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 http://www.chan-mobile.or
in
exchange of nothing).
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2006/6/2, Leon Sun <[EMAIL PROTECTED]>:
10$/channel
If you are connecting a device that uses g729 with another that don't
support it... let's say it uses gsm. Then you will use 2 channels, one
for encoding and one for decoding. Is it?
--
e similar to the one of the newer version for what the
patch has been done) and modify it with your favorite text editor (vi
rulez!). Then compile...
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e "beta" patches on a production machine.
Download the source code for the version that you are using and patch
it yourself. It should not be so difficult.
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lafax where in english.
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_var_spool_fax_bin_faxrcvd
Description: Binary data
_var_spool_fax_etc_FaxDispatch
Description: Binary data
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ink it sould be possible, but why not send an email with a
pdf directly instead of using hylafax?
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ms. I can't say it works perfectly, but in my few tests no faxes
where lost, and in production I di'nt receive complaints.
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\r\n");
sleep(2) ;
fclose ($fp);
}
echo "Extension $number should be ringing now." ;
else :
exit() ;
endif ;
?>
Add this to extensions.conf (or extensions_custom.conf if you are
using asteriskathome)
[dialtone]
exten => s,1,DISA(1234,from-internal)
exten => s,2,Hangup
where &qu
things you like to do, based on the
caller id i.e.
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r file and include it.
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tion click here ...
Some voip providers uses mgcp or even the old h323. This should work
but will be more difficult to configure because it is uncommon.
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ing as data.
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fun colaborating with other people.
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ck and unreliable way to show a document to
someone instead of dictating it by phone.
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olve for the developpers
what knows exactly where in the code to make the changes...
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f for sening via e-mail.
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strange
things like this. I never could find a way to download a sip firmware
to update the mgcp devices.
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cept that you want to accept
(http, smtp, etc.).
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x27;m using my own agi-bin for "patching" callerid and adding the name
if the number is found in a table (a csv that is mantained with a
spreadsheet), it adds the name taken from this table. Then you can see
the name in the display of the phones.
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2006/3/17, adriano ghezzi <[EMAIL PROTECTED]>:
> I'd like to use skpe as an extension or a channel with asterisk.
By the way, check this: http://www.voip-info.org/wiki-bounty+skype
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e as helper for
this connections. Obviously it should be dedicated (no users in it) in
order to reduce system faults. It runs using the skype api, then
another con is that in windows probably you will not be able to run
more than one copy of skype and you only will be able to mantain only
one conve
How is working the automatic fax detection? I'm making tests in
asteriskathome and the ivr plays, the fax sends little bips but
asterisk don't detects it as a fax.
(for testing I routed one caller id to the ivr).
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e the thing like this:
exten => _.,1,Goto(s,1)
exten => s,1,AGI(agenda,${CALLERIDNUM})
All starts working, but idn is lost and I can't do inbound routing
based on dnid. Is there a manual that explains about this configs?
P.D.: agenda is an agi-script that patches the incamming caller-i
2006/2/22, Alejandro Vargas <[EMAIL PROTECTED]>:
> I want to do inbound routing of calls comming from sip trunks. Is
> there a way to force the DID that comes from a trunk that does not
> have DID support? (something like using the outgoing caller-id for the
> trunk?)
I
hen it does not matter if it has tihs codecs.
If you install G729 and/or G723 in Asterisk. In this case, Asterisk
can decode the audio and re-encode with speex or iBLC.
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in a busy tone. ¿What can be the problem?
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ome softmodems uses chips that works, but you
must choose the right one. See www.voip-info.org.
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accesed as a
faxmodem. Then, you can use hylafax that is very powerfull and can be
configured to forward faxes to email, convert it to pdf, etc. etc
(read the documentation).
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As
this, when, making again a new call from
spa2100 to spa3000, spa2100 receives only white noise. I suspect a
codec mismatch. The problem disappears by powering off and on the
spa3000.
¿Any ideas on how to check?
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2006/3/7, Giridhar Bandi <[EMAIL PROTECTED]>:
> and when i place a call to local sip extension there is a long pause ( 15
> sec )
> before the call gets dialled
Use the # key as "enter".
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Has anybody a firmware for updating a mediatrix 1102 (sip)?
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I red that it is possible to send instant messages to the displays of
sip phones. How can I do it using Asterisk?
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asterisk pbx, use FXS cards, FXS
to sip adapters (like SPA2100) or sip phones.
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tware fax.
An interesting replacement for asterisk software fax is using iaxfax+hylafax.
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eex codec and use it if your connection is through
internet.
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I've seen in the asterisk configuration the way to call some internal
variables like caller-id-number, caller-id-name, language, etc. but..
What is the variable for changing the DID?
Is there a manual with this details?
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need to improve (or decress) is the bandwidth usage.
Check this: http://www.voip-info.org/wiki/view/Bandwidth+consumption
but try speex if it is supported by your sip phone. It is free,
variable bitrate and adapts to the available bandwidth, (it is based
on o
en => s/7045,1,SetVar(FROM_DID=s/7045)
exten => s/7045,2,Goto(ext-local,211,1)
exten => s/987073366,1,SetVar(FROM_DID=s/987073366)
exten => s/987073366,2,Goto(ext-local,211,1)
exten => _X./7045,1,Goto(s/7045)
exten => _X./9
think it would be cleaner
that if the trunk does not support did make it assumes the out caller
id. I do not know much of asterisk configurations, then I couldn't
find a way to do this, but I supose there must be a way.
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VR in all but one of them, the one that is connected to a
cellular adapter. In this line I want to let it ring until somebody
picks up because many times we checks the caller-id and calls him
back.
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omatic
fax detection, asterisk needs to answer the line in order to hear of
there is a fax carrier. If you disable it, asterisk never answers the
call and spa3000 also don't answer.
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2006/2/21, Alejandro Vargas <[EMAIL PROTECTED]>:
> I'm configuring a sip trunk. My problem is if I configure the sip
> device to dial to a sip phone, it works ok but when I dials to "s" or
> "", asterisk picks up the call immediatly and places it's
d for
answer the call picks up.
Is there a way to avoid it? Is it a problem of the sip trunk? Should I
post this question to devel list?
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e
first ring, even when I specified "Off Hook While Calling VoIP: NO".
Is this a problem of spa3000 or a problem with asterisk? Who is
deciding to answer the call?
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60:
OPTIONS sip:sip1.sipdiscount.com SIP/2.0
Via: SIP/2.0/UDP 81.172.52.3:5060;branch=z9hG4bK0596a5f2;rport
From: "Unknown" ;tag=as6c5807a2
To:
Contact:
Call-ID: [EMAIL PROTECTED]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 20 Feb 2006 08:16:56 GMT
Allow: INVIT
t, but It never worked before (always needed to use test).
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count.com.
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Username Refresh State
sip show peers
sipdiscount/test 80.239.235.200 N 5060 OK (60 ms)
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to-asterisk, you should try speex
compression. In my tests, speex had the better quality even with low
bandwidth or bandwidth very occuped by other applications (p2p).
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Asterisk-
user=test
fromdomain=stun.sipdiscount.com
dtmfmode=inband
disallow=all
canredirect=no
allow=gsm
allow=ulaw
allow=alaw
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Sipdiscount has replaced their asterisk servers for another thing.
Then, no more iax. Ok, but I can't make calls using sip also... I'm
getting a "forbidden" error when using sip1.sipdiscount.com. Anybody
got it working?
using asteriskatmome 2)
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Is this normal? Can I ignore this messages?
Dec 14 13:18:06 DEBUG[5312] db.c: Unable to find key
'MGCP/aaln/[EMAIL PROTECTED]' in family 'cfb'
Dec 14 13:18:06 DEBUG[5312] db.c: Unable to find key
'MGCP/aaln/[EMAIL PROTECTED]' in family &#
and will keep
it ok.
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2005/12/7, xcel <[EMAIL PROTECTED]>:
>
> I did use IAX2 but sound quality wasn't that good which codec are you using
> with IAX2 ?
The sound quality doesn't depend on the protocol but the codec you are
using and the bandwidth you are giving to
d any
module. For an easy config try amp.
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2005/12/6, C F <[EMAIL PROTECTED]>:
> Yeah, it shoud NOT work 100% of the time (maybe not even 50%)
Then... ¿there is not any way to connect a hardware fax to an asterisk pbx?
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k
I've found the answer:
http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers
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Memory at cfef (32-bit, non-prefetchable) [size=64K]
Expansion ROM at cfec [disabled] [size=128K]
Capabilities: [58] AGP version 3.0
Capabilities: [50] Power Management version 2
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cron, but what is the minumun change I must do from this script? Is
there some command from the asterisk interface for doing the change on
line without modificating the config files?
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Is there any way to create various trunks with the same priority.
I'm interested on usingo 2 trunks, but balancing the usage in both
because both has a number of free minutes. If I give preference to one
over other, this one will exceed the free limit much before the other.
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le to call
extensions in each other and use the external lines, then I supose I
must place it in from-internal context.
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registration from the
other side. I'm using amp (from [EMAIL PROTECTED]).
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c. But the more commonly supported is the propietary g729. To use
it you should pay for each communication. If your phone does not
support it, asterisk must do transcoding in both directions and you
must pay twice. There is a royalty-free version for private use
provided by intel.
--
A
rying to test it
with asterisk because if it works, I will place credit on my account.
But if I can't make it work with an account with credit, I must
suspect they are blocking the accounts with credit for avoiding people
to use it with other software than their propi
ser that has credit but they never answered.
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not using
standard iax or sip ports). Are the acconts with credit blocked for
avoiding it's use with ohter software than voipbuster's?
I tryed to send a mail to voipbuster's support but I never received an
answer (then do not support other thing than their software).
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Is there any way to tell asterisk that ignore protocol errors instead
of dropping the call?
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or install directly
asteriskathome. It solves all the problems of configuring and creating
extensions. Then you can start lerning how to do the difficult things.
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Asterisk-U
ge of RQNT 404 aaln/[EMAIL PROTECTED] MGCP
1.0 and the device responds 200 404 OK, but following this, the
asterisk server starts rejecting the packets of the port it were using
to communicate to the device.
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2005/12/1, Pablo Allietti <[EMAIL PROTECTED]>:
> hi sean you have a example please?
In your zapata.conf ensure there is context=from-pstn like this:
[channels]
context=from-pstn
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2005/11/30, Francesco Peeters <[EMAIL PROTECTED]>:
> When you do, make VERY sure the PCI slots are NOT sharing an IRQ! That'll
> break it every time!
Did you try to use APIC? This is suposed to solve the problem of IRQs
--
zaphfc /sbin/modprobe --ignore-install zaphfc && /sbin/ztcfg
And this to /etc/sysconfig/zaptel:
MODULES="$MODULES zaphfc"
Thank you again. The next step is to try 2 cards...
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g files)
>
> It can be most easily done with compile.sh in the BRIstuff folder, which
> should - normally - compile and install everything in the correct order...
But... I already tried it at first, and asterisk stopped working...
Well... I'll do it and check wh
asteriskathome, is it??
When it worked I will go to voip-info.org and add some detailed
instructions there...
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e_sem+0x1b2/0x1b6
Nov 30 13:51:37 asterisk1 kernel: [] init_module+0x65/0xe0 [zaphfc]
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fc]# insmod ./zaphfc.ko debug=3
(and the system hangs) Kernel panic, fatal exception. Is it necesary
the Florz patch?
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orz patch?
Yes, I applied it today and I'm repeating the tests, but with this
patch, when I modprobe zaphfc the system hangs.
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s error), then cat /proc/zaptel/* and the system
hanged again.
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channel 1: No
such device or address
here = 0, tmp->channel = 1, channel = 1
Nov 30 12:00:44 ERROR[3422] chan_zap.c: Unable to register channel '1-2'
Nov 30 12:00:44 WARNING[3422] loader.c: chan_zap.so: load_module
failed, returning -1
Nov 30 12:00:44 WARNING[3422] loader.c
p.c: Unable to open channel 1:
Device or resource busy
here = 0, tmp->channel = 1, channel = 1
Nov 29 14:14:25 ERROR[4240] chan_zap.c: Unable to register channel '1-2'
Nov 29 14:14:25 WARNING[4240] loader.c: chan_zap.so: load_module
failed, returning -1
Nov 29 14:14:25 WARNING[4240] lo
yes, but it broke asterisk installation Asterisk now
exits with this message.
Ouch ... error while writing audio data: : Broken pipe
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riskathome (why bristuff is not included?)
bristuff-0.3 is listed as experimental, should I use 0.2 (stable)?
And then... I will obtain the module zaphfc, then how to configure
asterisk to use it?
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way but chan_mdem_i4l does not appear
whan I type reload.
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nsfer= no
dtmfmode= rfc2833
line => aaln/2
The device, again works ok but I can call to 308 dialing to MGCP/aaln/[EMAIL
PROTECTED]
What can be the problem? Is correct to dial MGCP/aaln/[EMAIL PROTECTED]
I'm using [EMAIL PROTECTED] 2.0 beta 6
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2005/11/25, Bharath <[EMAIL PROTECTED]>:
> have you added allow=speex & allow = ilbc in the sip & iax conf files ?
Yes. And the connection is stablished, the log says it is using speex
(i.e.), but I don't receive any sound. This weekend I will do more
tests.
some config for using it?
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e driver stills unpatched. ¿Didn't anybody send this patch to
the kernel list?
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bluetooth part by using an usb
connection to the phone?
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people still thinking on doing it, because switchng from
skype to voipbuster is as sifficult as switching from
micro$oft-messenger to jabber: there is no reason for not doing it,
but people doesn't.
On other way, I must accept that skype codec has a very good compression.
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rested on doing it.
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rts. Choose the project nearest your needs and help the team to
improve it to meet what you need. It always will be better than
working alone without help.
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h. Is there a way to do this
(using a cellphone conected through USB port) already included?
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Alejandro Vargas
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