I'm looking for advise on troubleshooting QOS problems. After much
searching and reading online (Google, Voip-Info Wiki, etc.) I don't feel
any closer to finding the right tools to solve my problem. Any info you
would like to share would be much appreciated, and I'm sure the thread
will
On Mon, Sep 19, 2005 at 11:41:26AM -0400, Dean Collins wrote:
Anyway long story short there is a IRC module for xoops. But I was
thinking how cool would it be to have a skype module.
http://www.xoops.org/modules/newbb/viewtopic.php?topic_id=40632viewmode
=flatorder=ASCstart=10
Hi,
we have got same problem, sometimeit may depend from your telco due
to a bad data transmission synchronization.
Just leave your plug always inserted into your ISDN card (why should you
unplug it??).
Giorgio.
[EMAIL PROTECTED] wrote:
Hello,
I have hfc-pci card with zaphfc driver
http://www.asteriskguru.com/tutorials/sjphone_softphone.html
enjoy.
julien bossart wrote:
Hi all,
I am new to this forum.Say hello to all.
I need some help to make a example using sjphone with
asterisk (which will fonction as SIP server).
I use Fedora core 4, asterisk release version 1.2.
How about someplace central like South Africa?
--
Regards
Wayne Gemmell
Tel Fax: (011) 894-4081
Cell : 072 836 4325
Email : [EMAIL PROTECTED]
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OK, I solved the problem(s)
It was a completely wrong setup from the provider.
Now anything is OK
zttool shows OK in the alarms
asterisk works great, parameters in zaptel.conf are
span=1,0,0,ccs,hdb3,yellow
bchan=1-15 # set this to 1-15,17-31 for E1
dchan=16 # set this to 16 for E1
bchan=17-31
Chris Miller wrote:
I'm looking for advise on troubleshooting QOS problems. After much
Have a look at SineStatIAX:
http://www.sineapps.com/sinestatiax.php
--
Cheers,
Matt Riddell
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http://www.sineapps.com/news.php (Daily Asterisk News - html)
Hi,
On my FC3 box I am having *v1.0.9.
The problem is that when a user calls through POTS line and leaves a
message in voicemail, the channel doesn't detect the remote hangup.
After 10 seconds of remote hangup it plays messages like vm-thankyou,
vm-review etc as if user is still online.
Chris Miller wrote:
I'm seeing a number of these logged in full while my * system is idle,
but I haven't found a good description of what they mean. Can someone
oblige? I have a single SIP phone registered and an IAX trunk.
Sounds to me like the packets (ACKS maybe) are arriving late.
Can anyone please give advice how to make PTN calls that terminates on *
release immediately after call end?
It takes up to 3 min for a call to release on our server.
Thanks!
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.344 / Virus Database: 267.11.3/106 -
Does anyone have an example of how to use sipp and the matching
extensions.conf entries ?
Many thanks.
Julian
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I want to be able to dial a 'pager' extension from an phone on my
asterisk server, and have it ring all other extensions *except* the
extension from which I am calling (because call waiting is enabled on
most extensions by default) - effectively giving me the ability to
page all other
I 'm experiencing the same problem and another worst than ring
indicator, because i'm unable to call some numbers with automatic
response system. Calling these numbers with I4L gets always busy. If
you call with a cell phone, for example, the same number is free.
I googled and asked on
Hello,
I haven't solved following problem yet. I worry that:
CLI pri intense debug span
There is no any debug information. Does it give any idea about problem?
Baris Simsek wrote:
hi,
my asterisk version is 1.0.9
/etc/zaptel.conf
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
Is there any success in connecting Valiant E1 CB with Unicall to asterisk?
any help will be appreciated,
Paradise Dove
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Hi,
I've had a strange problem several times during the last days:
A call is established, both parties have audio in both directions,
but asterisk is still waiting for connect.
Thus after timeout (120secs) the call is terminated with either
busy or no answer.
This is annoying for the both
Assuming you can purchase online, just go to
voipsupply.com.
http://www.voipsupply.com/index.php?manufacturers_id=13
The switch between analog and digital makes a huge
difference to port density. With an analog
TDM card you can get 4 FXO/FXS ports per card.
With a digital T1PRI card, you
I just set
up a system with two ISDN pci cards and am using mISDN, plus chan_misdn
(multipoint only).
It seems to work fine except for a few annoyances, as I wrote in
another post.
I tried to ran chan_capi, afterwords, just to check on the difference
but had problems.
Of course, I did not
Hello all,
Updated versions of asterisk-oh323 are now available both for use with
Asterisk v1-0 (version 0.6.7) and Asterisk HEAD/v1-2 (version 0.7.3).
Download from the usual location:
http://www.inaccessnetworks.com/projects/asterisk-oh323
Regards,
Michael.
I dont want to start a RTFM thread, but can someone jsut clear this up for me.
In zapata.conf I have
;
; Zapata telephony interface
;
; Configuration file
[trunkgroups]
[channels]
language=en
context=incoming
signaling=v23
rxwink=300 ; Atlas seems to use long (250ms) winks
I have 3 locations I want to connect using (*) servers.
1 of those has a single BRI with a Siemens DECT PABX.
1 of those has two BRI's with 2 Siemens DECT PABX's, each serving a
different area.
1 of those has two BRI's and a 2 port Nova Compact PABX with DECT
First step would be to set up the
So the more reliable way to do QoS is with MAC adress and not on a port basis.
Am I right ?
Thanks for your help,
Adrien
On 9/19/05, Rich Adamson [EMAIL PROTECTED] wrote:
I know that SIP is using port 5060 for session initiation, but which port
does it use for audio ? is it dynamically
Yes, because then the MACs specified would be getting the QoS, not just
certain ports. This is how I set up my customers when they have QoS
available.
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-Adrien Laurent
-Sent: Tuesday, September 20, 2005
Ok... I asked a question a few months back about a 7960 that a user
claims to be shocking her in her ear from time to time. A few others
indicated they had similiar issues and alot of them seemed to stem from
power over ethernet. Here's what we've done... We replaced the phone,
ran two new
Have you done any testing to see if it made any difference what type of
trunk was being used?
Darren Wiebe
[EMAIL PROTECTED]
Ricardo Poppi wrote:
Hi all.
I´ve found a kind of solution (if we can call it this way...) and Im
reporting it here to help save some lives.
Editing into astcc.cgi I
Before I reinvent the wheel, is anyone implementing any monitoring of PRI
(or T1) Red or Yellow alarms?
I would like to get notified ASAP if this occurs. Or possibly automate the
fix since service zaptel reload seems to fix my random issue.
I was thinking of using tail of the full log file, but
But, if I have Xlite running on client PC and at the same time the
user is doing FTP, both service has the same QoS treatment?
Is there a way to differentiate these services besides the port?
Sebastian
On 9/20/05, Sherwood McGowan [EMAIL PROTECTED] wrote:
Yes, because then the MACs specified
Hello,
What could be the problem if [EMAIL PROTECTED] is not starting mpg123 even
though I did not touch the MOH-config files? There is no error message
in asterisk at debug/verbose level 9. It seems asterisk doesn´t even
launch mpg123, but it´s hard to say - maybe it launches it for 1 second
Then you'll have to make sure that other services are lower QoS. Past that,
find out what port XLITE uses and then QoS that port.
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-Sebastian Milioto
-Sent: Tuesday, September 20, 2005 9:50 AM
-To:
is it possible to use asterisk to do provisioning for a voip cable modem or an MTA device? If so how can this be done?
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So the more reliable way to do QoS is with MAC adress and not on a port basis.
Am I right ?
Thanks for your help,
Adrien
On 9/19/05, Rich Adamson [EMAIL PROTECTED] wrote:
I know that SIP is using port 5060 for session initiation, but which port
does it use for audio ? is it
I have a question about the Asterisk Application ICES. I've got Asterisk
setup to accept a phone call and call the ICES app which sends it to an
Icecast server.
exten = 1,1,SetGroup('stream')
exten = 1,2,GetGroupCount()
exten = 1,3,Ices('contrib/${GROUPCOUNT}-ices.xml')
exten = 1,4,Hangup
Hi there does any of
you use ip phones from cisco on asterisk and how is the quality of this
configuration ? i have to make a price of an asterisk server with 100 ip phones
but i need stable phones snom is nice but still i have trouble with echo on them
and budgetone is cheap and feels
I do recall your postings relative to this...
I've done some research and I found some people have had issues with
cell phone radiation locking up or rebooting a 7960. Has anyone else
experienced this? We tried removing her cell phone from the room and it
doesn't seem to make any
Hi I have set up an Asterisk System with One XLite Phone and when i call the trunk line or receive calls via a trunk line (FXO generic X100P) i'm getting one way Voice. I can hear the called party - but they cannot hear me...
Any ideas - is t a NAT issue or is it something to do with the generic
is your asterisk server outside of your internal network? If not then
nat should not be an issue and it would point to the X100P clone.
On 9/20/05, Mark D'Cruz [EMAIL PROTECTED] wrote:
Hi I have set up an Asterisk System with One XLite Phone and when i call the
trunk line or receive calls via
Hi there does any of you use ip phones from cisco on asterisk and how is the
quality of this
configuration ? i have to make a price of an asterisk
server with 100 ip phones but i need stable phones snom is nice but still i
have trouble with
echo on them and budgetone is cheap and feels
HooDaHek 0.6 has been released.
So soon, you say? Well, the best laid plans of mice and men...
Steven BerkHolz is a pretty sharp stick and said to me, Why don't you
have HooDaHek change the CallerID when it looks up the name in the
database on an incoming call? Much head smacketh ensued,
Nathan Pralle wrote:
HooDaHek 0.6 has been released.
So soon, you say? Well, the best laid plans of mice and men...
Steven BerkHolz is a pretty sharp stick and said to me, Why don't you
have HooDaHek change the CallerID when it looks up the name in the
database on an incoming call? Much
What is this sip-nat-helper thing, is there a website were we can get
some info on it, partly thinking as the question before was relating to
open source software, I would assume that I could download this thing.
Dan
On Wed, 14 Sep 2005 [EMAIL PROTECTED] wrote:
If you have a linux box, then
I'm having the same problem you had Frank, so I'm pleased you came up
with a fix. No luck for me yet!
Incoming outgoing calls work fine using X-Lite, I just cannot transfer.
It's the first time I've ventured in to features.conf so I'm likely
doing something silly. I'd be grateful if you could
Hi all,
I hate to ask such a simple question, but it has stumped me over the
past couple of days.
I have 2 asterisk servers connected to the house lan and also via a
crossover ethernet cable. The original purpose of the crossover was to
create a private lan for TDMoE.
I have a TE410P in
Paul wrote:
Nathan Pralle wrote:
HooDaHek 0.6 has been released.
snip
As always, information and download linkage available here:
http://www.nathanpralle.com/software/hoodahek.html
snip
Does that mean I could use it with no instant messaging? I would like to
have a local callerID
Some websites allow you to look up a phone number and return a name/address.
As a possible add-on to this, I have an agi script that looks up caller ID
information on a few of these websites.
It is written in C/C++.
Currently these scripts are limited to Dutch numbers, since those are
basically
I have a TDM card in a asterisk machine.
I found that once I used it to call out, the call status changed to
connected even the callee is still ring.
How could asterisk distinguish the ringing and connected in zap channel?
thanks.
___
--Bandwidth and
Ok.
I was sucessful in installing ODBC storage
I'm using MySQL in the backend as it is. but all my drivers are now ODBC.
I am running asterisk-cvs head as of last night 9/19/05
My question is this... the old voicemail.cgi script that allowed checking
voicemail no longer works etc, and never
Hi,
On Wed, Sep 14, 2005 at 04:53:54PM +0200, Roger Schreiter wrote:
I have some experience in sending SMSs using smsclient.
I call the german Vodafone SMSC (01722278020),
and smsclient takes approx 20 secs to send a SMS.
The hardware is an Sedlbauer ISDN card.
smsclient seems to be
when a call file is used to place a call FROM an agent the agent is flagged as
busy/unavail even
if the call is subsequently transfered.
call file has...Channel: AGENT/blah...
Any way to stop the agent channel being flagged as busy?
Cheers
__
I have used the pre20 package, with the latest CVS-head.
COmpile goes cleanly, NO ERRORS.
then I get this when I try to load asterisk
-cvv
[app_rxfax.so]Sep 20 14:00:23 WARNING[5924]: loader.c:325
__load_resource: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol:
Rene Kluwen wrote:
Some websites allow you to look up a phone number and return a name/address.
As a possible add-on to this, I have an agi script that looks up caller ID
information on a few of these websites.
It is written in C/C++.
I'm not aware of websites like this in the USA or other
On Tuesday 20 September 2005 15:10, Alexander Lopez wrote:
I have used the pre20 package, with the latest CVS-head. COmpile goes
cleanly, NO ERRORS.
then I get this when I try to load asterisk -cvv
[app_rxfax.so]Sep 20 14:00:23 WARNING[5924]: loader.c:325
__load_resource:
Yellowpages.com has a reverse lookup on it.
http://info.yellowpages.com/asp/partner/whitepages/reversephone.asp
As does whitepages:
http://www.whitepages.com/10001/reverse_phone
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nathan
Pralle
Alexander Lopez wrote:
I have used the pre20 package, with the latest CVS-head. COmpile goes
cleanly, NO ERRORS.
then I get this when I try to load asterisk -cvv
[app_rxfax.so]Sep 20 14:00:23 WARNING[5924]: loader.c:325
__load_resource: /usr/lib/asterisk/modules/app_rxfax.so:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
José Pablo Ezequiel Fernández
Sent: Tuesday, September 20, 2005 2:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] RxFax/TxFax - Compile Problem
On
Try
rm -rf /usr/include/spandsp*
rm -rf /usr/lib/libspandsp*
Then do a make install in the spandsp directory..
It may make you smile!
It made me!!
Alex
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Doug Lytle
Sent: Tuesday,
What version of libtiff are you using. Has anyone tried 3.7.x with spandsp?
Doug Lytle wrote:
Alexander Lopez wrote:
I have used the pre20 package, with the latest CVS-head. COmpile goes
cleanly, NO ERRORS.
then I get this when I try to load asterisk -cvv
[app_rxfax.so]Sep
Hi Sander,
Sander wrote:
Hi there does any of you use ip phones from cisco on asterisk and how is
the quality of this configuration ? i have to make a price of an
asterisk server with 100 ip phones but i need stable phones snom is nice
but still i have trouble with echo on them and budgetone
On Tue, 2005-09-20 at 14:31 -0400, Jonathan k. Creasy wrote:
Yellowpages.com has a reverse lookup on it.
http://info.yellowpages.com/asp/partner/whitepages/reversephone.asp
As does whitepages:
http://www.whitepages.com/10001/reverse_phone
Hi there,
Basically, youare supposed to play arround with indications.conf
To have the extensions configured with callprogress=yes but, be
carefull because it is quite experimental.
Also, what I did was to get an audio program (Cooledit, Adobe
audition, or other), and
use g711u for fax not 729
- Original Message -
From: [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, September 19, 2005 4:21 PM
Subject: [Asterisk-Users] T.38 Canreinvite (yes, again)
I know this has been
On 9/20/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Ok.
I was sucessful in installing ODBC storage
I'm using MySQL in the backend as it is. but all my drivers are now ODBC.
I am running asterisk-cvs head as of last night 9/19/05
My question is this... the old voicemail.cgi script
Michael Welter wrote:
What version of libtiff are you using. Has anyone tried 3.7.x with
spandsp?
I was running 3.7.2 without issues, but reverted to 3.5.7 because of issues I was trying to track down. Didn't do any better or worse then 3.5.7.
Doug
--
Ben Franklin quote:
Those who
On 20:38, Tue 20 Sep 05, Florian Overkamp wrote:
Hi Sander,
Sander wrote:
Hi there does any of you use ip phones from cisco on asterisk and how is
the quality of this configuration ? i have to make a price of an
asterisk server with 100 ip phones but i need stable phones snom is nice
(trimmed)
http://info.yellowpages.com/asp/partner/whitepages/reversephone.asp
http://www.whitepages.com/10001/reverse_phone
http://directory.google.com/Top/Reference/Directories/Address_and_Phone_Numbers/
and lets not forget google itself (residential only aparently)
phonebook:QUERY (smith, ca
Hi all,
it took a while, but on sourceforge.net I added the new release 0.6 of
chan_capi-cm driver.
Note: dial string and capi.conf has changed.
The main changes are:
- added 'relaxdtmf'.
- more BSD compatibility
- correct PROGRESS handling
- added verbose text for capi info/reason error
could you add it into cvs head?
thanks..
2005/9/20, Dan Littlejohn [EMAIL PROTECTED]:
On 9/20/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Ok.
I was sucessful in installing ODBC storage
I'm using MySQL in the backend as it is. but all my drivers are now ODBC.
I am running
1) how to config callprogress=yes ? in extensions.conf?
could you give me an example?
2) you means record the call (via zaptel) into a file and analyze it
with audio tool?
thanks..
2005/9/20, Alchaemist [EMAIL PROTECTED]:
Hi there,
Basically, youare supposed to play arround with
Figured it out. I didn't have tT in my dial command:
Dial(ZAP/1${ARG3},10,tT)
Thanks for posting your problem and solution. It sure helped me out...
Hugh
On 9/20/05, hugolivude [EMAIL PROTECTED] wrote:
I'm having the same problem you had Frank, so I'm pleased you came up
with a fix. No
Have you tested Aastra. Works great with * and reasoable pricing
Anders
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michiel van
Baak
Sent: den 20 september 2005 20:57
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Cisco Ip phones
On 20 Sep 2005, at 12:12, Baris Simsek wrote:Status: Provisioned, In Alarm, Down, Active Call your provider and ask them what they see. I guess they haven't enabled it yet.Tim.___
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OK Great, I'll give it a shot.
I did find this other option
http://archives.free.net.ph/message/20050309.013714.2d1bf446.en.html , but I
do not really want to imbed this info in the asterisk database if I can have
it external. (note: this other option did work when tested)
--
--
Steven
May
Matt Fredrickson wrote:
On Sun, Sep 18, 2005 at 11:32:00AM -0500, Brian Capouch wrote:
Senad J wrote:
If you are looking for the maximum number of cheap flights from around
the world, and plenty of convention and room space, the answer is Las
Vegas :-)
I would definitively agree!
Yes,
Hello,
I just bought a Snom-320 from ATAComm. I plugged it into my LAN,
registered it with *, etc. All my other SIP gear is Sipura and works
fine, both on the LAN and over the Internet.
The new Snom seems like it can't process the audio from the handset
mic. A steady tone is garbled, even
i checked the document about indicator.conf nd it is used to generator
the tone of busy, ringing, congestion or dialtone. Bt how can I detect
it in extension.conf?
I hope to know whether the callee is answered the call, or know the
duration of answered time. but even the callee doesnt picked the
On Tuesday 20 September 2005 15:36, Michael Welter wrote:
What version of libtiff are you using. Has anyone tried 3.7.x with
spandsp?
My setup:
tiff-3.7.3 *
spandsp-0.0.2_pre20 *
Asterisk HEAD with app_[rt]xfax-0.0.2_pre20
* These are Gentoo packages.
It compiled, it started, it worked,
Steven wrote:
I did find this other option
http://archives.free.net.ph/message/20050309.013714.2d1bf446.en.html , but I
do not really want to imbed this info in the asterisk database if I can have
it external. (note: this other option did work when tested)
Yeah, I tried that when I first
This kind of mistakes are very common, I made them myself a couple of times,
that's is why instead of going around removing and coping and symlinking
files I prefeer to use the packages:
emerge spandsp
would do the trick.
On Tuesday 20 September 2005 15:38, Alexander Lopez wrote:
Try
rm -rf
On 21:30, Tue 20 Sep 05, Anders Svensson wrote:
Have you tested Aastra. Works great with * and reasoable pricing
Nope, haven't seen any phone of them in real life yet.
Right now we deploy snom's for the price/quality rate they
deliver. I find them very stable and nice phones.
--
Michiel van
Anybody know anything about using Asterisk vertical service
activation codes as described in the wiki:
http://www.voip-info.org/tiki-index.php?page=Asterisk+vertical+service+activation+codes
Specifically I'm interested in *0 that (apparently) flashes an
external trunk on bridged channel.
I played around with finding the right place to call the agi.
Since my config started as [EMAIL PROTECTED], there are a lot of macros that
complicate things.
I put the agi in the macro-dial and it is working as expected. (just the
CLID record and change)
Thanks for the new tool.
ref:
Is there a guide anywhere which runs through how to set up asterisk with mysql?
I've looked and almost all the document misses out relevant information.
Thanks
Dan Journo
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I'm not expert on this matter,but base on experience that is a normal situation on SIP/IAX channels since the call made is answered by the other end regardless of the situation you might found.
I'm on PRI ISDN and for me dialstatus and hangupcause works pretty good.
Regards
Jb
On 9/15/05, Mark
Yes Darren. The problem is the same using Zap or SIP. I had no
oportunity to verify that using IAX or E1/T1.
Rgds, Ricardo Poppi.
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Hello,
On Tue, 20 Sep 2005, Darren Ellis wrote:
Hello,
I just bought a Snom-320 from ATAComm. I plugged it into my LAN,
registered it with *, etc. All my other SIP gear is Sipura and works
fine, both on the LAN and over the Internet.
The new Snom seems like it can't process the audio
We have tested this phone with a Asterisk system and deliver the phone with
pre installed SIP-firmware without License
What about the license?? And do you have to buy a license and changing the
phone to sip protocol looks scary :( and time consuming with 100 phones not
all suppliers will do it
Did you make any special configuration with the switch on the card? I have 2 TE400P that I haven't being able to use on 1 server.
jb
On 8/28/05, Asterisk [EMAIL PROTECTED] wrote:
I have 2 TE410P's and a TDM400P in same machine without issuesBart-Original Message-
From: [EMAIL
I have a snom 360 installed but the woman that is operating it complains
about it all the time i looked at it and sometimes when sh transfers a
phonecall it will just hang and stays in the phone the snom does not have
connection to the line you can only see the line is still there in the
display
On 9/20/05, Nathan Pralle [EMAIL PROTECTED] wrote:
database on an incoming call? Much head smacketh ensued, and as I made
Thou hast confused the present tense with the present participle.
Thou couldest have written smacketh head smartly but perchance it
is better to write there was much
Thou hast confused the present tense with the present participle.
Thou couldest have written smacketh head smartly but perchance it
is better to write there was much head-smacking and gnashing of
teeth in this case, if thou so desirest to express thyself in the old
tongue. The eth suffix is oft
On Tue, Sep 20, 2005 at 04:33:12PM -0400, Joan Bautista wrote:
Did you make any special configuration with the switch on the card? I have 2
TE400P that I haven't being able to use on 1 server.
IIRC, the T400Ps and E400Ps had a few problems with multiple cards together...
Unless you're
usecallerid=yes
hidecallerid=no
callerid=asreceived
usecallingpres=yes
callwaiting=no
callwaitingcallerid=yes
threewaycalling=no
transfer=no
cancallforward=yes
callreturn=no
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
group=1
callgroup=1
pickupgroup=1
immediate=no
busydetect=yes
Steven,
Do you think the below dialplan would be typical for almost any
[EMAIL PROTECTED] setup? If so, I'll add it as supplimental documentation
for HooDaHek for those wanting to use it on [EMAIL PROTECTED]
Thanks,
Nathan
Steven wrote:
I played around with finding the right place to
Dan Journo wrote:
Is there a guide anywhere which runs through how to set up asterisk with
mysql?
What, exactly, are you trying to do with MySQL and *?
Access MySQL from the DialPlan:
http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+MYSQL
CDR record keeping in MySQL:
Dan Journo wrote:
Is there a guide anywhere which runs through how to set up asterisk with
mysql?
I've looked and almost all the document misses out relevant information.
Thanks
Dan Journo
What do you want to do with mysql? Did you read on the wiki? There is
tons of info there.
On 22:28, Tue 20 Sep 05, Sander wrote:
We have tested this phone with a Asterisk system and deliver the phone with
pre installed SIP-firmware without License
What about the license?? And do you have to buy a license and changing the
phone to sip protocol looks scary :( and time consuming
You can always take a PCAP (Ethereal) trace from the phone's web page
and analyze it with the RTP Statistics tool in Ethereal. That should
give you a hint whats up with jitter Co.
CS
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Darren Ellis
Anyone know why I would be getting this error? All calls go
through without problem but I get the following message:
fixlocalprefix: Could not parse
/etc/asterisk/localprefixes.conf
___
--Bandwidth and Colocation sponsored by
Ive already set up the cdr mysql.
Now im trying to add realtime now but stuck on how to do it. those links didnt really help much. and the cli doesnt provide much info on what is going on.
any help would be appreciated.
Thanks
Dan
On 9/20/05, Nathan Pralle [EMAIL PROTECTED] wrote:
Dan Journo
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm
Sent: Monday, September 19, 2005 4:52 PM
To: Asterisk Users
Subject: Re: [Asterisk-Users] OT: Hardware Interrupts; Who is it?
CPU0 CPU1
0: 85 1703809954
I met same problem when dial via zap channel.
Does anyone know how to solve it?
thanks.
2005/9/15, Mark Edwards [EMAIL PROTECTED]:
Hi.
I'm dialling two numbers - one that's unobtainable, one that's busy.
${DIALSTATUS} is coming back ANSWER each time right before the channels hang
up.
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