martedì, luglio 19, 2005

Asterisk@home hfc isdn dondisperato

A@h 1.3 - 1.5
how to install hfc card

after unload asterisk and amportal whit
amportal stop

type "setup"
unselect zaptel in system service...
and set the lan

--->reboot<---

cd /usr/src
wget http://www.junghanns.net/downloads/bristuff-0.2.0-RC8j.tar.gz
tar -zxvf bristuff-0.2.0-RC8j.tar.gz
cd bristuff-0.2.0-RC8j
./download.sh
cd zaphfc
wget http://zaphfc.florz.dyndns.org/zaphfc_0.2.0-RC8j_florz-8.diff.gz
gunzip -v zaphfc_0.2.0-RC8j_florz-8.diff.gz
patch < zaphfc_0.2.0-RC8j_florz-8.diff
ln -s /usr/src/linux-2.4.21-32.EL /usr/src/linux-2.4
ln -s /usr/src/linux-2.4.21-32.EL /usr/src/linux
cd ..
cd zaptel-1.0.9
make clean
make
make install
cd ..
cd libpri-1.0.9
make clean
make
make install
cd ..
cd zaphfc
make clean
make

cp zaptel.conf /etc/zaptel.conf
------>yes<-------

nano /etc/rc.d/rc.local

at first add this line:

modprobe zaptel
insmod /usr/src/bristuff-0.2.0-RC8j/zaphfc/zaphfc.o
ztcfg -vv

nano /etc/asterisk/zapata.conf

this is my zapata:

[channels]
language=it
switchtype=euroisdn
;If you connect to a hicom PBX set your ISDN Numbering Plan Identifier to unknown.
pridialplan=local
prilocaldialplan=local


signalling = bri_cpe_ptmp
;signalling = fxs_ks
rxwink=300

usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
nationalprefix = 0
internationalprefix = 00
faxdetect=incoming
group=0
callgroup=1
pickupgroup=1
immediate=yes
context=from-pstn
channel => 1-2

________________________________________
now you can recompile asterisk
cd /usr/src/bristuff-0.2.0-RC8j/asterisk-1.0.9
make clean
make
make install

reboot...
________________________________________


that's all
donDisperato








Support donDisperato

I'm searching equipment voip / sip / h323 / isdn / gsm for testing..... if you can contribute contact me at disperato at gmail.com

Donations:
50€ Takahiro Inui
50€ Matteo Maggiani
30€ Marco Cipriani


p.s. this help work correctly for TE or NT mode...

for NT mode read the correct signalling type in www.voip-info.org...

for disconnect problem whit isdn phone and iax/sip provider please read at http://sourceforge.net/forum/forum.php?thread_id=1371545&forum_id=420324
many thanks at G.Veloce ;-)

Comments:
i thought your blog was cool and i think you may like this cool Website. now just Click Here
 
Hi Don!

I just tried your 'how to' with kernel 2.6, and the compilation of zaphfc doesn't work.

Any suggestions for that issue would be appreciated!

Thanks a lot!
Richie444
 
dear richie,

please whait....
 
Any idea, for to install hfc cards at a@h 2beta3, or 2.6 kernels?
 
Hello Don!
Your howto works perfectly.
I have only one problem:
After done that, the other card I have installed on my system (which is a original digium fxo card) doesn't work anymore because the modules wcfxo and wcfxs don't get loaded anymore, and if I try to "modprobe wcfxs" it just hangs up my pc :(
Any idea on how to use both this isdn card AND the standard digium fxo pstn card?
Thanks in advance
P.S. I am italian, so if u want to reply in Italian it is ok
 
Just checking out blogs for ideas to add to my site about voip network and other voip stuff. (I know its a boring subject) I liked your site
 
I really enjoyed your blog. This is a cool Website Check it out now by Clicking Here . I know that you will find this WebSite Very Interesting Every one wants a Free LapTop Computer!
 
Questo commento è stato eliminato da un amministratore del blog.
 
thanks very mutch for this DOC

I succed to receve call by ISDN

When I try to make out going it fail!!

cisco@no-log.org

Oct 16 10:38:51 VERBOSE[2263]: -- Goto (macro-record-enable,s,99)
Oct 16 10:38:51 VERBOSE[2263]: -- Executing NoOp("IAX2/200@200/4", "NO RECORDING NEEDED") in new stack
Oct 16 10:38:51 DEBUG[2263]: Expression is '1'
Oct 16 10:38:51 VERBOSE[2263]: -- Executing GotoIf("IAX2/200@200/4", "1?7") in new stack
Oct 16 10:38:51 VERBOSE[2263]: -- Goto (macro-dialout-trunk,s,7)
Oct 16 10:38:51 DEBUG[2263]: Expression is '1'
Oct 16 10:38:51 VERBOSE[2263]: -- Executing GotoIf("IAX2/200@200/4", "1?9") in new stack
Oct 16 10:38:51 VERBOSE[2263]: -- Goto (macro-dialout-trunk,s,9)
Oct 16 10:38:51 VERBOSE[2263]: -- Executing SetGroup("IAX2/200@200/4", "OUT_3") in new stack
Oct 16 10:38:51 VERBOSE[2263]: -- Executing CheckGroup("IAX2/200@200/4", "2") in new stack
Oct 16 10:38:51 VERBOSE[2263]: -- Executing SetVar("IAX2/200@200/4", "DIAL_NUMBER=0478855054") in new stack
Oct 16 10:38:51 VERBOSE[2263]: -- Executing SetVar("IAX2/200@200/4", "DIAL_TRUNK=3") in new stack
Oct 16 10:38:51 VERBOSE[2263]: -- Executing AGI("IAX2/200@200/4", "fixlocalprefix") in new stack
Oct 16 10:38:51 VERBOSE[2263]: -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
Oct 16 10:38:51 VERBOSE[2263]: fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf
Oct 16 10:38:51 VERBOSE[2263]: -- AGI Script fixlocalprefix completed, returning 0
Oct 16 10:38:51 VERBOSE[2263]: -- Executing SetVar("IAX2/200@200/4", "OUTNUM=0478855054") in new stack
Oct 16 10:38:51 VERBOSE[2263]: -- Executing Cut("IAX2/200@200/4", "custom=OUT_3|:|1") in new stack
Oct 16 10:38:51 DEBUG[2263]: Expression is '0'
Oct 16 10:38:51 VERBOSE[2263]: -- Executing GotoIf("IAX2/200@200/4", "0?19") in new stack
Oct 16 10:38:51 DEBUG[2263]: Not taking any branch
Oct 16 10:38:51 VERBOSE[2263]: -- Executing Dial("IAX2/200@200/4", "ZAP/1/0478855054") in new stack
Oct 16 10:38:51 VERBOSE[2263]: -- Requested transfer capability: 0x00 - SPEECH
Oct 16 10:38:51 VERBOSE[2263]: -- Called 1/0478855054
Oct 16 10:38:51 VERBOSE[2263]: -- Channel 0/1, span 1 got hangup
Oct 16 10:38:51 DEBUG[2263]: Set option AUDIO MODE, value: ON(1) on Zap/1-1
Oct 16 10:38:51 DEBUG[2263]: Hangup: channel: 1 index = 0, normal = 15, callwait = -1, thirdcall = -1
Oct 16 10:38:51 DEBUG[2263]: Already hungup... Calling hangup once, and clearing call
Oct 16 10:38:51 DEBUG[2263]: disabled echo cancellation on channel 1
Oct 16 10:38:51 DEBUG[2263]: Set option TDD MODE, value: OFF(0) on Zap/1-1
Oct 16 10:38:51 DEBUG[2263]: Updated conferencing on 1, with 0 conference users
Oct 16 10:38:51 DEBUG[2263]: Set option AUDIO MODE, value: OFF(0) on Zap/1-1
Oct 16 10:38:51 DEBUG[2263]: disabled echo cancellation on channel 1
Oct 16 10:38:51 VERBOSE[2263]: -- Hungup 'Zap/1-1'
Oct 16 10:38:51 VERBOSE[2263]: == No one is available to answer at this time
Oct 16 10:38:51 DEBUG[2263]: Exiting with DIALSTATUS=NOANSWER.
Oct 16 10:38:51 VERBOSE[2263]: -- Executing Goto("IAX2/200@200/4", "s-NOANSWER|1") in new stack
Oct 16 10:38:51 VERBOSE[2263]: -- Goto (macro-dialout-trunk,s-NOANSWER,1)
Oct 16 10:38:51 VERBOSE[2263]: -- Executing NoOp("IAX2/200@200/4", "Dial failed due to NOANSWER") in new stack
Oct 16 10:38:51 VERBOSE[2263]: -- Executing Macro("IAX2/200@200/4", "outisbusy") in new stack
Oct 16 10:38:51 VERBOSE[2263]: -- Executing Playback("IAX2/200@200/4", "allison7/all-circuits-busy-now") in new stack
Oct 16 10:38:51 DEBUG[2263]: Scheduling timer at 160 sample intervals
Oct 16 10:38:51 VERBOSE[2263]: -- Playing 'allison7/all-circuits-busy-now' (language 'en')
Oct 16 10:38:51 DEBUG[2263]: Ooh, voice format changed to 2
Oct 16 10:38:51 VERBOSE[2263]: -- Registered '220' (AUTHENTICATED) at 119.69.1.100:4569
Oct 16 10:38:53 DEBUG[2263]: Scheduling timer at 0 sample intervals
Oct 16 10:38:53 DEBUG[2263]: Scheduling timer at 0 sample intervals
Oct 16 10:38:53 VERBOSE[2263]: -- Executing Playback("IAX2/200@200/4", "allison7/pls-try-call-later") in new stack
Oct 16 10:38:53 DEBUG[2263]: Scheduling timer at 160 sample intervals
Oct 16 10:38:53 VERBOSE[2263]: -- Playing 'allison7/pls-try-call-later' (language 'en'
 
I tried these instructions and I now have a problem starting asterisk. The actual message is:

Dec 2 08:09:39 WARNING[3263] chan_zap.c: Unable to specify channel 1: No such device or address
Dec 2 08:09:39 ERROR[3263] chan_zap.c: Unable to open channel 1: No such device or address
here = 0, tmp->channel = 1, channel = 1
Dec 2 08:09:39 ERROR[3263] chan_zap.c: Unable to register channel '1-2'
Dec 2 08:09:39 WARNING[3263] loader.c: chan_zap.so: load_module failed, returning -1
Dec 2 08:09:39 WARNING[3263] loader.c: Loading module chan_zap.so failed!

I also notice when I run 'ztcfg -vv'
by hand I get:

ZT_SPANCONFIG failed on span 1: Invalid argument (22)

after some success messages. I don't know if the above is related. Any help would be appreciated
 
I tried everything from scratch and I notice that when I manually do /etc/rc.d/rc.local Asterisk does start up. But not on reboot...
 
We put a line,
sleep 10
in front of:
ztcfg -vv

and now on reboot Asterisk starts up fine.
 
Ciao,
ho utilizzato già un paio di volte le tue indicazione e tutto funziona perfettamente.

Il 02/12 mi sono collegato al blog ed ho visto che avevi iniziato una pre per a@h 2.0, quindi oggi 06/12 ho scaricato a@h (la 2.1) e mi sono collegato al blog per seguire le istruzioni, ma ho visto che hai rimesso quelle per a@h 1.3-1.5, qualche problema per la 2.0/2.1 ?

Vittorio
 
sistemato ;-)
 
Hi Thanks for your interesting blog. I also have a blog/site, covering flyfone voip related stuff. Feel free to visit my flyfone voip site.
 
It appears a hugh amount of integration work for automated HFC-S installation has already been done here:

https://sourceforge.net/forum/forum.php?thread_id=1394672&forum_id=420324
 
Posta un commento



<< Home

This page is powered by Blogger. Isn't yours?