Michael,
I just check'd my kernel configuration...
I have APIC support and no Enhanced Real Time Clock, exactly as you have on
your hardware.
It *could* be a timer issue, except that i can't manage how to "accelerate"
mi timer or to slow down my t1xxp driver...
--
Stefano
--
Outgoing mail is c
ope of someone who found the same problem
recently and found an appropriate solution.
Following one of the advices from this thread, i checked for busydetect and
busycount. Now i'm monitoring if calls are still dropped randomly. But it
seems that PRI error and Call drops are not so-str
line from Telco and problems due to the cable
(it's a shielded cable made from a telco operator)
Thanks in advance,
--
Stefano Finetti
Technical Coordinator
Lynx Autodelta S.r.l.
Tel.: 199797930
Fax.: 06233227934
email: [EMAIL PROTECTED]
--
Outgoing mail is certified Virus Free.
Chec
"Klaus-Peter Junghanns" wrote:
> >
> > I forgot to mention that i'm using Snom105 phones. It seems that with GS
> > BT101 with Ilbc firmware the value 160 works fine, but with snom it
> > introduces an ugly distortion and choppy audio.
> >
> This is really surprising. What codec are you using on
Well, i think i've solved the problem by myself :-)
I had to change a line in chan_capi_pvt.h:
/* was : 130 bytes Alaw = 16.25 ms audio not suitable for VoIP */
/* now : 160 bytes Alaw = 20 ms audio */
/* you can tune this to your need. higher value == more latency */
#define AST_CAPI_MAX_B3_BLOC
From: "Wolfgang Pichler" <[EMAIL PROTECTED]>
> hi,
>
> what happens if you change the rxgain and txgain to something lower than
> 1.0 ?
>
> best regards
> Wolfgang
Well, actually it just change volume on the headset...
This is not a distortion case, sound is choppy and trembling, but volume is
Hello all,
I've just finished to install chan_capi with 3 AVM Fritz PCI cards.
It correctly loads the 3 drivers, and * starts without errors.
immediately after * start, audio quality is really fine, but, after the
first incoming call, all incoming audio is broken, trembling and stuttering.
>Fro
its]) option to the Dial string. Both works only
after the answer (the latter after answer and before bridging if I have well
understood).
I'm using CVS-HEAD-05/10/04-14:56:11 with latest zaptel and libpri archives
and the latest chan_capi source (honestly i don't think it's a capi
* related issue since zaphfc is a
third party module, but i'd like to know if someone using * has already been
involved in solving this problem.
I can continue offlist if this is not of interests of people using this ML.
Best Regards,
--
Stefano Finetti
Technical Coordinator
Lynx Autodelta S.r.
Very Nice!
I'm experiencing a bit of troubles in using for some kind of channels.
Actually it shows correctly the status only on ZAP/## channels, while i
can't see anything happening on SIP/ channels neither on IAX2/ channels
(neither with the new .pl you posted).
Regards,
--
Stefa
w version solved something but it's the same.
i googled a bit to find a solution, but found only a my own question of
about 1 year ago :-(
Any hints?
Thanks in advance
--
Stefano Finetti
Technical Coordinator
Lynx Autodelta S.r.l.
Tel.: 199797930
Fax.:
From: "Pavel Litvinenko" <[EMAIL PROTECTED]>
> what incomming channel do you use, how is your * connected to the world ?
Ooops!
I just forgot to write that part of the email :-)
My * box is connected with 4 passive ISDN HFC cards (hisax driver).
So, I've 8 ttyIs usable.
When i receive a Cal
r company B
I've noticed that in kern.log the incoming MSN is logged, but didn't find
anything in the * log nor in the *CLI> screen logs
Is there a way to manage this in asterisk in order to make the different
welcomes using only different contexts (let say, [main-menu-A] and
[main-menu
it, feel free to mail me offlist.
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Stefano Finetti
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try the app. WaitMusicOnHold(time)
Just an example:
exten => 555,1,Answer
exten => 555,2,WaitMusicOnHold(30)
exten => 555,3,Hangup
--
Stefano
From: "Dan" <[EMAIL PROTECTED]>
> How can I play Music On Hold on a channel for just a limited period of
time.
> The "Musiconhold" application plays in
o activities.
on the CLI it remains the voice " CLI> called g1/803121"
Usually, when the number is really ringing and answered, after the "Called"
line, appears the "ZapX-X is ringing" line, that doesn't appear here.
Thanks again,
Stefano Finetti
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k/messages)
but i found no useful informations.
It's quite important to solve this problem 'cause i'm not able to call some
*very* important number used for my job (Telecom HelpDesk, and so on).
Thanks,
--
Stefano Finetti
___
Asterisk
esn't recognize the answer?
It stays there, waiting, even if i'm sure the other side of the line has
answered the call (tried in the same time from * and using my mobile phone).
I can't figure out what kind of problem can be, I encountered it in many *
insta
es over
ISDN now. The problem that still remains is:
1) i receive a lot of dtmf tones from the line (i4l) when i speak to phone
2) some IVR system is not recognized from *, it still wait for an answer
(ringing the line), even when the remote IVR has answered the call.
--
ed to set-up our
pbx ASAP - prior to holidays... :( )
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Stefano Finetti
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hat part is now solved... still remaining the random
hangups :(
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Stefano Finetti
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From: "Brad Bergman" <[EMAIL PROTECTED]>
>
> Curiously also, to digress, Nortel in Meridian Mail and its derivatives
> favours the term "number sign".
Just to say, in order to continue this interesting digression and to talk
about * globalization ;)
In italy the # sign is often called "Cancellet
have them.
Just take a couple of SIP phones (also soft-phones), set up the sip.conf
properly, and you'll have a perfectly running * box for VoIP use.
If you take the fxs card, you wil be able to make a normal phone ring if
called from the VoIP phones, and call f
- Original Message -
From: "WipeOut ." <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, July 16, 2003 10:19 PM
Subject: Re: [Asterisk-Users] grandstream sip phone
>
> I have tried many public NTP servers and all have the same result..
>
Wait.
I have tried many public ntp t
I'm working greatly with 40+ Grandstream phones. Audio quality is good
enough for production environment, the cost is really low and the
configuration is *Really* easy.
But a little answer to Wipeout is:
> The only issue that I still have is that the phone does not seem to be
able to pickup the t
To enable any of TxxxP or ExxxP card you must compile the zaptel package
from cvs.
Then to enable the T100P or E100P you should load the wct1xxp module.
Then you can use the zttool (/sbin/zttool) to check the status of the cards
and of the spans you've configured in /etc/zaptel.conf
--
Stefano
would be appreciated, since I've not been able to understand on my
own even reading the whole lot of messages about IAX on this list :(
--
Stefano Finetti
Technical Coordinator
Lynx Automotive srl
[EMAIL PROTECTED]
Tel: 199 79 79 30
Fax: 06 233
From: "The Traveller" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
> What does your "/etc/zaptel.conf" look like?
>
> If you have something like:
>
> span=1,0,0,ccs,hdb3
>
> Try:
>
> span=1,1,0,ccs,hdb3
>
I have tried with both, now i'm using the second one, but doesn't seems to
have changed some
12 successfully restarted on span 1
-- B-channel 13 successfully restarted on span 1
-- B-channel 14 successfully restarted on span 1
-- B-channel 15 successfully restarted on span 1
Thanks
--
Stefano Finetti
System Consultant
Lynx Automotive srl
[EMAIL PROTECTED]
t
understood by *?
--
Stefano Finetti
Technical Coordinator
Lynx Automotive srl
[EMAIL PROTECTED]
Tel: 199 79 79 30
Fax: 06 233 227 934
Linux Registered User #271978
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>
> i use X-Lite on windows
> in setup ;
>
> Display name : roseau
> user name : 1000
> authorization user :
> Password :
> Domain/Realme : 192.168.0.2
> SIP Proxy : 192.168.0.2:5060 ; i can't have this field empty
>
You're using a username that is different from [username] in sip.conf.
You can
Lynx Automotive
Research & Development Area
In order to have a fully functional * box over a PRI E1 i know I have to buy an
E100P, but:
I've to configure an analog FAX with a direct number incoming from the
PRI.
Since all the telephony solution will be over
Tt
exten => _80X.,104,Goto(main-menu,#,1)
Can also this be a problem of codec? I don't think so, but really I've no
idea of why * doesn't detect the answer on the opposite end of the line.
Suggestions? Comments?
--
Stefano Finetti
Technical Coordinator
Lynx Automotive srl
[EMAIL
>
> Do you use the same codecs in both sides (CISCO and H.323 channel
> in Asterisk)?
I have to check but... If it is a codec issue, why internally between SIP
and H323 it works perfectly?
I think I'm using for both way the G711u codec.
--
Stefano
___
; i see nothing about the answer from the 7905.
I was wondering about an oh323.conf error in my configuration, but I can't
say what error and where...
(i use asterisk-oh323 from inaccessnetworks)
--
Stefano Finetti
Technical Coordinator
Lynx Automotive srl
[EMAIL PROTECTED]
Tel: 199 79 79 30
5:22:45 asterisk kernel: isdn: hisax,ch0 cause: E021C "
I've tryed to figure out the error code, but i can't understand german
language :-(
Ideas?
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Lynx Automotive
Research & Development Area
Greetings.
In order to test * for a simple installation, i've installed a passive isdn
card (ASUSCOM in-100-p pci) that I know it's recorded to work under linux.
When i put in modem.conf the line device=/dev/ttyIx
I'm testing it.
It works quite well, making my Cisco 7905 works with GnuGK and asterisk
--
Stefano Finetti
Technical Coordinator
Lynx Automotive srl
[EMAIL PROTECTED]
Tel: 199 79 79 30
Fax: 02 233 227 934
Linux Registered User #271978
- Original Message -
From: "Michae
;modem/ttyI0:0/$(EXTEN)" it starts calling doing nothing, nor errors not
warning, just going in "nobody picked up after ms".
--
Stefano Finetti
Technical Coordinator
Lynx Automotive srl
[EMAIL PROTECTED]
Tel: 199 79 79 30
Fax: 02 233 227 934
Linux Registered User #271978
___
o use
h323 unless you buy the last release with h323 software installed)
--
Stefano Finetti
Technical Coordinator
Lynx Automotive srl
[EMAIL PROTECTED]
Tel: 199 79 79 30
Fax: 02 233 227 934
Linux Registered User #271978
___
Asterisk-Users mailing l
extensions data at given
times, and the php will be able to read from mysql the extensions, write in
Mysql the amendments and, at given times, syncronize the files (to be
flagged "modified" by php).
--
Stefano Finetti
Technical Coordinator
Lynx Automotive srl
[EMAIL PROTECTED]
Tel: 199
ve the Gui ready for
use not so far from now (if someone obviously is interested in)
(the big idea is to make a single gui for both * and OpenH323 Gatekeeper in
order to have the full management of telephony)
Regards,
--
Stefano Finetti
Technical Coordinator
Lynx Automotive srl
[EMAIL PROTECTED]
T
they will stai in H.323 context when calling internally.
Do you have any suggestion or advice about that kind of solution?
Since I've not started installing the machines yet, I'm open to any
modification on my plans ;-)
Thanks,
--
Stefano Finetti
Technical Coordinator
Lynx Automotive
Title: Message
Congratulations :-)
You just confirmed me that will work what I'm
working for (not that i didn't believe it, * is one of the best software i've
found working under linux)
--Stefano
FinettiTechnical CoordinatorLynx Automotive srl[EMAIL PROTECTED]Tel:
199 79 79 30Fax: 02 233
king about a T100P with T1 Channel Bank, after reading this ML.
Is this the right way?
Can you indicate me some Channel Bank findable in italy?
Thanks,
--
Stefano Finetti
Technical Coordinator
Lynx Automotive srl
[EMAIL PROTECTED]
Tel: 199 79 79 30
Fax: 02 233 227
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