I am receiving this message intermittently. It is happening
during call setup. My phone is registering correctly. I am also having this
problem between Asterisks.Any ideas where this comes from?
Jun 29 01:05:04 NOTICE[13192]: chan_iax2.c:1601
iax2_destroy: Avoiding IAX destroy deadlock
On Wed, 2006-06-28 at 23:00 -0500, Aaron Daniel wrote:
If anyone's interested, I've just put together a sip realtime patch,
figured anyone that uses realtime may want to have a look at it. The
patch basically takes the stuff asterisk updates (fullcontact, ipaddr,
port, regseconds, and
Hello,
I have setup an asterisk server with two EICON cards, a 4BRI and 2FX.
How do I know, which card is the first, so that I can setup capi.conf with the
right entries?
Thanks for your help,
Stefan
--
in-put GbR - Das Linux-Systemhaus
Hello,
I'm currently testing the SNOM softphone for one of our clients.
Is anyone on this list using this software on Windows 2000 as a normal user?
When we configure the softphone as an administrator and restart the software,
the configured values stay the same.
But when we configure it as a
Ronald Wiplinger wrote:
He want to use 100 phones at the same time!!!
Alas, he won't be able to.
Re: ASTCC in-use flag
You'll have to disable the in-use flag for his account.
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at
2006/6/28, Tzafrir Cohen [EMAIL PROTECTED]:
The absense of USB?Use kernel 2.6?--Tzafrir Cohensip:[EMAIL PROTECTED]icq#16849755
iax:[EMAIL PROTECTED]+972-50-7952406[EMAIL PROTECTED]http://www.xorcom.comHi,
There's no PCI slot expansion on Intel Macmini.It's possible to install Debian and Asterisk
Hi,
[EMAIL PROTECTED] wrote:
is the following zaptel.conf configuration correct for TDMoE used for
pri-cpe signalling - is this possible at all ?
I couldn't find an example...
Any kind of Zaptel signalling should be fine.
Check out http://www.voip-info.org/wiki/index.php?page=Asterisk+TDMoE
One quote from the Melb Asterisk users group - Trixbox is great if you
like learning things twice.
(once the gui way, then once the right way.)
PaulH
--
Paul Hales
Technical Manager
AsteriskIT
www.asteriskit.com.au
bus: 03 8320 8100
mob: 0434 673 529
shadowym wrote:
Just my opinion and
Hi,
Does some one experience the Sangoma A20X-ec series card that cant detect
the hangup tone?
I got * server 1.2.5 and running on Centos 4.3, I hock up to two PSTN lines
and each time some one call in and my phone delay 1-2 sec (this is Asterisk
delay nothing to do with Sangoma) and it rings on
On Thu, June 29, 2006 8:40, Stefan-Michael. Guenther (in-put GbR) said:
Hello,
I have setup an asterisk server with two EICON cards, a 4BRI and 2FX.
How do I know, which card is the first, so that I can setup capi.conf with
the
right entries?
Thanks for your help,
lspci should tell
hello
I have an asterisk server with a
te110pE1 digium card. the server is a hp ML370 3,2 Ghz 64bits,
1Mo L2 , 1Go Ram, 3 SCSI 73Go in raid5.
I want to use in the same machine the kannel SMSC.
i have no big trafic in the two gateway but I want to know if it generate a
performence problem
chan (Alpha Trilogies Networks) wrote:
Hi,
Does some one experience the Sangoma A20X-ec series card that cant detect
the hangup tone?
snip
[channels]
context = from-pstn3
switchtype = national
usecallerid=yes
hidecallerid=no
transfer=yes
echocancel = yes
echocancelwhenbridged = yes
Hi I am sending the results of my research to the list. Unfortunately any combination of hangupcause worked :(.But I also try this L option on other machine, on one of my Zap channels and this time L worked perfectly. The channel went to hangup state and Asterisk executed the DeadAGI.
So I guess
On Thu, 29 Jun 2006, Francesco Peeters wrote:
On Thu, June 29, 2006 8:40, Stefan-Michael. Guenther (in-put GbR) said:
Hello,
I have setup an asterisk server with two EICON cards, a 4BRI and 2FX.
How do I know, which card is the first, so that I can setup capi.conf with
the
right
hi,
Just wanted to inform everyone, if you're using the latest bristuff's
you might (depends on the country!) have hangup issues.
The issue appears every time you dial an external number, and hangup
after letting it ring for a few times. Then the remote party keeps
ringing.
In some situations
UT Starcom T1000. Ive tested it in a LAN environment and its
cheap and easy to configure, gives a great sound quality and the roaming behavior
is pretty correct
In a WAN env we havent tested it .
Regards
Josep
De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Alessio
2006/6/29, stoffell [EMAIL PROTECTED]:
I have emailed junghanns.net to let them know.Did they acknowledge the issue ?The issue appears every time you dial an external number, and hangup
after letting it ring for a few times.Is it really every time ?Cheers
Hello all,
I've been using * for quite some time and yesterday I decided to add
sipbroker to my config. It was pretty simple and it works for some
numbers (e.g. I can call *258-9123, UK date time - which is on the
phone numbers you can call page -) but fails for some others.
For
Hello all
i wondered what telephones should you recommend to use with asterisk,
sip compatible, that could use as many functions as possible, like any
modern digital phone with programable keys. It should have leds that
display who is busy at the moment, let transfer calls as simple as
possible,
Hi,
can anybody confirm if there are any patches required for Caller ID to
work on a Sangoma A200 card on a BT line in the UK?
With Asterisk 1.2.9.1
thanks
Mike
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On 28 Jun 2006, at 23:36, Corporate IT Solutions - Michael Dunne wrote:
If price is an issue, then Grandstream is the go.
If quality is the issue, then Snom or Cisco.
I like the elmeg 290 - nice feel to the phone and
not too expensive.
Tim Panton
[EMAIL PROTECTED]
On 6/29/06, Olivier [EMAIL PROTECTED] wrote:
I have emailed junghanns.net to let them know.
Did they acknowledge the issue ?
I didn't get any reply yet. (but I'm used to that ;))
But yes, the -q release CHANGES file contained this:
- libpri fix for P2P BRI in Belgium
But the bug still
On 29 Jun 2006, at 02:08, Aaron Daniel wrote:
Has anyone considered the idea of splitting the sip registration
information in a realtime database from the actual configuration of
the
peers?
I mean, instead of having a table full of the configuration
information
(i.e. name, regexten,
On 28 Jun 2006, at 19:50, Douglas Garstang wrote:
-Original Message-
From: Doug Lytle [mailto:[EMAIL PROTECTED]
Sent: Wednesday, June 28, 2006 12:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Standard Sound Files Distortion
Douglas
On 6/28/06, Forrest Beck [EMAIL PROTECTED] wrote:
So far we have a
Grandstream 2000
Cisco 7912
Very good phone but not so big display.
Polycom SoundPoint IP
What model? they recently released an alternative to the 501, being a
430. Looks promising.
And we are looking at getting a Linksys
Hi everyone,
I have Asterisk SVN-trunk-r7498 running for a few months and I'm quite
happy with it. However, I am experiencing a quality issue with my AVM
Fritz!card PCI which is used with chan_capi. When somebody calls me on
this line he hears a lot of noise and I hear scratches and plops. It
is
Hiya all,
I have had no end of trouble trying to get my A101 E1 card working on
a new asterisk installation. The sangoma tech people have ignored my emails
about this.
All the installation of wanpipe seems to go ok, and zaptel. it all installed
compiled and does all the wanpipe hwprobe exactly
Hi,
I have been using app_sms for a few weeks now, since I recently
upgrade to asterisk 1.2.9.1 (latest bristuff, -q) however, app_sms
doesn't seem to work that well anymore..
On receiving an sms, I execute the app_sms script, and get this as output:
-- Accepting voice call from '171701' to
On Thu, Jun 29, 2006 at 08:56:05AM +0200, Olivier wrote:
2006/6/28, Tzafrir Cohen [EMAIL PROTECTED]:
The absense of USB?
Use kernel 2.6?
--
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755 iax:[EMAIL PROTECTED]
+972-50-7952406
[EMAIL PROTECTED] http://www.xorcom.com
On Thu, 29 Jun 2006, Benjamin Sebbah wrote:
Hi everyone,
I have Asterisk SVN-trunk-r7498 running for a few months and I'm quite
happy with it. However, I am experiencing a quality issue with my AVM
Fritz!card PCI which is used with chan_capi. When somebody calls me on
this line he hears a
Hi @ all,
after installing and compiling Asterisk there is a strange error.
No sounds are played. There is a log entry, e.g. Playing 'vm-intro'
(language 'en') but nothing happened.
asterisk-sounds-1.0.9 is allready installed.
Can you help me?
Thanks and greets,
Boerni
Why did you set rx/txgain to 0.5 ? Most people use 0.8, but I
use no gain by setting it to 1.0, which works here good.
Does anyone know if you need to set rx/txgain to 0.0 to disable gain... or
it is a percent value...
DV
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On 6/29/06, Mark Ackroyd [EMAIL PROTECTED] wrote:
Jun 27 11:12:18 ERROR[10528]: chan_zap.c:10614 setup_zap: Unknown
signalling method 'pri_cpe'
Jun 27 11:12:18 ERROR[10528]: chan_zap.c:10239 setup_zap: Signalling
must be specified before any channels are.
Am I right in thinking that's it's
I thought that this was me going mad. I'm trying to use SVN trunk and
have exactly the same problems.
So, I think it's a bug.
Julian.
stoffell wrote:
Hi,
I have been using app_sms for a few weeks now, since I recently
upgrade to asterisk 1.2.9.1 (latest bristuff, -q) however, app_sms
- Original Message -
From: Armin Schindler [EMAIL PROTECTED]
Date: Thursday, June 29, 2006 11:48 am
Subject: Re: [Asterisk-Users] Very bad quality with AVM Fritz!card PCI
and chan_capi
On Thu, 29 Jun 2006, Benjamin Sebbah wrote:
Hi everyone,
I have Asterisk SVN-trunk-r7498
On 6/29/06, Julian Lyndon-Smith [EMAIL PROTECTED] wrote:
I thought that this was me going mad. I'm trying to use SVN trunk and
have exactly the same problems.
So, I think it's a bug.
Can you confirm sending out works fine?
I send out an SMS without any problem, on receiving however, I have
Hi,
I'm trying to make
an apparently simple thing work, but I don't see how it is possible with
Asterisk.
This is my
extensions.conf:
exten =
1234,1,Dial(SIP/123456/555-555-|20|D()) ;After call connects, send DTMF
exten =
1234,2,VoiceMail([EMAIL PROTECTED]);
What I
Hi,
I have tried and here it works fine (asterisk 1.2.1), with the following
configuration:
zapata.conf
context=testing
channel = 5
extensions.conf
[testing]
exten = s,1,Dial(ZAP/1/07XX)
from CLI:
-- Starting simple switch on 'Zap/5-1'
-- Executing Dial(Zap/5-1,
Hi,
my Asterisk records CDR logs in a MySQL table.
Is there anyone having a SQL query to find max load (max concurrent calls)
of my system?
Thanks in advance
--
Domenico Viggiani
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On Thu, 29 Jun 2006, Mimmus wrote:
Why did you set rx/txgain to 0.5 ? Most people use 0.8, but I
use no gain by setting it to 1.0, which works here good.
Does anyone know if you need to set rx/txgain to 0.0 to disable gain... or
it is a percent value...
It's percent. Meaning: gain=1.0
On Thu, 29 Jun 2006, Benjamin Sebbah wrote:
- Original Message -
From: Armin Schindler [EMAIL PROTECTED]
Date: Thursday, June 29, 2006 11:48 am
Subject: Re: [Asterisk-Users] Very bad quality with AVM Fritz!card PCI
and chan_capi
On Thu, 29 Jun 2006, Benjamin Sebbah wrote:
Hi
Do I still need an ATA adapter for my analog phones once I was able to connect my Siemens HiPath 3750 to Asterisk?Thanks in advance.On 6/27/06,
richard Coco [EMAIL PROTECTED] wrote:
hi all,The HG3550 V1 and HG3550v1.1 only supports H.323 V.2.I'am not sure but i thing that the feature CallerIDName
On Thu, 29 Jun 2006, Benjamin Sebbah wrote:
- Original Message -
From: Armin Schindler [EMAIL PROTECTED]
Date: Thursday, June 29, 2006 11:48 am
Subject: Re: [Asterisk-Users] Very bad quality with AVM
Fritz!card PCI
and chan_capi
On Thu, 29 Jun 2006, Benjamin Sebbah
Yeah, sending works fine.
Julian.
stoffell wrote:
On 6/29/06, Julian Lyndon-Smith [EMAIL PROTECTED] wrote:
I thought that this was me going mad. I'm trying to use SVN trunk and
have exactly the same problems.
So, I think it's a bug.
Can you confirm sending out works fine?
I send out an
Hello all,My Siemens PBX is hipath 3750, since HG3550 i think is applicable only to hipath 4000 for interfacing with asterisk,what do you think will I needing for asterisk and hipath 3750?Thanks.
Lito
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Could anybody post a working sip.cfg and phone1.cfg for the polycom IP501
thats working with the agent login.
Thanks,
Dean.
-Original Message-
From: Dean @ INKnBITs [mailto:[EMAIL PROTECTED]
Sent: 28 June 2006 17:25
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users]
That is possible, how could I check that?
I can see that IRQ 17 is shared between eth0 and my fritz!card but I
don't know if it changes anything:
Can you try it (or eth0) in a different slot (or change IRQ's in the
BIOS if possible) to see if it makes any difference? That's the only way
to
Has anyone here successfully tried this?hipath 3750 -- hg1500 -- asteriski'm not sure with the flowlines though.Thanks.Lito
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It's not in the right syntax. Debugging the console should display
that. It probably comes from my original message having the 'u' in the
front, sorry about that - was in a hurry typing.
For #1:
-
usg(2)[EMAIL PROTECTED] should be:
[EMAIL PROTECTED]|usg(2)
For #2:
-
[EMAIL
Check
out http://www.digium.com/en/ecosystem/partners/interoppartners.php
CS
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
RicardoSent: Thursday, June 29, 2006 11:16 AMTo:
asterisk-usersSubject: *** Spam *** [Asterisk-Users] recommended
telephones
Hello
Okay, that would make sense if you wanted 2 different volume levels for
the messages.
Just typically if the email attachment has low volume, usually the
message on the phone is low too.
In any case - you have 2 options now for adjusting volume. :-)
Aaron Daniel wrote:
The other problem is
That is possible, how could I check that?
I can see that IRQ 17 is shared between eth0 and my fritz!card
but I
don't know if it changes anything:
Can you try it (or eth0) in a different slot (or change IRQ's in the
BIOS if possible) to see if it makes any difference? That's the
only
Hi,
I am running * 1.2.9.1 on a server recording calls via MixMonitor. I
have recorded one call which according to the cdr logs was 40 minutes,
but the recording seems to stop after 22.
I know this problem was fixed ages ago, but has anyone else noticed
this? Any idea what could be
That is possible, how could I check that?
I can see that IRQ 17 is shared between eth0 and my fritz!card
but I
don't know if it changes anything:
Can you try it (or eth0) in a different slot (or change IRQ's in the
BIOS if possible) to see if it makes any difference? That's the
If you do not use USB then I would suggest to disable this in the bios.
Henk
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James Harper
Sent: donderdag 29 juni 2006 14:56
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE:
You seem to have 2 wctdm adapters. Can you swap one of them with the
fritz card?
James
If you do not use USB then I would suggest to disable this in the
bios.
Henk
I'll try the usb trick first, and then if it doesn't work I'll try to
swap one of the TDM400 with the fritz. But I
Hello List,
Can anyone here has a working configuration of any digium e1 card that is
connected to cisco 3800.
Any help will be appreciated.
THanks,
Lito
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-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Angelito Manansala ha scritto:
Hello List,
Can anyone here has a working configuration of any digium e1 card that is
connected to cisco 3800.
The problem is the router configuration... you need these setups to
try some configuration on the Linux
Thanks for your reply. here is my zapata.conf configuration
[trunkgroups]
[channels]
context=default
switchtype=national
signalling=pri_cpe
usecallerid=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
W2K had problems with Security (Surprising huh?) You may need to grant
write access for the user to the Folder where SNOM is installed. I don't
think SNOM is writing to the registry if so you will need to open
permissions up on those keys in the hive.
I just tried serving the files off Apache, port 80, no
change... Most parameters are taken into account by the phone, except for SIP
proxy and SIP registrar...
Coud someone post an excerpt from their 46xxsettings.txt
where I could see the format they use?
Thank you in advance,
Silviu
Hi,
I've just installed a beronet BNS40 on Asterisk 1.2.9.1. Everything
seems ok, asterisk gives no error (nothing inside logs) but the 4 led on
the back of the card (which is NOT connected to an ISDN line) are red
and flashingwhat does it mean? Is it not properly working or it
means the
Have you startet the asterisk allready?
When i boot my machine, and dont start the astersik, the LED's keep
flashing all day. (even when lines are connected) and
even if /etc/init/misdn_init has been startet
TIP: First connect all Lines/Phones to the card, then start asterisk.
not 100%
Well we do write to the registry... Sorry about that, but how would we
otherwise store the information that is needed for the phone?!
CS
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Alexander Lopez
Sent: Thursday, June 29, 2006 4:01 PM
To:
On Thu, 2006-06-29 at 10:04 +0100, Tim Panton wrote:
Yes, except, if I understand you correctly, you would also
need to write insert and update triggers on the view, so that
when asterisk writes to the compiled config, the correct changes
are applied to your separate tables.
That might limit
Sorry had to jump in. I had a similar problem with Mozilla.
Make sure the Users can write to the config file. I just made all the Users
an Administrator at the local machine from Local Users menu, and that fixes
write to issues.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
I have setup several Calling Queues, each setup
with RoundRobin strategy. When I call the queue, the first
member/agent phone rings. Great! I call it again, the second
member/agent rings??
I thought that was the RRMemory strategy, but it
seems RoundRobin is also doing it.
Anyone know what
Hello,
I have set pickupgroup and callgroup for zap, sip and iax2 devices.
Everything is working good with zap and sip and between these two.
Iax2 pickupgroup and callgroup seems to be broken. I cannot pickup a call
to IAX2 from SIP.
Is there somewhere a bug ?
I am running: Asterisk 1.2.9.1
Could anybody post a working sip.cfg and phone1.cfg for the polycom IP501
thats working with the agent login, I need to get this sorted to go live
next week. If anybody can share their experience or pointers.
Thanks,
Dean.
-Original Message-
From: Dean @ INKnBITs [mailto:[EMAIL
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Angelito Manansala ha scritto:
I noticed that when i reload chan_zap.so command there is a warning like
this:
== Parsing '/etc/asterisk/zapata.conf': Found
Jun 29 21:50:58 WARNING[739]: chan_zap.c:10886 setup_zap: Ignoring
switchtype
Jun 29
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Wednesday, June 28, 2006 7:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Realtime SIP Registrations
Has anyone considered the idea of splitting the sip registration
On Thu, 2006-06-29 at 08:39 +0200, Patrick wrote:
If I get some spare time I wouldn't mind playing around with the patch
for 1.2.9.1. Can you please stick that one on bugs.digium.com too.
I've uploaded the 1.2.9.1 patch as well. Let me know if you find
anything I did wrong (I'm not much of a
On Thu, 2006-06-29 at 09:15 -0600, Douglas Garstang wrote:
How about fixing realtime SIP so that multiple Asterisk boxes can reference
the same database?
Doug.
That's kinda what I'm hoping to work towards :)
--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL
Couldn't you also create a separate GPO that allows for Read-Only
permissions??
Just in case.
RandyW
SANS wrote:
Sorry had to jump in. I had a similar problem with Mozilla.
Make sure the Users can write to the config file. I just made all the Users
an Administrator at the local machine
A quick fix has been posted a while ago by Marcel van der Boom (in
libpri/q931.c), this works.
The fix works, but it creates another bug afaik. Once you apply the q31.c
ourcallstate/peercallstate patch as I call it, the line gets hung up
normally, but for some odd reason all the scripts in
I have set pickupgroup and callgroup for zap, sip and iax2 devices.
Everything is working good with zap and sip and between these two.
Iax2 pickupgroup and callgroup seems to be broken. I cannot
pickup a call to IAX2 from SIP.
Is there somewhere a bug ?
I am running: Asterisk 1.2.9.1
I too have noticed problems with Asterisk native sounds using ulaw on
Asterisk 1.2.9.1. Don't know about other versions but it seems to work
quite well in Astlinux 0.40. In theory, since I am using ulaw for SIP there
is no transcoding so it is a more efficient use of CPU resources and it
should
Hey everyone,
I was wondering if anyone is able to help me with a
solution.
I have a small office set up with GXP-2000 phones and the
one thing I cannot get to work is them being able to transfer a caller directly
to another persons voicemail.
If I have a dial tone (and not on a
Yes, You are.
Libpri$ make clean
Libpri$ make install
Zaptel$ make
Zaptel$ make install
Asterisk$ make
Asterisk$ make install
In that order. All should be well.
Ben Bawkon
Varion, Inc.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Ackroyd
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 29, 2006 9:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Realtime SIP Registrations
On Thu, 2006-06-29 at 09:15 -0600, Douglas Garstang wrote:
shadowym wrote:
I too have noticed problems with Asterisk native sounds using ulaw on
Asterisk 1.2.9.1. Don't know about other versions but it seems to work
quite well in Astlinux 0.40. In theory, since I am using ulaw for SIP there
is no transcoding so it is a more efficient use of CPU
I cannot explain that:
One of my users shows up in sip show peers as 654200/Elmit_Unl
I can set it back to 654200/654200 but it will change back to
654200/Elmit_Unl
Why?
bye
Ronald Wiplinger
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Hi Kai,
when I connect the ISDN line the LED is not blinking anymore. I think it
is working now.
Thanks.
Giorgio Incantalupo
Kai Ober wrote:
Have you startet the asterisk allready?
When i boot my machine, and dont start the astersik, the LED's keep
flashing all day. (even when lines are
Richard,
I ran into this problem today myself
I am using the latest trunk to take advantage of the Jingle support
(works nicely :) ) . But I need h.323 support as well. Any
suggestions or patches?
Thanks,
Chaim
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Hi,
I bought a 7905G Cisco IP Phone and want to connect to Asterisk with SIP
protocol, but can't find a way to download this protocol update from Cisco,
Can anyone please help me?
Support the SIP protocol also the XML applications that I can use with SCCP?
What is better, try to configure
What kinda phone is it? That shouldn't affect the actual calls to the
phone, I would expect.
On Thu, 2006-06-29 at 23:49 +0800, Ronald Wiplinger wrote:
I cannot explain that:
One of my users shows up in sip show peers as 654200/Elmit_Unl
I can set it back to 654200/654200 but it will
I can also add that this happens on em_w lines as well. I've had issues
where callers start getting dead air when dialing out. Talking with the
phone company the lines were in an off-hook state even though Asterisk hung
up the call. I done exactly as below where I hang up before the other party
Hi,
Am Donnerstag, 29. Juni 2006 09:46 schrieb Francesco Peeters:
On Thu, June 29, 2006 8:40, Stefan-Michael. Guenther (in-put GbR) said:
Hello,
I have setup an asterisk server with two EICON cards, a 4BRI and 2FX.
How do I know, which card is the first, so that I can setup capi.conf
I think lots of us know about it... We're just not sure how to go
about fixing it. :-(
I know it's been a thorn in my side since I started using Asterisk.
I would suspect that many of those saying works for me have never
actually tested their system in failure scenarios, or they are working
in a
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To get the SIP firmware for these phones, you need to buy a Cisco
SmartNet support contract (about $75 USD in the USA, though I've heard
rumors a Europe-only contract exists for about $10 USD.) You can
purchase one through most Cisco resellers. That will give you access to
Cisco's download site.
- Original Message -
From: Brian Capouch [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, June 16, 2006 6:47 AM
Subject: Re: [Asterisk-Users] Gumstix!
James Harper wrote:
http://www.gumstix.com
For a
What I did was modify sip to update the status on the sip friends in
realtime. Then via FAGI dial them directly with the data found in real-time.
(ie dial (SIP/[EMAIL PROTECTED]:5060) Of course you need to check the status
in realtime data before you dial. This allows MANY Asterisk servers
Welcome to my personal hell ! :)I'have been discussing this previously on the list and also with some digium staff: to my experience there is NO way to archieve a linear distribution of calls from a queue.I mean
When a call comes in first member of the queue is ring, then second, etcSubsequent
Hi,
I'm planning to develop a solution with SMS using Asterisk within
Portuguese PSTN landline.
Any one has made it before?
I'm looking for Telco's and details using Portugal Telecom landline.
Thanks in advance,
--
Best regards,
Marco Mouta
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can you elaborate on modify sip to update the status on the sip friends in realtimethanksOn 6/29/06, Doug G
[EMAIL PROTECTED] wrote:What I did was modify sip to update the status on the sip friends in realtime. Then via FAGI dial them directly with the data found in real-time. (ie dial (
Hope this could help,
Please note Inband DTMF won't work unless the codec is ulaw or alaw
(G711). Use out of band DTMF aka rfc2833 or info.
http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+dtmfmode
best regards,
Marco Mouta
ps.give me some feedback if it worked
On 6/29/06, Shane
The linear function helps me too. I've built
an extensive multi-queue technical support system strategy. Based on the
initial queue, ALL calls goes to Tier1 first. Then, if Tier1 does not get
the call (on the phone/away from desk), Tier2 should get it, so on, and so
forth.
In Tier1, the
Hi all,
i successfully connected our old PBX to an asterisk server with a
junghanns quadBRI, the quadBRI ports running in bri_cpe_ptmp mode
connected to the interal PBX ISDN ports.
Now i tried to turn it round as our PBX depends on it for some features
and changed one of the quadBRI ports to
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