d by 10.0.0.2 (format ulaw)
-- Format for call is ulaw
-- IAX2/800-2 is ringing
-- IAX2/800-2 answered SIP/10.0.0.1-bc0039a0
-- Hungup 'IAX2/800-2'
== Spawn extension (incoming, 2800800, 1) exited non-zero on
'SIP/10.0.0.1-bc0039a0'
Why can it be? I use last sv
o such host: sipnet-in
May 11 13:01:00 WARNING[1091873088]: chan_sip.c:7463 transmit_register:
Probably a DNS error for registration to [EMAIL PROTECTED], trying
REGISTER again (after 20 seconds)
Why callweaver use sipnet-in as host name?
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> That code sip is correct, dunno why is doesn't work.
> It can be either a hostname or a section name for that provider.
> Could be a bug... fill a ticket for that bug.
http://trac.openpbx.org/cgi-bin/trac.cgi/ticket/290
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t=sipnet-incoming
allow=alaw
allow=ulaw
nat=no
canreinvite=no
dtmfmode=info
# cat extensions.conf
[sipnet-incoming]
exten => s,1,Answer()
exten => s,2,Playback(hello-world)
exten => s,3,Hangup()
On calling I got:
May 11 14:09:13 NOTICE[1091873088]: pbx.c:2115 pbx_extension_helper:
Cannot find extension context 'incoming'
So, how can I redirect call to context sipnet-incoming?
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>>I use this in sip.conf:
>>
>>[general]
>>...
>>register => login:[EMAIL PROTECTED]
>>...
>>On starting I got:
>>
>>*CLI> May 11 13:01:00 WARNING[1091873088]: chan_sip.c:2736 create_addr:
>>No such host: sipnet-in
>>May 11 13:01:00 WARNING[1091873088]: chan_sip.c:7463 transmit_register:
>>Probab
Hi,
What about using Proc instead of Macro in AEL? Trivial patch is attached.
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--- pbx/pbx_ael.c.orig 2007-05-13 18:58:16 +0400
+++ pbx/pbx_ael.c 2007-05-13 18:57:08 +0400
@@ -2931,7 +2931,7 @@ void gen_prios(struct ael_extension *ext
strcat(buf1
2.15:9002 (len 6)
Got UDPTL packet from 192.168.2.15:9002 (len 6)
Got UDPTL packet from 192.168.2.15:9002 (len 6)
Sent UDPTL packet to 192.168.2.15:9002 (seq 1, len 8)
...
Sent UDPTL packet to 192.168.2.15:9002 (seq 167, len 33)
-- Executing NoOp("SIP/801-bb34", "RX: RE
On saving message I got:
-- Executing VoiceMail("SIP/sipnet.ru-00645710", "[EMAIL PROTECTED]") in
new stack
May 14 13:30:48 WARNING[1092139328]: app_voicemail.c:2345
leave_voicemail: No entry in voicemail config file for '101'
What's wrong?
--
Thanks,
So, fax is not received. What's wrong?
Nobody knows.
See: checklist and troubleshooting.
http://wiki.openpbx.org/tiki-index.php?page=Faxing%20with%20Openpbx.org
Gzipped full log is attached. Can anybody say why all retries are not
successfull?
--
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fu
> Don't really know if voicemail works with sqlite but extconf is wrong.
>
>
>># cat extconfig.conf
>>voicemail => mysql,callweaver,voicemail_users
>
>
> this only work with mysql. Perhaps something like:
>
> voicemail => sqlite,callweaver,voicemail_users
Thanks. But after replacing all works
many other softswitches. But this problem is
reproducable with Asterisk 1.4.4 running without ztdummy and not
reproducable while ztdummy is loaded. CallWeaver can't use ztdummy, so
it can't work with HuaweiSoftX3000.
How to resolve this problem?
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<-- SI
how to resolve this problem?
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This problem is reproducable with CONFIG_HZ_1000=y and CONFIG_HZ_250=y :(
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The same problem is decribed in http://bugs.digium.com/view.php?id=6669,
so I rewrite patch for current CallWeaver and now all works fine :)
Patch is attached. Is it possible to include it in SVN?
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--- channels/chan_sip.c.orig 2007-05-09 18:09:28 +0400
+++ channels
http://www.callweaver.org/ticket/58
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Hi,
Is it possible to use TxFax/RxFax to send/receive faxes over voice
instead of T.38?
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Hi,
Is it possible to use TxFax/RxFax for sending/receiving fax on
established connection after Dial and before Hangup? Can anybody give me
dialplan example?
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shing connection first user push some button on
phone (softphone or phone via sip gateway) and after this TxFax sends
fax from some file to second user which must start fax receiving on
hardware fax.
How can I implement this?
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; h,1,NoOp(RX: REMOTESTATIONID is ${REMOTESTATIONID})
exten => h,n,NoOp(RX: PHASEESTATUS is ${PHASEESTATUS})
exten => h,n,NoOp(RX: PHASEESTRING is ${PHASEESTRING})
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Hi,
Why I can't hear music on hold? On console I see Stopped music already
after Started music. My configuration:
# cat musiconhold.conf
[default]
mode=files
random=yes
directory=/var/lib/asterisk/mohraw
application=/usr/bin/rawplayer
# ls /var/lib/asterisk/mohraw
1.raw
# cat extensions.conf
Hi,
Is it possible to use any connection pooling instead of
Connect/Query/Fetch/Clear/Disconnect on every database operation?
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end[5221]: SEND FAX: JOB 14 SENT in 0:26
CallWeaver application TxFax works too, but I need HylaFAX client tools.
I can't replace it with AsterFax because it has very bad design and very
strange dependencies. I don't wish to use Asterisk instead of CallWeaver to.
Which trace files or some
end[5221]: SEND FAX: JOB 14 SENT in 0:26
CallWeaver application TxFax works too, but I need HylaFAX client tools.
I can't replace it with AsterFax because it has very bad design and very
strange dependencies. I don't wish to use Asterisk instead of CallWeaver to.
Which trace files or some
> You may need to run two commands
>
> iaxmodem ttyIAX
> faxgetty ttyIAX
>
> Please also check at /var/log/iax/ and /var/log/messages for any hint.
I run iaxmodem ttyIAX in non daemon mode, so I see all info on console
instead of /var/log/iax
Contents of /var/log/messages I show in my pre
> The errors who reported have nothing to do with Callweaver. They are
> errors from your iaxmodem/hylafax setup/. These are completely separate
> programs, that communication with Callweaver by TCP/IP, and dot not
> share any configuration information with it.
I agreed but all begins to work a
:
...
-- Executing Set("SIP/10.0.0.1-0061f5d0", "CDR(userfield)=2422718")
-- Executing Dial("SIP/10.0.0.1-0061f5d0", "SIP/708,25,tT")
...
to file? I can't find any suitable option in logger.conf
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Hi,
I can setup verbose default value in asterisk.conf in [options] section.
Is it possible to do the same with CallWeaver? I need this because
console started with -R option don't save verbose value after reconnecting.
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so the main part
if my installation is monit :( I hope it will be fixed in 1.2 release.
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> There's only 9 open tickets
Why do say about only 9 tickets? I see 22 open tickets for 1.2 release
on http://callweaver.org/tickets, which displays 50 tickets
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What's wrong?
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Hi,
Is it impossible now to use database via odbc in CallWeaver? I can't see
odbc option in configure script.
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| NO ANSWER | 1 | 0
(8 rows)
Who is anonymous, +0, +00 and C?
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Max CtRiX пишет:
> Thomas Kenyon ha scritto:
>> I compiled from source, configured with these options:
>>
>> --enable-builtin-sqlite3 --enable-postgresql --enable-srtp
>> --enable-zaptel --enable-iax-trunking --enable-t38 --with-chan_fax
>> --with-chan_bluetooth --with-cdr_pgsql --with-res_config_p
Hi,
Why can I see calls which was finished about 1 hour or more ago? I use
last branch 1.2.
pbx*CLI> show channels verbose
Channel Context ExtensionPrio State
Application CallerIDDuration Accountcode BridgedTo
SIP/outbound-stc-5a9 default
and via DLink VoIP gateway DVG-2032S. G.711 alaw without T.38
is using.
Whay can it be?
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Hi,
I use hardware fax machines via dlink voip gateways and TxFax with
G711, not T.38. Some times ago fax machines and TxFax worked fne, but
now only fax machines works. With TxFax I got PHASEESTATUS 13
(Unexpected message received) every time. Now I use last 1.2 version
(svn revision 4249) and s
u provide logs of onw or two failed calls?
I'll send tcpdumps for G711 and T.38 faxing after upgrading to last
CallWeaver 1.2 and last SpanDSP tomorrow.
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of anything which
> > might have broken. Can you provide logs of onw or two failed calls?
>
> I'll send tcpdumps for G711 and T.38 faxing after upgrading to last
> CallWeaver 1.2 and last SpanDSP tomorrow.
see http://www.callweaver.org/ticket/299
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see http://www.callweaver.org/ticket/302 for TxFax and fax machine
dumps via G711
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Hi,
There are no activity in BTS at least for faxing related problems and
no letters in this maillists from Steve Underwood, Roy Sigurd
Karlsbakk, Massimo Cetra and some other developers.
Is CallWeaver development alive?
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> How alive do you expect it to be?
I expect answers to try last callweaver/spandsp versions in BTS :)
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> How alive do you expect it to be?
What about bug #299? It is reproducable with svn 4498 and last
spandsp. Why one fax is successful (but image was negative) and
another one is fail?
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http
INVITE to [EMAIL PROTECTED] and CallWeaver says "404 Not
Found".
Is it possible to not send Contact: and
redirect any incoming call from Hiawei SoftX3000 to context incoming
without any checking?
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> I read a bit online about this issue, but it's not clear as to the current
> situation. I understand that older CW versions managed to compile with
> G.729, but I wonder if someone has a working solution to integrate G.729
> into CW RC5.
http://asterisk.hosting.lv/
__
> Is there a clear/detailed online manual that explains how to setup SIP
> buddies ,voicemail... through MySQL tables on CW?
http://www.callweaver.org/wiki/Realtime+SIP+Howto
As far as I know real-time for voicemail is not implemented :( It was
implemented in Asterisk for ODBC, but ODBC is suppor
Hi,
Are recomendations from http://www.callweaver.org/wiki/Kernel still valid?
I can't see warnings with kernel 2.6.18 with CONFIG_HZ=250 and I can't
see timer.c via http://www.callweaver.org/browse. It's looks like
CallWeaver works without significant problems.
__
Last Message
192.168.101.200 104 901db4f8-36 00101/2 alaw No
Rx: ACK
192.168.101.9103 d4769c9f-37 00101/2 alaw No
Rx: ACK
2 active SIP channels
Why can it be?
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Call
2008/7/31, Rob Gillan :
> Eugene,
>
> Have a look at the tickets on www.callweaver.org. It's a known
> problem that Max is going to fix at some point. If you run the
> conferences in quiet mode (Mq, Sq) they work without announcements.
Thanks. What is the bug number?
_
Hi,
Is it possible to set canreinvite=yes only for local peers and set
canreinvite=no for PSTN peers?
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Is it possible to store VoiceMail accounts with Realtime instead of
flat configuration file?
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Hi,
I need to replace SIP header from:
From: "101" ;tag=as3d2399f9
to:
From: ;tag=as3d2399f9
as described in http://www.softarmor.com/wgdb/docs/draft-mahy-iptel-cpc-00.txt.
Is it possible to implement this feature? Can do It with existing
applications/functions?
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ng like:
-- Executing [2440...@local:1] Set("SIP/102-fff4", "SIP_HEADER(From")
Feb 4 09:13:28 WARNING[1092921664]: pbx.c:7024 pbx_builtin_setvar:
Ignoring entry 'SIP_HEADER(From' with no '=' (and not last 'options'
entry)
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ous gcc version
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2009/5/21, Max CtRiX :
> Eugene Prokopiev ha scritto:
>
>
> > libtoolize-default: copying file `libltdl/libltdl/lt_dlloader.h'
> > libtoolize-default: copying file `libltdl/libltdl/lt_error.h'
> > libtoolize-default: copying file `libltdl/libltdl/lt_sy
Hi,
$ svn co http://svn.callweaver.org/callweaver/branches/rel/1.2/
svn: OPTIONS of
'http://svn.callweaver.org/callweaver/branches/rel/1.2': could not
connect to server (http://svn.callweaver.org)
When svn.callweaver.org becomes available?
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wrong in my configuration?
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