Hi,
Does anybody know what information is used when a SIP call is authenticated by
IP address?
I'd guess it's the socket the call was received on but was wondering (or more
correctly hoping) it would be done on the address in the bottom Via header.
That way IP authentication could work
--- Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Mon, Oct 29, 2007 at 03:44:13PM +, Jeng Yu
wrote:
[Oct 29 12:03:09] WARNING[2073] chan_zap.c: Unable
to
specify channel 4: No such device or address
[Oct 29 12:03:09] ERROR[2073] chan_zap.c: Unable
to
open channel 4: No such device
Brian Hutchinson wrote:
The web site is Russian (Serbian I think). Company is Hybird Systems
(Hibridni System AD). Best I can tell which probably does not help much
except to say it is a legit company that has been around a long time
making computer stuff since the 60's.
Here is original
Ditto what Lyle said. It is in the D channel. Your zaptel zapata configs
tell Asterisk what signaling you are using so look there. Google PRI or
ISDN and zaptel to see example .conf files.
On 10/27/07, Michelle Dupuis [EMAIL PROTECTED] wrote:
Ok..so how would the CALLED and CALLERID ID be
Cool! I guess the other guys just repackage it. Good info. Would be nice to
hear if anyone has used it. Looks like a nice phone.
On 10/30/07, Senad Jordanovic [EMAIL PROTECTED] wrote:
Brian Hutchinson wrote:
The web site is Russian (Serbian I think). Company is Hybird Systems
(Hibridni
On Mon, Oct 29, 2007 at 05:58:23PM -0700, Douglas Garstang wrote:
I'm trying to build an Asterisk rpm from the supplied asterisk.spec file.
Made numerous changes to get it to work.
That spec is known to be broken. Please use an alternative packaging.
--
Tzafrir Cohen
On Tue, Oct 30, 2007 at 07:31:07AM +, Jeng Yu wrote:
--- Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Mon, Oct 29, 2007 at 03:44:13PM +, Jeng Yu
wrote:
[Oct 29 12:03:09] WARNING[2073] chan_zap.c: Unable
to
specify channel 4: No such device or address
[Oct 29 12:03:09]
Of course, use the codec for the pentium 4!!
bilal ghayyad a écrit :
Dear Marc;
Thanks a lot for your kindly help.
My output of the command cat /proc/cpuinfo is:
[EMAIL PROTECTED] /]# cat /proc/cpuinfo
processor : 0
vendor_id : GenuineIntel
cpu family : 15
model
Gordon Henderson schrieb:
On Fri, 26 Oct 2007, Zoa wrote:
I would stay with DECT, the battery in WIFI devices only lasts a couple
of hours. (Unless you want to take the phone with you and use it on
public hotspots etc)
The battery in my UT Starcom F1000G lasts several days, as does the
I can confirm that this problem occurs in the latest svn version (revision
87498) as well.
What's the best way to work with the developers to have this tracked down and
addressed? I assume it needs to become a bug report.
___
--Bandwidth and Colocation
Hi there !
So it is possible to have logged a single unanswered channels ?
What sould be stted into the cdr.conf ?
Regards Andrea
Steve Murphy ha scritto:
Sorry!
I've gotten some complaints on this; I will try this week to
mod 1.4 so that you can choose to see the single-channel unanswered
Douglas Garstang wrote:
I'm confused about something.
It's the way Asterisk handles the A leg (ie the first party dialed) on
an originate command via the Manager Interface.
Lets say our originate commands looks like this:
ACTION: Originate
Async: yes
Timeout: 6
Exten: callback
I'm trying to load ztdummy on my Asterisk, running in a XEN
domain.
I've modified the code to disable the use of an RTC.
I can load the zaptel module just fine, the ztdummy also
loads without problem. But when running ztcfg I get this
error.
- s n i p -
graham:~# ztcfg -
Zaptel
You have to revert patch for issue http://bugs.digium.com/view.php?id=10659
More specific - that is - remove those two lines from main/cdr.c
if (cdr-disposition AST_CDR_ANSWERED
(ast_strlen_zero(cdr-channel) || ast_strlen_zero(cdr-dstchannel)))
continue; /* people don't want to see
On Tue, Oct 30, 2007 at 11:23:32AM +0100, Turbo Fredriksson wrote:
I'm trying to load ztdummy on my Asterisk, running in a XEN
domain.
I've modified the code to disable the use of an RTC.
I can load the zaptel module just fine, the ztdummy also
loads without problem. But when running
Dear all
This is survey of Digium Vs Sangoma Hardware i am going to
purchase some Asterisk supported hardware and i have confusion between both
company hardware performance i have read mailing list there is some text
sangoma has better performance then Digium E1 card so which
Dear all
I have plan to implementation of Asterisk PBX in my organization
now i have Read some doucment and we know Asterisk is not SIP proxy but it can
handle SIP call now I have 150 SIP Device in current setup and i have
Intel(R) Pentium(R) D CPU 3.40GHz + 2GB RAM + TE120P so
Tzafrir == Tzafrir Cohen [EMAIL PROTECTED] writes:
Tzafrir Take a look at /proc/zaptel/1
It's empty.
Tzafrir Any chance that this is ztdummy ?
It is.
Tzafrir You don't need a span line (or running ztcfg at all) for
Tzafrir ztdummy.
Ah! Doh. That isn't in any documentation
--- Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Oct 30, 2007 at 07:31:07AM +, Jeng Yu
wrote:
--- Tzafrir Cohen [EMAIL PROTECTED]
wrote:
On Mon, Oct 29, 2007 at 03:44:13PM +, Jeng
Yu
wrote:
[Oct 29 12:03:09] WARNING[2073] chan_zap.c:
Unable
to
specify
I have a repeated error in the asterisk console.
Incoming call: Got SIP response 400 Bad Request back from 10.0.2.136
It is repeated every few seconds and never stops until I restart
Asterisk.
It starts after I make a call to a certain extension, 202 which
transfers the call to another
On Tue, Oct 30, 2007 at 12:08:08PM +, Jeng Yu wrote:
--- Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Oct 30, 2007 at 07:31:07AM +, Jeng Yu
wrote:
--- Tzafrir Cohen [EMAIL PROTECTED]
wrote:
On Mon, Oct 29, 2007 at 03:44:13PM +, Jeng
Yu
wrote:
On Tue, Oct 30, 2007 at 04:49:25AM -0700, satish patel wrote:
Dear all
This is survey of Digium Vs Sangoma Hardware i am
going to purchase some Asterisk supported hardware and i have
confusion between both company hardware performance i have read
mailing list there is
Tzafrir Cohen wrote:
On Tue, Oct 30, 2007 at 04:49:25AM -0700, satish patel wrote:
BTW: vi beats emacs.
It would appear that you think the list has been a little too quiet
lately *smirk*
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little
Tzafrir Cohen wrote:
On Tue, Oct 30, 2007 at 04:49:25AM -0700, satish patel wrote:
Dear all
This is survey of Digium Vs Sangoma Hardware i am
going to purchase some Asterisk supported hardware and i have
confusion between both company hardware performance i have read
I have the same problem.
I trying with more 4 SIP providers, the account is registering, receive
inboud calls, but can`t make outbound calls for congestion.
Can be the out call id the problem?
Thanks
Gabriel
- Original Message -
From: [EMAIL PROTECTED]
To: Asterisk Users Mailing List
Michael Graves wrote:
On Mon, 29 Oct 2007 15:01:38 -0400, [EMAIL PROTECTED] wrote:
Well, just general office use. They are a real-state construction
company, so the phones will get some heavy use since most of the
phones are going to sales associates.
Now, one of the things we are most
Douglas Garstang wrote:
[LegA]
exten = _X.,1,Dial(SIP/[EMAIL PROTECTED])
exten = _X.,n,Playback(tt-monkeys)
I wanted to have control over the call both before and after it is
placed. I wanted to be able to play a prompt to the caller before the
call is placed to the destination number.
I am using the same enpression and all thing well except sound.
asterisk showing channel is spying but only problem is i am unable to listen
sound.
Rajeev.
On Monday 29 October 2007 12:47, [EMAIL PROTECTED]
wrote:
m: satish patel [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Chanspy or
Hi,
We are use IAX protocol between two asterisk servers.
Now we send information through this protocol by using EXTEN
We see that the variable EXTEN only holds 66 characters.
If we set a value larger then 66 characters, for example 70 characters.
The last 4 characters are cut off.
Hi again,
Could anyone recommend a good option for integrating Asterisk to any of
the following:
VoIP-PSTN Media Gateway
E1 cards for connection to telephony switches.
I am assuming the T1/E1 cards from digium are highly likely to be
Asterisk supported?
What about AudioCodes Media
I'm trying to compile chan_mobile for asterisk 1.4
I've installed 1.4 from SVN and downloaded addons from SVN also. I
make ./configure, make menuconfig, select only chan_mobile, and make.
Then I obtain the following errors. (I've also tryed applying the
patches I found at
Thank you very much !
Andrea
Atis Lezdins ha scritto:
You have to revert patch for issue http://bugs.digium.com/view.php?id=10659
More specific - that is - remove those two lines from main/cdr.c
if (cdr-disposition AST_CDR_ANSWERED
(ast_strlen_zero(cdr-channel) ||
Where is chan_unicall.so? Remember, Asterisknow haven't source code, and it
is necessary for install chan_unicall.so.
Idea??
Thanks!!
- Original Message -
From: Moises Silva [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
The version of chan_mobile in trunk is only compatible with the trunk
version of Asterisk, not the 1.4 branch. Whoever maintains the patches on
chan-mobile.org will not to update in order for you to compile the trunk
version of chan_mobile against Asterisk 1.4.
Sean
On 10/30/07, Alejandro
On Tuesday 30 October 2007 08:40:51 Arjan Kroon | Mobillion wrote:
We are use IAX protocol between two asterisk servers.
Now we send information through this protocol by using EXTEN
We see that the variable EXTEN only holds 66 characters.
If we set a value larger then 66 characters, for
On 8/29/07, Christian Peter [EMAIL PROTECTED] wrote:
Am Mittwoch, den 29.08.2007, 20:13 +0800 schrieb Steve Underwood:
Christian Peter wrote:
Hi list,
I'm running current SpanDSP
http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.4pre6.tgz
with Asterisk 1.2.22 somewhat
Steve Totaro wrote:
Tzafrir Cohen wrote:
On Tue, Oct 30, 2007 at 04:49:25AM -0700, satish patel wrote:
Dear all
This is survey of Digium Vs Sangoma Hardware i am
going to purchase some Asterisk supported hardware and i have
confusion between both company
I'm trying to setup zoiper ( formerly idefisk ) to use my asterisk
server at work from home.
I've setup zoiper for iax, set the ip address to work's fixed ip
address, user: home, password: password
but the zoiper log shows:
11:02:35 Rejected registration for '[EMAIL PROTECTED]' with
cause
On 10/30/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Oct 30, 2007 at 04:49:25AM -0700, satish patel wrote:
Dear all
This is survey of Digium Vs Sangoma Hardware i am
going to purchase some Asterisk supported hardware and i have
confusion between both company
On 10/30/07, Steve Davies [EMAIL PROTECTED] wrote:
Hi,
Did any progress occur on this? I too am using 0.0.4 spandsp snapshots
to receive faxes with quite a significant degree of success on a
1.2.24 asterisk box. I also get the only 8-bytes in a TIFF file
problem occasionally. Generally this
If you setup voicemail to allow them to hit * and then it jumps to
extension 'a' in the calling context, how do you see the original number
that called? If each user is going to have their own jump-to number for
'a', then I have to do a db lookup based on the called number to see
where to
I'm trying to setup zoiper ( formerly idefisk ) to use my asterisk
server at work from home.
I've setup zoiper for iax, set the ip address to work's fixed ip
address, user: home, password: password
but the zoiper log shows:
11:02:35 Rejected registration for 'home@my-office-ip-address' with
Tzafrir Cohen wrote:
On Tue, Oct 30, 2007 at 04:49:25AM -0700, satish patel wrote:
Dear all
This is survey of Digium Vs Sangoma Hardware i am
going to purchase some Asterisk supported hardware and i have
confusion between both company hardware performance i have read
Dear all
I have Asterisk which is connected with avaya through E1 back 2
back now i have on asterisk side G.711 codec and Avaya also useing G.711 codec
everything fine.
I need G.729 on my asterisk side. can i have lots of SIP phone
on my lan and issue is i have
Sorry for the double posting :(
sean
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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2007/7/5, gincantalupo [EMAIL PROTECTED]:
Hi,
I'm testing attended transfer with 3 SIP phones. I noticed about 10% of
my transfers make the call drop and I get this on my log:
Jul 5 10:42:32 WARNING[23960]: file.c:592 ast_readaudio_callback:
Failed to write frame
-- Playing 'beep'
Steve Davies wrote:
On 8/29/07, Christian Peter [EMAIL PROTECTED] wrote:
Am Mittwoch, den 29.08.2007, 20:13 +0800 schrieb Steve Underwood:
Christian Peter wrote:
Hi list,
I'm running current SpanDSP
http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.4pre6.tgz
with
On 10/30/07, Steve Underwood [EMAIL PROTECTED] wrote:
This was fixed a few weeks ago. There was an error in the FAX decoder,
but only a very few encoders create images that hit the issue. If you
try 0.0.4pre11 you will find it fixes several other quirky compatibility
issues. I have cleaned up
satish patel wrote:
i have download Free G.729 codec from digium and install on asterisk
and configure SIP device to use codec G.729 now when i make call from
asterisk to avaya i got this error message
Who said it was Free? You have to buy the $10 license per channel.
Andres.
sean darcy wrote:
I'm trying to setup zoiper ( formerly idefisk ) to use my asterisk
server at work from home.
I've setup zoiper for iax, set the ip address to work's fixed ip
address, user: home, password: password
but the zoiper log shows:
11:02:35 Rejected registration for
Steve Davies wrote:
On 10/30/07, Steve Underwood [EMAIL PROTECTED] wrote:
This was fixed a few weeks ago. There was an error in the FAX decoder,
but only a very few encoders create images that hit the issue. If you
try 0.0.4pre11 you will find it fixes several other quirky compatibility
On 10/30/07, Troy Ayers [EMAIL PROTECTED] wrote:
sean darcy wrote:
I'm trying to setup zoiper ( formerly idefisk ) to use my asterisk
server at work from home.
I've setup zoiper for iax, set the ip address to work's fixed ip
address, user: home, password: password
but the zoiper log
I don't have experience with Aterisk Now, but from the web page it
clearly states it is GPL, that means, code available. In any case, is
just matter of compiling a proper version of chan_unicall.so to match
the Asterisk version used in Asterisk Now. All information required is
here:
Well, my Zoiper works just fine with Asterisk 1.4.13 and this in the
iax.conf:
[fred]
type=friend
username=fred
secret=abc123
host=dynamic
context=default
NB. I have [fred] and username=fred
What does your configuration look like?
Mike
sean darcy wrote:
On 10/30/07,
Mike Tubby wrote:
Well, my Zoiper works just fine with Asterisk 1.4.13 and this in the
iax.conf:
[fred]
type=friend
username=fred
secret=abc123
host=dynamic
context=default
NB. I have [fred] and username=fred
What does your configuration look like?
Mike
I have patched chan_mobile into ast-addons 1.4.3 and have a patch file
for it. Nothing blows up, but I haven't had a chance to fully test this
yet.
Sean Bright wrote:
The version of chan_mobile in trunk is only compatible with the trunk
version of Asterisk, not the 1.4 branch. Whoever
Hi Jay,
On Tue, 2007-10-30 at 17:45 -0500, Jay Milk wrote:
I have patched chan_mobile into ast-addons 1.4.3 and have a patch file
for it. Nothing blows up, but I haven't had a chance to fully test this
yet.
I'm interested in trying it out on 1.4.13. Mind sharing that patch? If
you want to
A big G'day to everybody on the Asterisk list.
I am having a lot of trouble getting the TDM2400P card working in
asterisk. I will give the important details below, please let me know if
I am doing anything obvious or ideas for debugging this.
SUMMARY:
I get the right voltages on the line with
Anyone know if the MySQL() application has a configurable timeout?
If it tries to connect to a bogus IP, it's timeout seems to be a few minutes.
I'd like to cut it down to a few seconds.
Doug.
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail has
Alex R Green wrote:
A big G'day to everybody on the Asterisk list.
snip
signalling=fx0_ks
I don't know if this was cut and pasted from your zapata.conf or if it was just
a typo in this email, nor do I know if this is the breaking point of your
setup,
but signalling should be fxo_ks, not
Thanks Mark, that was just an email typo, it is fxo_ks in the
zapata.conf file. Any more suggestions? Thanks -alex
Mark Michelson wrote:
Alex R Green wrote:
A big G'day to everybody on the Asterisk list.
snip
signalling=fx0_ks
I don't know if this was cut and pasted from your zapata.conf
Douglas Garstang wrote:
Anyone know if the MySQL() application has a configurable timeout?
If it tries to connect to a bogus IP, it's timeout seems to be a few
minutes.
I never got a response on that question myself.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty
Alan Lord wrote:
Steve Davies wrote:
On 10/30/07, Steve Underwood [EMAIL PROTECTED] wrote:
This was fixed a few weeks ago. There was an error in the FAX decoder,
but only a very few encoders create images that hit the issue. If you
try 0.0.4pre11 you will find it fixes several other
Well, I'll bite and get the war going.
I'm putting in my first Sangoma card at the moment...so I have some current
experience.
The card installs great. Hardware compatibility is good (tried in a few
machines). Documentation on website is weak.
Tech support...mixed. I spent a lot of time
I have a T1 link from asterisk 1.2.23 (also tried with 1.4.13) to a Meridian
Option 61C. Calls either way drop with error Channel 0/23, span 1 got
hangup, cause 100. Can anyone offer insight into the cause and
solution/workaround? (I tried upgrading to Ast 1.4.13, and upgrading
matching zaptel
And just for confusion, one of the guys I work with swears by Sangoma. (I
have not done a lot of T1 stuff personally...so maybe as your expertise
grows Sangoma becomes a better fit).
Perhaps I should have voted for Heinz...
MD
-Original Message-
From: [EMAIL PROTECTED]
-- Processing IE 24 (cs0, Channel Identification)
-- Processing IE 28 (cs0, Facility)
Handle Q.932 ROSE Invoke component
-- Processing IE 108 (cs0, Calling Party Number)
The Meridian is trying to Invoke the Remote Operations Service Element
(ROSE). That is used to support interactive
I guess... it shouldn't be too hard to find the time out value in the source
and change it
- Original Message
From: Doug Lytle [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, October 30, 2007 5:23:35 PM
---
Jamshed Zaidi wants to stay in better touch using some of Google's coolest new
products.
If you already have Gmail or Google Talk, visit:
http://mail.google.com/mail/b-41f7263365-9f176e071a-d710297ddb46eb39
You'll need to
I have an Asterisk server running Elastix but patched to use Unicall.
Everything seems to be working fine and the TE220 card is up and running with
port 1 configured as PRI and port 2 as MFC/R2. We can already send and
receive calls on port two but we cannot on port one. That is when we
Hi all,
Recently, I have upgraded the asterisk as following.
asterisk-1.4.13
asterisk-addon-1.4.4
libpri-1.4.1
zaptel-1.4.5.1
Usage of the server: inbound and outbound call, queue, mixmonitor, meetme, moh
After upgrade, the server get segfault randomly and asterisk crash
and restart itself.
On Wed, Oct 31, 2007 at 12:06:25AM -0600, Carlos Chavez wrote:
I have an Asterisk server running Elastix but patched to use Unicall.
Everything seems to be working fine and the TE220 card is up and running with
port 1 configured as PRI and port 2 as MFC/R2. We can already send and
This happens when your pri line is down. This has happened a couple of times
to me and the pri commands come back when the pri line is up. My guess is
that your pri line is down.
Hope it helps.
Arpit
On 10/31/07, Carlos Chavez [EMAIL PROTECTED] wrote:
I have an Asterisk server running
On Wed, Oct 31, 2007 at 02:28:19AM -0400, Arpit Mehta wrote:
This happens when your pri line is down. This has happened a couple of times
to me and the pri commands come back when the pri line is up. My guess is
that your pri line is down.
Huh?
The 'pri' command is registered by chan_zap.so
Dear all
I have already post this question but i need more input for
this setup
[IPphone]--[Asterisk]E1---[Avaya]---[ip_Extention]
Asterisk - codec (G.711/ulaw)
Avaya - codec ( G.711/ulaw)
Now I need G.729 on my asterisk side and i have put G.729 codec setting on my
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