tell me what I have to do to get the 256M battery backed up RAM going and if
I have to do anything special due to the SAS drives on the new PERC 5/i
controller!
I'm running AsteriskNOW to build my project while I wait for
Is there some part of the debug output I need to tell the telco about?
When I was on to them earlier today, the engineer only seemed to know
how to turn bits of their network on and off, not much about settings
I need my end etc.
Dave
On 10/25/07, Matthew Fredrickson [EMAIL PROTECTED] wrote:
Hi all. Newbie to the list, been using VOIP with Sipura Grandstream
hardphones for a few years, via a VOIP service provider (who I won't name
here). I haven't stepped up to running my own Asterisk box yet, because of
poor reliability of our Internet connection during non-business hours, but
I'm
Hi, in meantime if you have another type of digium pri
card you can plug it into your box to confirm that it's not related to
that card!
Better eliminate any doubt about that card... it made me suffer !
BR,
On 10/25/07, Matthew Fredrickson [EMAIL PROTECTED] wrote:
David Kennedy wrote:
Hi
Hi,
Is there a GUI for Asterisk 1.2 compiled from source or would I need to
upgrade to the 1.4 version to get the GUI that can be installed on
servers complied from source? Any help is appreciated.
Otis
___
--Bandwidth and Colocation Provided by
On Thu, Oct 25, 2007 at 01:46:53PM -0500, OCOSA ListAcct wrote:
Hi,
Is there a GUI for Asterisk 1.2 compiled from source or would I need to
upgrade to the 1.4 version to get the GUI that can be installed on
servers complied from source? Any help is appreciated.
asterisk-gui[tm] requires
I am trying to set up a Grandstream GXV-3000 Video
phone to Asterisk ver 1.2.21.1. The problem I'm
having is that it can call other SIP phones, but not
vice versa. Can someone tell me where is the problem?
TIA!
Here's part of my configurations:
--
sip.conf
--
; 113 is the
Thanks...
Tzafrir Cohen wrote:
On Thu, Oct 25, 2007 at 01:46:53PM -0500, OCOSA ListAcct wrote:
Hi,
Is there a GUI for Asterisk 1.2 compiled from source or would I need to
upgrade to the 1.4 version to get the GUI that can be installed on
servers complied from source? Any help is
Hi.
Check the codec allowing, disallow=all and allow=ulaw etc.
At 02:25 p.m. 25/10/2007, hin lee wrote:
I am trying to set up a Grandstream GXV-3000 Video
phone to Asterisk ver 1.2.21.1. The problem I'm
having is that it can call other SIP phones, but not
vice versa. Can someone tell me
Can you comment on the use of these phones with asterisk with the Skinny
images? I think you're talking about Cisco phones converted to using
the SIP image.
Moj
Alex Balashov wrote:
Roy,
While there is a difference in the feature set provided by the
SIP and Skinny images for the Cisco
Is your snippet from extensions.conf in the [default] context?
Are spaces OK around the '=' in sip.conf? They might be, just an idea.
Now I notice that for friend 112, you say codecs 'allow=all' and for
friend 113 you say 'allow=h263' -- maybe you need to explicitly allow
something like
Thanks for advices.
The last one from Tilghman fit better for my needs.
Thanks a lot.
--
Frederico Madeira
[EMAIL PROTECTED]
www.madeira.eng.br
2007/10/25, Tilghman Lesher [EMAIL PROTECTED]:
On Thursday 25 October 2007 10:36:02 Brett Crapser wrote:
[outbound]
exten=
On Thu, 2007-10-25 at 11:35 -0500, Tilghman Lesher wrote:
On Thursday 25 October 2007 10:36:02 Brett Crapser wrote:
[outbound]
exten= _X.,1,GotoIf([${CALLERID(num)} == 3000]?path0|1)
exten= _X.,1,GotoIf([${CALLERID(num)} = 3000]?path0,${EXTEN},1)
exten= _X.,2,GotoIf([${CALLERID(num)}
Hi list!
Is anyone using the Kirk IP600/3 with SIP firmware connected to Asterisk?
Any experiences / caveats?
If anyone would be willing to share the dump of their IP600 config file,
i would really appreciate it.
Is there anything special i should put in my asterisk config?
Thanks !!!
Remco
Steve Murphy wrote:
Although late model expression parser code allows ==, and treats it
like =,
and, and are interchangeable, and so also | and ||.
murf
late model is SVN TRUNK, 1.4, SVN BRANCH 1.4, or 1.2?
___
--Bandwidth and Colocation
Jordi wrote:
Any one have experience with this CISCO Wireless
IP phone running with Asterisk??
It doesn't support SIP protocol I believe, so I need
to know if the skinny channel can work with the 7921.
The 7921 works fine with SVN trunk, and I think the
trivial changes required to support
Thanks for the replies. Found my error, it's in the
extensions.conf. The word dial was misspelled. LOL!
Felt like a fool now.
--- Mojo with Horan Company, LLC
[EMAIL PROTECTED] wrote:
Is your snippet from extensions.conf in the
[default] context?
Are spaces OK around the '=' in
lol that's a good stumper :) I missed that too!
hin lee wrote:
Thanks for the replies. Found my error, it's in the
extensions.conf. The word dial was misspelled. LOL!
Felt like a fool now.
exten = 113,1,Dail(SIP/113)
___
That is correct. Our Cisco rep is sending us a 7911G and 7941G so we can
test with asterisk. We plan on converting them over to SIP for testing.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mojo with
Horan Company, LLC
Sent: Thursday, October 25,
On Thu, 2007-10-25 at 10:36 -0500, Brett Crapser wrote:
On Thu, 25 Oct 2007, Frederico Madeira wrote:
Philipp
This didn't wotk.
Let's suppose that my sip extension 3000 want to call to (302).123.3211
I need a rule in extensions.conf to match with this number, right ?
So, I can't use
Hi. I'm still a bit of a newb to linux, I see this in my messages log:
Asterisk init: Id ax respawning too fast: disabled for 5 minutes
What does this mean?
and how severe is it?
--
Anything else, let me know.
- Dominic
It is not the force of a stroke that makes fine art
On Thu, 2007-10-25 at 15:26 -0500, Eric ManxPower Wieling wrote:
Steve Murphy wrote:
Although late model expression parser code allows ==, and treats it
like =,
and, and are interchangeable, and so also | and ||.
murf
late model is SVN TRUNK, 1.4, SVN BRANCH 1.4, or 1.2?
Some days ago, I was looking for some mobility solutions...
My conclusion is Wi-Fi phones are growing up fast and I think it's only a
time question they became a standart for mobility in pbx, as well as pure IP
telephony. Even manufactures of DECT systems are preparing their products
line to
Does realtime work reliably on Asterisk 1.2.24?
Are there any definitive guides, I can only find bits and pieces here
and there. Any accurate howtos would be of great help.
I am missing func_realtime.so. Where does this file come from?
Asterisk or asterisk-addons? I saw in one of the
Hi,
I have downloaded the ast-ax-snmpd package in Ubuntu Edgy Eft (6.10)
kernel 2.6.17 ; Asterisk 1.2.12.1 server is already installed in
my system and its working fine . As per README file I have made
patching and after that when I issued the make command in Asterisk
its
Hello All
Has anyone integrated ARI with Asterisk 1.4 ? Is there any manual or steps
available ? Also let me know if someone know about any other similar
software.
Regards,
--
Kashif Naeem
Director
Hadi Telecom
www.haditelecom.com
Cell: +92 (0)345 4226006
Office: +92 (0)42 5692766
Email:
79 matches
Mail list logo
