On 1/7/2013 9:22 AM, Dan Austin wrote:
The last PRI I setup in Hong Kong was configured as a Primary-Net5,
which maps to euroisdn in DAHDI. That was eight years ago, so things
may have changed, but it is worth a try.
You should also collect some Q.931 logs, as I have seen silly things
like cal
hi folks.
i recently setup an Asterisk system in Hong Kong. their phone
company told me that their T1 PRI switch type is Primary-NTT.
however in chan_dahdi.conf there's no such option. i have it
set to national. it worked fine for a while, but now suddenly
stop working. in coming call just keep
achine check exceptions
MCP: 2068 2068 2068 2068 Machine check polls
ERR: 0
MIS: 0
--
Edwin Lam
Systems Engineer, OfficeWyze, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:1
Fri, Nov 2, 2012 at 8:40 PM, Edwin Lam mailto:edwin@officegeneral.com>> wrote:
hi folks.
recently some of our customers complained about bad voice
quality on the phone system. i looked at the logs and found
a lot of these:
[2012-11-03 08:26:38] NOTICE[11305] chan_
group=1
callgroup=1
pickupgroup=1
faxdetect=incoming
context=defaultspan1
channel => 1-23
--
Edwin Lam
Systems Engineer, OfficeWyze, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?o
es. you have to
renumber all the messages so that they are consecutive otherwise
the voicemail application may give you grief.
A little doubt here, once the user hears the voicemail using the phone, the
message is automatically moved to Old folder, is that right?
yes
--
Edwin Lam
Systems Engineer,
sterisk-users@lists.digium.com
Sent: Wednesday, December 7, 2011 11:56:20 AM
Subject: [asterisk-users] Realtime Registration
[Dec 7 19:31:18] DEBUG[4217]: res_config_pgsql.c:821 pgsql_reconnect:
Postgresql RealTime: Everything is fine.
[Dec 7 19:31:18] DEBUG[4217]: res_config_pgsql.c:201 realtime_pgsql:
et.
(and minimum of x permission for the directory tree
it resides in)
--
Edwin Lam
Systems Engineer, OfficeWyze, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=get&searc
connect). i just hear ring back tone for about 20 seconds
and then become fast busy. is there any setting i'm unaware of
when setting up sip w/ skype connect?
any suggestions would be appreciated.
--
Edwin Lam
Systems Engineer, OfficeWyze, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http:
works perfectly. maybe you can
upgrade yours
--
Edwin Lam
Systems Engineer, OfficeWyze, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20
--
_
-- Bandwidth and
e this:
81dc9bdb52d04dc20036dbd8313ed055:Virendra
9996535e07258a7bbfd8b132435c5962:Vijay
7bccfde7714a1ebadf06c5f4cea752c1:VirendraBhati
the original format (i.e. :) was correct.
one question, when you create the md5hash did you use the echo
command? if so, did you specify the "-n" option?
e.g. echo -n 12345 | m
isk 6.2.x . Any ideas what can I do to figure out why it does not
detect
the arriving faxes ?
in your dialplan, did you "Answer" the call and "Wait"
a few seconds for Asterisk to detect the fax tone before
you do some other things?
--
Edwin Lam
Systems Engineer, OfficeWyze, I
he OS will always pick
the primary address when sending out packets on that subnet.
--
Edwin Lam
Systems Engineer, OfficeWyze, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20
--
rt 6060 to forward to 5060.
--
Edwin Lam
Systems Engineer, OfficeWyze, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20
--
_
-- Bandwidth and Colocation Provided by
oing_ident=04457${MATH(8900+${MATH(${CALLERID(num)}-100)})})
i just did this out of my head, i haven't test it.
but this should map all 100-399 extensions to DID 044578900-0444579199
--
Edwin Lam
Systems Engineer, OfficeWyze, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http
be its master. Likewise, BRI TE ports should always be configured
> as a slave. Any number of ports can be marked as 0.
>
> [1] http://svn.asterisk.org/view/dahdi/tools/trunk/system.conf.sample?view=co
--
Edwin Lam
Systems Engineer, Office
lkes about _another_ config file!<==
So which file should i configure:
A) res_config_mysql.conf
B) res_odbc.conf
C) res_mysql.conf
But even when i put my credentials in all three of them, still no show!
i believe 1.8.x uses res_config_mysql.conf for mysql realtime.
you can get rid of res_mysq
ld do to customize the file name of the recording.
I guess some changes to the dialplan is required ?
try something like:
Monitor(wav,${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${EXTEN})
--
Edwin Lam
Systems Engineer, OfficeWyze, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.
here. I would really
appreciate
it if someone could give me some pointers on where to go next, what additionnal
debugging steps I should perform. I would also really appreciate if someone
could
propose a solution.
Please help!
David
Never give up, never surrender
--
Edwin Lam
Systems Engineer, Of
like reading from a
text file, database rely on column values for sorting.
i don't think having 'n' as the priority will sort the way
you want.
--
Edwin Lam
Systems Engineer, OfficeWyze, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgp
h versions (1.8.3.2 & 1.6.2.17). the output are
pretty much identical except on 1.8.3.2, after the
"PROGRESS with cause code 127..." message. i would hear
nothing until the other side timed out & hang up, whereas on
1.6.2.17. i got the "DAHDI/... is making progress passing it
lly downgraded it back to 1.6.2.17, now everything work.
--
Edwin Lam
Systems Engineer, OfficeWyze, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20
--
_
-- Bandwidt
list's archive and found this:
http://lists.digium.com/pipermail/asterisk-users/2006-December/174258.html
--
Edwin Lam
Systems Engineer, OfficeWyze, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0
ld involve installing 2 extra servers for such
purpose. however you can look into csync2 to sync all asterisk
files between the 2 asterisk servers if you don't want extra
hardware.
--
Edwin Lam
Systems Engineer, OfficeWyze, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11
hi folks.
i've been experimenting with SILK codec and meet with some
success on incorporating it in pjsip (an open source sip client).
now i'm trying to do the same thing on Asterisk. any documentations,
pointers, etc i should look into? any help is appreciated.
--
Edwin Lam
System
rom anywhere.
normally i'd just omit the "defaultip" parameter.
also is your softphone and/or asterisk box behind NAT?
--
Edwin Lam
Systems Engineer, OfficeWyze, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20
--
_
ver you
can use port forwarding on OS level to achieve that.
iptables -t nat -I PREROUTING -i -d -p
udp
-m udp --dport 6080 -j DNAT --to-destination :5060
p.s. the default SIP port is 5060.
--
Edwin Lam
Systems Engineer, OfficeWyze, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkey
On 10/3/10 11:20 AM, Daniel Knoll wrote:
> Hello,
> is it possible to check more than one condition for GOTOIF in the dialplan?
yes. check out asterisk expressions on wiki pages
http://www.voip-info.org/wiki/view/Asterisk+Expressions
--
Edwin Lam
Systems Engineer, OfficeWyze, Inc.
Ph:
{EXTEN})
exten => _X!,n,...
.
.
exten => t,1,Hangup() ;hang up if no input for 7 sec.
--
Edwin Lam
Systems Engineer, OfficeWyze, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20
--
___
.3.0...
>
> Can anyone tell me how to build fxstest?
cd to tools directory then enter the command "make menuselect"
select "fxstest", save & exit, then execute make.
--
Edwin Lam
Systems Engineer, OfficeWyze, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 33
sterisk -> PRI -> phone co.
i call the same cell# and if it's unavailable. the PRI return
cause code 31 and hangup, asterisk will then send a SIP BYE to
the sip phone and the channel will simply hangup. how do i
get the message on the sip phone?
--
Edwin Lam
Systems Engineer, Offic
lly like applications against it, even though
> /usr/local is not in the library search path. Dumb, but true.
>
> The installation information page for spandsp, at
> http://www.soft-switch.org/installing-spandsp.html , warns about these
> issues.
--
Edwin Lam
Systems Engineer
i gave up on ReceiveFAX and uses iaxmodem/hylafax instead.
Tommy Botten Jensen wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA512
>
> Edwin Lam skrev:
>> Klaus Darilion wrote:
>>> The backtrace is not useable. Try to rebuild Asterisk with the "Do
Klaus Darilion wrote:
> The backtrace is not useable. Try to rebuild Asterisk with the "Don't
> Optimize" Option ("make menuconfig" and the the build options)
did that. no effect.
i've got exactly the same result.
> Edwin Lam wrote:
>> Philip A.
Philip A. Prindeville wrote:
> On 03/08/2010 04:31 PM, Edwin Lam wrote:
>> hi folks.
>>
>> i recently upgraded asterisk to 1.6.1.17(from 1.2) and i'm having
>> problems with fax. after receiving fax with the ReceiveFAX app.
>> everything seems ok. the .tiff
terisk/fax/4502-1268079069.417.tif x...@.com ") in new stack
[Mar 8 12:12:31] VERBOSE[30115] chan_dahdi.c: [Mar 8 12:12:31] -- Hungup
'DAHDI/8-1'
asterisk: 1.6.1.17
spandsp: 0.0.6pre17
--
Edwin Lam
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 41
is situation.
what you can do is setup automatic dial to different extensions on
the 2 ports on audiocodes.
--
Edwin Lam
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20
--
__
t; If you don't want to turn qualify off you could play with the qualify
> times. I did a bunch of this before I just gave up.
>
> I'm sure there is a better or proper way of handling this. I'm
> interested to hear it.
>
> Paul
>
>
>
> On No
-12619
after a while the whole system will become unresponsive
until i kill the asterisk process.
i've checked our network switches/routers and connections.
they all work fine without any packet lost.
any suggestions?
--
Edwin Lam
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988
used. will the 121 person able to make
calls? will it support more channels if i put 2 cards in the system?
thanks.
--
Edwin Lam
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0
ges:
= a number, 0 to 59, inclusive
so. look back at your logic, your welcome-morning message will
be played before 12:01. and your welcome-afternoon message will
be played on & after 12:01.
--
Edwin Lam
Systems Engineer, Office General, Inc.
Ph: +1 415 439
macs is a nice operating system, but it lacks a decent editor. :-P
forget about all these editors.. TECO is the Sh*t.
--
Edwin Lam
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20
_
i've been playing with 1.6 voicemail w/ IMAP storage. it
seems to work fine. however once IMAP storage is enabled.
everyone VM will use IMAP. is there a way to configure
some users use IMAP and other users use traditional
file base storage?
--
Edwin Lam
Systems Engineer, Office General
correct if you run asterisk as non-root.
--
Edwin Lam <[EMAIL PROTECTED]>
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20
___
-- Bandwidth and
characters.
>
> It still matches a blank callerid(num).
try:
exten =>s,n,Set(CALLERID(name)=${IF($[ ${REGEX("021245711.."
0${CALLERID(num)})} = 1] ? "Office":${CALLERID(name)})})
--
Edwin Lam <[EMAIL PROTECTED]>
Systems Engineer, Office General, Inc.
Ph: +1 415
arm. just tell them you can't get any
clocking signal. they'll probably send someone on site and test
the line.
p.s. note that T1/E1 crossover cable pin out is not the same
as ethernet crossover cable.
--
Edwin Lam <[EMAIL PROTECTED]>
S
ject_get+0xf/0x13
> [] __dentry_open+0xea/0x1ab
> [] nameidata_to_filp+0x19/0x28
> [] do_filp_open+0x2b/0x31
> [] do_ioctl+0x47/0x5d
> [] vfs_ioctl+0x24a/0x25c
> [] __fput+0x13f/0x167
> [] sys_ioctl+0x48/0x5f
> [] syscall_call+0x7/0xb
> =
different. they never have
this kind of problems.
--
Edwin Lam <[EMAIL PROTECTED]>
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20
___
-- Bandwidt
ll this work for you? http://sourceforge.net/projects/iaxmodem
--
Edwin Lam <[EMAIL PROTECTED]>
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20
___
aiting=yes
you may need to add
usecallingpres=yes
in zapata.conf
and also add
exten => _9.,n,SetCallerPres(allow)
before the Dial command in extensions.conf.
--
Edwin Lam <[EMAIL PROTECTED]>
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415
Doug Lytle wrote:
> Edwin Lam wrote:
>> in FaxDispatch:
>>
>> FILETYPE=pdf
>> case "$CALLID4" in
>>1000)
>> [EMAIL PROTECTED]
>>1001)
>> [EMAIL PROTECTED]
>>*)
>> [EMAIL PROTECTED]
>> esac
&
x27;ve found $CALLID4
should have the DID info. but in fact it's blank.
any idea?
--
Edwin Lam <[EMAIL PROTECTED]>
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20
terms of analog is local loop, Do we need to have echo cancel in this
>> scenario ?
--
Edwin Lam <[EMAIL PROTECTED]>
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20
i have Asterisk 1.4.18, SpanDSP 0.0.4pre16, AGX addons 1.4.5
linux kernel 2.6.18 AMD64. it (Asterisk) segfault on rxfax
when i enable faxdetect in zapata.conf. since then it disabled
faxdetect and use nvfaxdetect function in dialplan, it works
fine afterward.
also it seems to works fine using
enter the CLI and type "reload", it resets to "off"
> again. I've tried setting the clearglobalvars=no as well as just
> commenting out that line, but no luck so far.
>
> Any ideas?
we use MySQL db to store those global vars in our installation. i
guess you
ing i'm missing when
upgrading to 1.4?
--
Edwin Lam <[EMAIL PROTECTED]>
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20
___
-- Bandwi
use rx_fax.
--
Edwin Lam <[EMAIL PROTECTED]>
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20
___
-- Bandwidth and Colocation Provided by http:/
WARNING[8513]: app_voicemail.c:6281 vm_authenticate:
> Couldn't read username
> Really destroying SIP dialog '[EMAIL PROTECTED]' Method:
> BYE
>
> So it plays the greetings, and is working, I just cant hear it.
what's your voicemail.conf looks like?
hi guys.
i'm writing some simple applications for the cisco 7970
services button. i read the asterisk wiki and it mention
there's a CMXML_App_Guide.pdf file but there's nowhere
can i find a link for it. does anybody know where can
i find it?
regards.
--
Edwin Lam <[EMAIL P
ndle this through tech support.
yes. i did.
the problem solved by getting the latest snapshot with subversion.
--
Edwin Lam <[EMAIL PROTECTED]>
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/loo
:
loadzone=cn
defaultzone=cn
span=1,1,0,ccs,hdb3
span=2,0,0,esf,b8zs
bchan=1-15
dchan=16
bchan=17-31
fxoks=32-55
any clues?
p.s. the same setting works fine on HP Proliant server.
--
Edwin Lam <[EMAIL PROTECTED]>
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
ht
=
> nat=yes
i believe you also need to set nat=no.
on my setup (i'm using asterisk 1.2.24) it works with
both qualify=yes or no.
--
Edwin Lam <[EMAIL PROTECTED]>
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/
true
--
Edwin Lam <[EMAIL PROTECTED]>
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20
___
--Bandwidth and Colocation Provided by h
at
decent speeds (minimum of 28.8) that anyone knows of? If not, has
anyone used a Digium FXS card for this?
i've done:
PRI -> Asterisk -> SIP to analog adapter -> modem or fax
and
PRI -> Asterisk -> channel bank -> modem or fax
they both work fine.
--
Edwi
sions of various components:
asterisk: 1.2.7.1, zaptel: 1.2.5, libpri: 1.2.2
any clues would be appreciated?
--
Edwin Lam <[EMAIL PROTECTED]>
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20
synews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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Edwin Lam <[EMAIL PROTECTED]>
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/
x27;t work.
(maybe it's fixed in the newest firmware)
--
Edwin Lam <[EMAIL PROTECTED]>
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20
___
ckage did you installed?
make sure your /tftpboot directory and all the files inside is at
least readable by everyone.
--
Edwin Lam <[EMAIL PROTECTED]>
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=get&se
Eric "ManxPower" Wieling wrote:
rename bootrom.ld to something else like bootrom.ld-disabled.
did that. it hung on sip.ld, rename sip.ld, it hung on
phone1.cfg. seems like if the file is bigger than say 1k.
it'll hang.
--
Edwin Lam <[EMAIL PROTECTED]>
Systems Engineer
Carla Schroder wrote:
On Monday 23 October 2006 17:38, Edwin Lam wrote:
Re: [asterisk-users] Polycom SP4000 ftp problem
From: Edwin Lam <[EMAIL PROTECTED]>
To: asterisk-users@lists.digium.com
Carla Schroder wrote:
Sooo...stick with tftp? :) Seriously, that's what it's
provide the security and flexibility
of ftp server.
--
Edwin Lam <[EMAIL PROTECTED]>
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20
___
--B
ip 2.0.1 but it makes no
difference. any thoughts?
p.s. i'm using debian sarge proftpd 1.2.10 and the setting works
fine w/ SP501 with bootrom 3.1.2/sip 1.6.3
--
Edwin Lam <[EMAIL PROTECTED]>
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.m
explain to me how to use
these parementers or point me to some documentations on them?
thanks.
--
Edwin Lam <[EMAIL PROTECTED]>
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=get&a
hi folks.
does anybody know what's the phone number for SBC Nothern
California's 102-type milliwatt test line? (specifically
in 415 area code)
--
Edwin Lam <[EMAIL PROTECTED]>
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.m
eceived
channel => 1-15
channel => 17-31
any clues?
--
Edwin Lam <[EMAIL PROTECTED]>
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20
___
databases:
Master.csv - cdr_custom.conf
Mysql - cdr_mysql.conf
odbc - cdr_odbc.conf
postgreSQL - cdr_pgsql.conf
FreeTDS - cdr_tds.conf
--
__ Edwin Lam <[EMAIL PROTECTED]> __
__ Systems Engineer, Office General, Inc.
__ Ph: +1 415 439 4988 Fax:
2.
however. i switched over to use realtime SIP. now the voicemail
light doesn't work. also has anyone use the MailboxExists() function
in dial plan? seems like no matter what i do. it'll just just execute
the next proprity. :(
--
Edwin Lam <[EMAIL PROTECTED]>
Systems Engineer, Offic
PA-941. finally
they gave me a phone number to call, which appears to be a fax
machine. that's when i gave up on those idiots.
--
Edwin Lam <[EMAIL PROTECTED]>
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/
;' and destination '', then the user at
extension '' transfer the call to ''. the CDR always
show '' as the src and '' as the dst. is there any way
to change this behavior and have '' as the src?
--
Edwin La
' exten and not priority:
exten => h,1,blah
yeah. i figured that.
but that would execute on everything in the context. Someone else
suggested the g option on Dial.
that might work better. i'll have to experiment on it.
thanks.
--
Edwin Lam <[EMAIL PROTECTED]>
Systems Enginee
et
executed.
--
Edwin Lam <[EMAIL PROTECTED]>
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20
___
--Bandwidth and Colocation sponsored by Easynews.c
e with 2 Cisco phones & 8 Grandstream phone
before, until i replaced the Grandstreams with Ciscos. the following is
typical setting in sip.conf:
[1234]
context=default
type=friend
host=dynamic
username=1234
secret=test123
mailbox=1234
callerid="John Smith" <1234>
qualify=yes
dtmf
upply. we're not using
Cisco switches but the server & all the phone units are on the same
vlan and i've set the QOS priority to highest on our switch for
that vlan.
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Edwin Lam <[EMAIL PROTECTED]>
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
htt
TED] for seqno 102
(Non-critical Request)
--
Edwin Lam <[EMAIL PROTECTED]>
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20
___
--Bandwidth and
6.12, strange
thing is the system works fine with 2 Cisco phones & 8 Grandstream phone
before, until i replaced the Grandstreams with Ciscos. the following is
typical setting in sip.conf:
[1234]
context=default
type=friend
host=dynamic
username=1234
secret=test123
mailbox=1234
callerid="Joh
B rings ( to tell someone is
in wait : A)
So it seems to fail
i believe the sequence should be:
A call B
B pushes flash button (A hears mp3)
B calls C pressing send
C answers
B press transfer (B will hangup, A & C is now connected)
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Edwin Lam <[EMAIL PROTECTED]>
Systems Engine
se numbers to E1 lines w/ DID seems out of the question.
i guess the way to go is using channel banks to convert those to E1 then
connect Asterisk that way.
further research, how about using these:
http://www.welltech.com.tw/product_e_03.htm
will that work?
--
Edwin Lam <[EMAIL PROTECTED]>
PCI slots) link
them together with some insane dialplan? or is there an easier way?
any suggestions? comments? remarks? parameters?
thx.
--
Edwin Lam <[EMAIL PROTECTED]>
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=
does anybody has experienece with Sipura SPA-841 phone unit?
how's its sound quality especially speaker phone? i have several
Grandstream phones and was getting fustrated about the quality
and bugs of their firmware.
--
Edwin Lam <[EMAIL PROTECTED]>
Systems Engineer, Office General,
lso i have to set emdigitwait=1000
to get it to work reliably.
--
Edwin Lam <[EMAIL PROTECTED]>
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20
___
ts). some time if i
dialed the 4 digits very fast it might get through. seems like there's a timming
issue of the DTMF. what can i do to solve this? i looked through the docs
for zaptel.conf & zapata.conf and doesn't seems there's any parameters to
control the DTMF timing.
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