types for 'smp_send_reschedule'
> /lib/modules/2.4.21-4.ELsmp/build/include/asm/smp.h:41: error: previous> declaration of 'smp_send_reschedule' was here> make: *** [zaptel.o] Error 1>>> What could I be missing? :) Thank you
>> --> Leandro Rzezak> [E
ude/linux/smp.h:31: error: conflicting types for 'smp_send_reschedule'
/lib/modules/2.4.21-4.ELsmp/build/include/asm/smp.h:41: error: previous declaration of 'smp_send_reschedule' was here
make: *** [zaptel.o] Error 1
What could I be missing? :
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IP Phone <--g729--> VoIP provider
Please help me accomplish that.
Thank you
-- Leandro Rzezak[EMAIL PROTECTED]
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WE NEED:
IP Phone <--alaw--> IP Phone
IP Phone <--g729--> VoIP provider
Please help me accomplish that.
Thank you
-- Leandro Rzezak[EMAIL PROTECTED]
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To
o/${EXTEN},60,t)
On serverB
exten => _74XX,1,Dial(IAX2/campinas/${EXTEN},60,t)
Is there anything missing ???
Happy holidays to you all !!!
Leandro Martini
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:
70 61 73 73 77 64 69 61 78 79 31
0c:
00 00 00 05
Provisioning is 57 bytes
Total packet is 71 bytes
The device is working properly, I can ping it, but it does not allow to
provision it.
Is there anything missing to help solve this issue ?
Thanks in advance
Leandro Martini
Probably by preference and peer type matching, try setting a new VoIP peer
for inbound calls from asterisk
LTenorio
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Ivan Vershigora
> Sent: Thursday, November 03, 2005 10:27 AM
> To: asterisk-use
Are u serius?
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Jonathan k. Creasy
> Sent: Friday, October 28, 2005 9:49 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Geneys
>
> Anyone using the Gene
Any expert that configured Asterisk with Vontel in Argentina? Please give advice on the sip.conf configuration. Thank you!-- Leandro Rzezak[EMAIL PROTECTED]
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Edwin,
They are on the same VLan and on the same Subnet? If that's the case
check you switch log for details, if you havent changed anything on the *
Server. Looks like a serious package lost, even with a high segment this
shouldn't occur. At least for the info you send, these are the POF.
K, I'll make a page under wiki when I have my password back (I forgot it), I
saw a lot of msg like this one.
There are several ways to configure it, below is one for in/out.
BTW, In Cisco config's it's important to add security, to just let pass the
call from your asterisk, Qos, etc. (not
Just to receive a recommendation on switchtype for Argentina, Buenos Aires, 114816.
Thanks a lot-- Leandro Rzezak[EMAIL PROTECTED]
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: Friday, September 02, 2005 9:54 AM
To: Leandro Tenorio
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] oh323 or h323
LTenorio,
Then this is my problem. I can only register to the gatekeeper as a
terminal, they do not allow me to register as a gateway.
Is
details..
Protocol = H323
Gatekeeper = 210.21.118.xxx
H323ID = .HMA0200.10szxn-hxxx
e164 = 02022xx2912
H323ID = .HMA0200.10szxn-kxxx
e164 = 02022xx2913
Thanks for your help, it is much appreciated.
Kind Regards,
Steve Ducat.
On 9/1/05, Leandro Tenorio <[EMAIL PROTECTED]>
I'm using oh323 too without any issues,
but in Steve specific configuration, depends on how his provider expect to be
register as (Terminal or Gw) afaik, oh323 just could be binded as gateway, so
better ask the provider.
LTenorio
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Look for information on google there are several examples, 5800 it's
just like 5300 or 5350
LTenorio
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of kaws elchamal
Sent: Monday, August 29, 2005 3:26 PM
To: asterisk-dev@lists.digium.com; asterisk
Search for some of the configs for AS53XX out there. They are pretty the
same.
LTenorio
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of W. Kevin Hunt
Sent: Thursday, August 25, 2005 5:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subj
Nope, working as we expect.
Any more clues?, "unable to receive...Correctly" is not a easy to
understand tech. phrase.
LTenorio
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nir
Simionovich - CTO
Sent: Thursday, August 25, 2005 7:47 PM
I got it working, below it's the oh323.conf, I'd done
nothing in particular to make it work, just define it as node in the
gnugk
ListenAddress=0.0.0.0listenPort=1720
connectPort=1720
tcpStart=1tcpEnd=2udpStart=1udpEnd=2
fastStart=yes h245Tunnelling=yes
h245inSetup=yes
inB
rolaLara < AT+CLIP=1
[AG] MotorolaLara > OK
[AG] MotorolaLara < AT+CGMI
[AG] MotorolaLara > +CGMI: "Motorola CE, Copyright 2000"
[AG] MotorolaLara > OK
[AG] MotorolaLara < AT+CGMI
[AG] MotorolaLara > +CIEV: 5,5
[AG]
I'm a native Portuguese (European) speaker but I really haven't been
following this thread. Is there anything I can help out with?
(Reply directly to my email for a quicker response).
Regards,
Leandro
Paul Davidson wrote:
Unfortunately, I do not have the correct pronounciations-
s
interested in adquiring one/both of these drop me a line.
More info: http://www.nokia.com/nokia/0,8764,38621,00.html
Leandro
Juraj Bednar wrote:
>Hello,
>
>
>
>>There's this device called VoiceBlue GSM gateway.
>>It talks gsm on one side and SIP on the other side.
Hey Tim,
I'll be glad to help you out if I am able to.. but I honestly don't
recall which thread you are talking about. Maybe if you refresh my mind
and/or explain your problem?
Leandro
Tim King wrote:
I was reading a thread where you were helping someone out and noticed
it end
obile operator decide to say me. If I try to use a SIP phone (or a IAX
phone) attached to my asterisk box, I cannot ear the ringing from the
mobile operator nor any message the mobile operator decide to say me.
Any idea why?
Leandro
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] H323 vs OH323 GK registration issues
On Thu, Jun 23, 2005 at 11:35:38PM -0300, Leandro Tenorio wrote:
>
> IMHO, That's because both works on a different matter, while H323
> register itself as terminal (usually for phones and ATAs), OH323 on
> the other hand register itse
IMHO, That's because both works on a different matter, while H323
register itself as terminal (usually for phones and ATAs), OH323 on the
other hand register itself as gateway. In some switches you can change the
node type, I don't know in Nec Aspire. AFAIK there is no way to change the
wa
pi
0.3.5 to support relaxdtmf)
Question (I'm from a software eng. background, not telco):
What would be the reason for not receiving DTMF from a GSM
phone/gateway? Do you have the time to explain why? (I'm really
interested in learning :)
Thanks,
Leandro
I'm trying to use LookupCIDName to tag outgoing calls on my CDRs but it
seems that application only tags incoming calls?
Any sugestions?
Leandro
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Yes you can. There are some examples @ cisco look for TDM switching.
LTenorio
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marcelo
Pacheco
Sent: Wednesday, June 08, 2005 9:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
The configuration in the blog does not depend on the product, it
depend on the IOS used. Should work for your 5300, the only problem you
could have, AFAIR is with the SIP-ua config. Authentication, starts after
12.2.something.
If you have problem come back and I give u a workaround
asterisk channel
are you using? chan_capi? chan_misdn?
Thanks,
Leandro
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Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users]
oh-323 / Cisco AS5300 problem
Leandro,
I don´t have the licenses on porpuose, I shouldn't
need'em cause the phone I'm using has only g729 enabled, the oh323 has only
g729a enabled and the Cisco (over which I hav
The first error is probably because you don´t have licenses
for 729 and you are trancoding the audio.
The second is well dialed and you get from the 5300 a busy
message, the reason could be user busy (as the message saids), wrong dial peer
config, wrong dialing rules in 53xx.
LTenorio
Sure???
I think that most probably he still needs the CSU.
LTenorio
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke
Sent: Friday, May 13, 2005 7:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] P
I use 53xx with asterisk (both protos OH323 and SIP). It's runns smoothly.
Drop me an email if you need help.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Wednesday, May 11, 2005 2:09 PM
To: 'Asterisk Users Mailing List - Non-Commerci
Oh.. the voltage values are dumped in syslog!
Leandro Morgado wrote:
Damian Minkov wrote:
Is there a way to measure the signal of the connected line on the FXO
port ( without the help of digital oscilloscope )
Yes there is. But you need to edit the source code of wcfxs (for the
TDM400 card
ere regarding voltage values
#endif
You can simply replace it for
a) if 1
This will always print out voltage values... LOTS of them!
or
b) if (debug>1) {}
This will print out voltage values when you "modprobe wcfxs.c debug=2"
There should be similar bits of code in the X100P
Josiah Bryan wrote:
On Tuesday 05 April 2005 1:24 pm, Dov Bigio wrote:
Hello all,
I am looking for a list of all available sound files for asterisk and a
transcription of their content, so that I can have someone translate them
into portuguese.
I vaguely remeber reading some file in my serv
If u want some help put your 53xx and sip config files.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jafar mohammed
Sent: Sunday, April 03, 2005 9:41 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] AS5300+SIP+ASTERISK or AS5300+MGCP
it wasn't
very dificult with help from the very same links you gave (wiki and
junghanns docs).
Maybe it's a problem with your ISDN card? I've tried 2 other cards and
just couldn't get them to work. The Fritz works great though!
Leandro
Damian Funnell wrote:
Hi there,
We re
Anyone has an example of how a working record for agress and rates
tables should look?
I'been trying all the thinkable patterns, obviously not the right ones,
for the last two days.
Tkx, LTenorio
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Chris,
I suggest the same, but in case you want to use the fixup
feature
http://www.cisco.com/en/US/products/hw/vpndevc/ps2030/products_configura
tion_example09186a00801fc74a.shtml
LTenorio
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julio
Arr
Any other source?, I'm getting not found on that host
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike
Tkachuk
Sent: Monday, January 17, 2005 6:06 PM
To: asterisk-users@lists.digium.com
Subject: Re: FW: [Asterisk-Users] Radius on *
Hello all,
It's m
In the next couple of weeks we will be starting the beta phase of our
Guatemala POP. If you could wait, welcome.
LTenorio
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Phil Astin
Sent: Sunday, January 16, 2005 6:23 PM
To: [EMAIL PROTECTED]; asterisk-us
I'm currently trying to use a Radius server for acct and auth, cause
much of our systems are using it.
Anyone has an asterisk server working with Radius Auth and Acct?
Tkx, LTenorio
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IMHO, In your environment I would recommend to use several
Gateways (Cisco, Quintum, whatever you choose) to concentrate and do the
transcoding, and use * to switch them and do the other stuff you need. A
lot of times where seen in the list people that measure the load on *
servers that can
That's probably a timeout problem in the nat
box.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David
NortonSent: Wednesday, January 12, 2005 6:44 PMTo:
'Asterisk Users Mailing List - Non-Commercial Discussion'Subject:
[Asterisk-Users] Cant receive calls after network g
I got it, but email it to the list is not a good option.
Who 're interested just email me, I'll send it asap.
But AFAIK, you still need the wrapper.
LTenorio
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kanuri,
Seshu (Company IT)
Sent: Wednesday, J
Try sending 5350 config and oh323.conf, versions, etc...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Goran Dj.
Sent: Wednesday, December 22, 2004 4:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Asterisk-
jafar mohammed wrote:
Hi,
I purchashed a Telular Phonecell Fixed Cellular
Terminal. I hooked it up to my wildcard fxo card. I
can receive calls and these calls are passed on to the
Asterisk Calling Card application. My problem is that
i can't get DTMF to work properly. If a pin number is
48444354
n't had a change to try it out. Google around.. you find nice stuff
like: http://www.thehightechstore.com/cellinterface.html
Let me know about your experiences.
Regards,
Leandro Morgado
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Pls, post your Cisco and * config files.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jorge
Verastegui G
Sent: Friday, December 10, 2004 12:30 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Cisco AS5XXX to asterisk debugging.
Hi,
I have a ser
What configuration have I to use to enable Caller
ID for italian pstn?
usecallerid=yes;; Type of caller ID
signalling in use; bell = bell202 as used in US, v23 = v23 as used in the
UK, dtmf = DTMF as used in Denmark, Sweden and
Netherlands;cidsignalling=???
Leandro
Nope, each phone has it's own firmware.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew
Marlowe
Sent: Friday, December 03, 2004 7:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] 7905G Firmware
Is the 79
Some more info would be nice
What software version? Did you setup the Micronet as peer, proxy or as
gateway? Did you setup the user and password?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Federico
Gonzalez
Sent: Wednesday, December 01, 2004 7:03 P
What software version do u've, just 12.3T, support IP2IP feature.
I suggest you to use * instead
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Tuesday, November 30, 2004 10:53 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Use
but in /dev I have /dev/zapctl
As the permissions are now it will only work if * runs as root (which is
a bad idea). Do a 'ps aux|grep asterisk' and find out which user
asterisk is running as.
Leandro Morgado
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[E
On Wed, 2004-11-24 at 13:53, Dave Cotton wrote:
> On Wed, 2004-11-24 at 13:45 +0000, Leandro Morgado wrote:
> > Jose Hernandez wrote:
> >
> > >I installed TDM400P and X100P pci cards in a system running mandrake 10.1
> > >
en
'/dev/zap/channel': No such file or directory
Maybe the zaptel devices in /dev were not created properly. If they are
there, make sure that asterisk is running as a user that has permission
to read the /dev/zaptel devices.
Ho
If I choose immediate=no in zapata.conf, I can use the simple switch to
get digits, but how can I change the dialtone played by the simple switch
when I off hook?
Leandro
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Adding more info on my pickup weirdness, I try
other "embedded extensions", like *70 or *69. No embedded extensions are
working. Asterisk version is stable 1.0.2. Channels are Zap via channel bank and
a T100P.
Leandro
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Aste
- Original Message -
From:
Yusuf Alakavuk
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion' ; 'Walt Reed'
Sent: Friday, November 19, 2004 5:02
PM
Subject: RE: [Asterisk-Users] Call
pickup
Hi,
Have you configured features.conf file? the l
- Original Message -
From:
Walt
Reed
To: Leandro
Cc: Walt Reed ; Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Tuesday, November 16, 2004 2:11
PM
Subject: Re: [Asterisk-Users] Call
pickup
On Tue, Nov 16, 2004 at 01:26:22PM +0100
pile the kernel drivers from AVM and have capi
support in the kernel for it to work. I could not get the Fritz working
with hisax (in fact I had to remove hisax!) . Again, check the wiki..
It's all there. :)
Leandro Morgado
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Adam Goryachev wrote:
On Thu, 2004-11-18 at 05:43, Leandro Morgado wrote:
Matt Riddell wrote:
Leandro Morgado wrote:
Although, I still think that there is some kind of incompatibility or
battery drop timing problem between Asterisk and the Nokia 32. I wish
I knew more about
Matt Riddell wrote:
Leandro Morgado wrote:
Correction! I do need busydetect (i had forgotten to comment it out) to
detect hangups. I'm not as familiar with this Smartcell GSM terminal but
I dont think it drops the battery when call are hung up (in
/var/log/messages voltage stayed at 9V durin
On Wed, 2004-11-17 at 18:43, Leandro Morgado wrote:
[SNIP]
> Well.. i'm just leaving this but the mailing list archives. It really
> must be some kind of Disconnect Supervision incompatibility between the
> Nokia 32 GSM and Asterisk. Maybe asterisk doesnt like the duration tha
Matt Riddell wrote:
Leandro Morgado wrote:
Although, I still think that there is some kind of incompatibility or
battery drop timing problem between Asterisk and the Nokia 32. I wish
I knew more about telecomms and wcfxs.c to fix it!
:-) You and I both!
Well.. i'm just leaving this bu
, so I just randomly choose France (after hearing about
nightmares with CLI in UK mode!) seeing that zaptel.conf supports
defaultzone=fr .
Although this is really a work-around to the problem (the real problem
is still out there) I thank you very much for being so helpful!
Tha
- Original Message -
From: "Walt Reed" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Tuesday, November 16, 2004 1:04 PM
Subject: Re: [Asterisk-Users] Call pickup
>
> On Tue, Nov 16,
I don't understand how to get call pickup to work with asterisk.
Have I to define *8 extension in the dialplan? to what?
Have I to include something, like for parked call?
Has the stable 1.0.2 version the pickup group feature?
or I need to patch it with bristuff?
Thank you
Le
Matt Riddell wrote:
Leandro Morgado wrote:
[EXTENSIVELY SNIPPED]
zaptel.conf:
loadzone=fr
defaultzone=fr
zapata.conf:
---
busydetect=yes
busycount=7
[EXTENSIVELY SNIPPED]
1. Does the Nokia 32 GSM provide you with a hangup tone? (I.E. beep
beep beep once it has hung up)
Yes it
ortable with messing around with it's low level internals. Hints
on any "hacks" to the code that could solve this would be great.
Oh, and can anyone tell me what "Debounce" is/does ?
Thanks,
Leandro
-- /var/log/messages
-->NOTE: 48V meaning line
I get a Rhino Channel Bank and connect it to a
Digium T100P T1 card.
If I pickup analog phone and hangup without dialing
any number, I am getting extra ring after hangup.
Any idea?
Leandro
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Has anyone bought from cheapentstuff?¿
LTenorio
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Marlowe
Sent: Monday, November 01, 2004 10:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Where to bu
ed also to manual configure the channel bank
with the same result... nothing.
Can you provide me some hint on configuring these
pair of devices?
Leandro
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Just take a look at the list, or search in google, you will find the
answer.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeff
Borders
Sent: Friday, September 17, 2004 4:41 PM
To: Asterick Users
Subject: [Asterisk-Users] FC2 zaptel compile failure
I've
TO use * with a 5300 using SIP, just make a dial-peer and send
the calls using dial([EMAIL PROTECTED]) should work, I'm using it go to and
from.
Actually its MUCH MORE Stable to use * in native SIP. The code
for H323 it's not very updated.
LTenorio
-Original Message-
From:
Actually, * it's not a GK, you should configure it as regular Terminal
(Not a Gateway)in your GNUGK.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Carlos
Maynard
Sent: Thursday, September 16, 2004 10:56 AM
To: [EMAIL PROTECTED]
Subject: [Asteri
I've seen a lot of times, people that try to get R2 MFC to
*, most of them trying to use Dialogic Boards (BTW They 're Very expensive),
none of them where succesfully,
If you want to use PCI Cards on your server, why don´t
u ask to your carrier to provide you E1/PRI? or better put a Gateway w
OH323 registers itself as a Gateway, and the H323 channel as a terminal.
Afaik there is no easy way to change it.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andreas
Sikkema
Sent: Tuesday, September 14, 2004 9:30 AM
To: Asterisk Users Mailing List - No
PROTECTED] On Behalf Of Billy
Huddleston
Sent: Wednesday, September 08, 2004 11:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Cisco GW and DTMF problems
c2600-is5-mz.123-9
rfc2833
- Original Message -
From: "Tenorio, Leandro" <[EM
What version of IOS 're u using, and what's your dtmfmode in *?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Billy
Huddleston
Sent: Wednesday, September 08, 2004 6:29 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Cisco GW and DTMF problems
I'm st
Try, in the 53 (depends on the SW version u're using
voice call send-alert
Also if you're using PRI trunks you can use, in the Serial interface,.
isdn send-alerting
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Maxim
Litnitsky
Sent: Wednesday, Sep
Actually, I got almost the same issue (i´m not having such load), but I got
defines 4 different moh and got 10 process (I check every time I restart * to kill all
the mpg123 processes also.
LTenorio
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf
Just guessing, but 've you tried the to rename Sip_4602ap1_0.ebin to
appsip.ebin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aaron
Johnson
Sent: Tuesday, August 17, 2004 1:28 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Avaya firmware
Aaron
The other way, at least with Cisco Switches (I saw it also with 3com
switches some time ago), is to set every port with a secondary vlan.
What it does, it sent those 2 vlan u set to the port, it consume less
resources in your switch, and keep you far from confinguring vlan
filtering.
LTenorio
Use sip debug, that should print all the trace info for the call, id you
know the ip or the originating gateway/proxy, you cal also use sip debug
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike Roberts
Sent: Monday, August 16, 2004 6:38 PM
To: [EMA
Yep, 4602 Works. That's what I said 'most of the phones' and ask for the
model, I've tried 46XX without success, several times.
LTenorio
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian Elton
Sent: Friday, August 13, 2004 12:38 AM
To: [EMAIL PROTECTE
Be carefull, what's you phone type, most of the Avaya IP Phones use
custom h.323 protocol.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Katerina
Sadri
Sent: Wednesday, August 11, 2004 4:06 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Avaya IP P
Have u try disabling fast start?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Asterisk .
Sent: Friday, August 06, 2004 8:13 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] H323 Call Dropping
Please Help...
--- "Asterisk ." <[EMAIL PROTECTED]
hose users, trunks or systems. When that time comes, and trust me it
could be done, * could be used in orgs.
Sorry 4 my English, it's hard 2 explain in a foreign language what I try
to.
Just my 2c, Leandro
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On B
I'm looking for Spain Voice Numbers, anyone know a trustable company
that provides them?
TKX, Leandro
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Seshu, I'm interested could u provide more info...
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanuri, Seshu
Sent: Wednesday, July 14, 2004 11:02 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] SMDR/CDR - Asterisk integration
Hi All,
The CDR
256]
Memory at df80 (32-bit, non-prefetchable) [size=4K]
Anyone has any clue what might be causing this strange behaviour?
Thanks in advance,
Leandro Morgado
Eurotux / Portugal
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