[Asterisk-Users] sipsak with asterisk
i am using sipsak to test asterisk. i use the command $ sipsak -U -I -s sip:[EMAIL PROTECTED] -vv -x 1800 -e3 and i get the message SIP/2.0 407 Proxy Authentication Required as a response to INVITE message (REGISTER was successful) and error: could not find To in the reply does anyone have some idea as to what is missing ? tulika _ Kareena or Rani? Saif or SRK? http://server1.msn.co.in/sp05/iifa/ Rock your vote now at IIFA. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can the originator of a call transfer it?
See below. Add a T to your Dial options. CLI show application Dial -= Info about application 'Dial' =- ... 't' -- allow the called user to transfer the calling user by hitting #. 'T' -- allow the calling user to transfer the call by hitting #. --Luki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] spandsp configuration
Hi, I receive fax using spandsp. It works, however the tif file it stored has no good quality. Any method to configure that? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to decrease Asterisk load
On 5/12/05, Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote: Digium cards do not have a built in DSP. Neither do the Sangoma as far as I know. I don't know about VoiceTronix. As yet ! As for digium cards latest cvs commits suggest that there is some ongoing development on hardware based echo cancelation and dtmf detection. So its just a matter of time. rgrds m. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sound card Line-In as MOH source
On Thu, 12 May 2005, Chris Coulthurst wrote: Does someone have a link to step-by-step instructions to making the Line-In on the console sound card a MOH source? You can probably use the Remote MoH patch from http://bugs.digium.com/view.php?id=3565 Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 1-800 free calls
How do you dial 1800 number using FWD? Ive tried (fwd prefix) 1800numberblah and I get congestion.. This is information clearly stated on the FWD site. *1800 as in http://www.freeworlddialup.com/advanced/peering_numbers ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UNREACHABLE messages
I get these on a consistant basis for most of the providers I have configured. Some less than others. I even get it from my phone at home to my * box at our data center. So do I, on both our DSL connections. It's a result of the connectivity of the ISP network (or the rest of the internet sometimes). I usually use qualify=300. Most providers will drop out once in a while each day for a few seconds. Some become unreachable more often such as several times an hour. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_capi, chan_misdn and chan_modem
Could someone please comment on the current state of chan_capi, chan_misdn and chan_modem channel drivers in terms of functionality (echo cancelation, fax, latency etc) and stability. Specifically, which channel driver would be best for a passive PCI HFC or W6692 ISDN card. The chan_misdn wiki claims that chan_capi distinguishes between junghanns and non-junghans cards, and that chan_misdn is better suited for general misdn compatibility. A second point I'd like some clarification on is the purpose of Junghann's BRIStuff patch. Is this patch only necessary for chan_capi or also for chan_misdn? Does this patch add functionality to asterisk or is it only intended to smooth chan_capi integration into asterisk? Thanks in advance! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] French SIP or IAX phones
On Thu, 2005-05-12 at 19:28 -0400, Nabeel Jafferali wrote: I have a customer that's located in France and he wants french phones if possible. So I'm wondering if there's any one out there that found a phone that can be change to french. I believe snom phones have the option. Confirmed -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] French SIP or IAX phones
Polycom will do the trick.. On Fri, 13 May 2005 09:00:35 +0200, Dave Cotton [EMAIL PROTECTED] wrote: On Thu, 2005-05-12 at 19:28 -0400, Nabeel Jafferali wrote: I have a customer that's located in France and he wants french phones if possible. So I'm wondering if there's any one out there that found a phone that can be change to french. I believe snom phones have the option. Confirmed -- Using Opera's revolutionary e-mail client: http://www.opera.com/mail/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Problem with MeetMe
In article [EMAIL PROTECTED], Daniel Salama [EMAIL PROTECTED] wrote: I agree. I also prefer 2.6.x. That's what I run on my Debian machines. However, REL3 does not support 2.6.x. They don't even have an RPM for it. I would have to get one from kernel.org and that would void any support Red Hat may provide. Upgrade to RHEL4? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] chan_capi, chan_misdn and chan_modem
I battled with chan_capi during the week, and it was not fun. PaulH -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jan Louw Sent: Friday, 13 May 2005 4:46 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] chan_capi, chan_misdn and chan_modem Could someone please comment on the current state of chan_capi, chan_misdn and chan_modem channel drivers in terms of functionality (echo cancelation, fax, latency etc) and stability. Specifically, which channel driver would be best for a passive PCI HFC or W6692 ISDN card. The chan_misdn wiki claims that chan_capi distinguishes between junghanns and non-junghans cards, and that chan_misdn is better suited for general misdn compatibility. A second point I'd like some clarification on is the purpose of Junghann's BRIStuff patch. Is this patch only necessary for chan_capi or also for chan_misdn? Does this patch add functionality to asterisk or is it only intended to smooth chan_capi integration into asterisk? Thanks in advance! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: HELP: ASTCC (AGI) meets call forward ERROR
On 5/12/05, Charles Wang [EMAIL PROTECTED] wrote:
Hi, ALL:
When I use astcc to do the prepaid function, but if I want to enable
call forward.
The result of CDR seems not correct.
UA 1011 make a call to UA , and UA forwards this call to a PSTN
number.
I think we shall charge the credit from UA not UA 1011 because UA
1011 don't know where UA forwards to.
But in CDR, I can only find the from(1011) and destination(PSTN number).
I can't find UA from this CDR so I can't charge to UA .
It seems unreasonable.
I use asterisk -r and sip debug to debug my sip channel.
And I build my sip proxy(5060) and asterisk(5065) on the same machine.
I make a call from 1011 to on sip proxy,
sip proxy forwards this call to 0939749001.
And this 0939749001 will be sent to asterisk by sip proxy.
sip ua(1011) = sipproxy = sip ua ( call forward 0939749001)
||
== asterisk === cisco 5300 ==
0939749001 (pstn)
I can find $EXTEN is equal to 0939749001 ( a mobile phone number )
and my $CALLERIDNUM is 1011
But how can I get the value of from To: field? ( via this sip ua)
In another word, I want to record the middle man.
My extensions.conf :
exten = _.,1,Answer
exten = _.,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN})
exten = _.,3,Hangup
My log on asterisk CLI:
-- Executing DeadAGI(SIP/61.220.xxx.xxx-081888c8,
astcc.agi|1011|0939749001|4) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/astcc.agi
ser*CLI
-- SIP read from 61.220.xxx.xxx:5060:
ACK sip:[EMAIL PROTECTED]:5065 SIP/2.0
Record-Route: sip:61.220.xxx.xxx;ftag=915860198;lr=on
Via: SIP/2.0/UDP 61.220.xxx.xxx;branch=0
Via: SIP/2.0/UDP
220.134.18.190:47286;rport=47286;branch=z9hG4bKB90B5F6F80294C48AACF7BDE31B9D2F1
From: 1011 sip:[EMAIL PROTECTED];tag=915860198
To: sip:[EMAIL PROTECTED];tag=as1c0a7e38=== I want to get this value
Contact: sip:[EMAIL PROTECTED]:47286
Call-ID: [EMAIL PROTECTED]
CSeq: 57194 ACK
Max-Forwards: 16
Content-Length: 0
--
Best Regards
Charles
--
Best Regards
Charles
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[Asterisk-Users] About Voip Technology : RTP over TCP
hello All I am reading information about VoIP technology For that i am concentrating on SIP (Session Initiation Protocol) and RTP (Real Time Transport Protocol). I am interested in implementing RTP over TCP I found that there are some disadvantages of TCP, some are 1) TCP doesn't support multicasting. 2) Through TCP is reliable, it heavily depends on retransmission of lost or corrupted packets. That increases Delay. Our aim is real time delivery of packets not reliability. 3) TCP congestion control mechanism decreases the congestion window when packet losses are detected. Apart from these flaws i am interested in implementing RTP over TCP Is it possible ? What is feasibility of implementing RTP over TCP ? One more point is What is feasibility of implementing RTP over TCP in case of NAT (Network Address Translation) is there ? Need your valuable suggestion. Thank You ! Atul Thosar ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] chan_capi, chan_misdn and chan_modem
On Fri, 13 May 2005, Paul Hales wrote: I battled with chan_capi during the week, and it was not fun. Since I'm working on chan_capi, I would like to know what problems exist. Can you please be more specific on what problems you have encountered? Armin PaulH -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jan Louw Sent: Friday, 13 May 2005 4:46 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] chan_capi, chan_misdn and chan_modem Could someone please comment on the current state of chan_capi, chan_misdn and chan_modem channel drivers in terms of functionality (echo cancelation, fax, latency etc) and stability. Specifically, which channel driver would be best for a passive PCI HFC or W6692 ISDN card. The chan_misdn wiki claims that chan_capi distinguishes between junghanns and non-junghans cards, and that chan_misdn is better suited for general misdn compatibility. A second point I'd like some clarification on is the purpose of Junghann's BRIStuff patch. Is this patch only necessary for chan_capi or also for chan_misdn? Does this patch add functionality to asterisk or is it only intended to smooth chan_capi integration into asterisk? Thanks in advance! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] GXP 2000 Conference Button and ILBC
The unofficial GXP-2000 resource, bugs, and information page is at http://www.aussievoip.com.au/wiki-GXP-2000 --Rob -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Thursday, May 12, 2005 5:53 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] GXP 2000 Conference Button and ILBC Guys. I just downloaded the recent firmware for GS GXP 2000 and I must say the phone works great but... How do you make the conf button work?? Anybody done that? Also, with great dissapointment I must ask, where is ILBC support? GS web page mentions it and the manual says it supports it almost using bolds :) soo where is it Any light on this? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX to FWD?
On Thu, 2005-05-12 at 12:40 -0600, Tim Pushor wrote: I had trouble calling people who were using FWD/SIP from my FWD/IAX account. I switched back to using SIP and could call SIP users, but not IAX users. I've since de-registered myself for the IAX *beta* and can now talk to everyone again. I noticed something similar. My Asterisk box just uses FWD:SIP. I have two hardware capable IAX phones, couldn't get SIP to work on them (NAT problems) so tried IAX which worked fine from the phones to my box, but they could not call each other.. The IAX phones are in different countries/continents on different ADSL services (Parents, Brother - etc) Assumption - two IAX devices both registered at FWD can not talk to each other. ps - how does one set up a Proxy - so machines on foreign NATs can talk? -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: oh323 compile problem in FreeBSD
Following is the errors when I tried to compile oh323 in FreeBSD 5.3.
Asterisk is updated from cvs.
asterisk# gmake
for x in wrapper asterisk-driver; do gmake -C $x build || exit 1 ; done
make: illegal option -- -
usage: make [-BPSXeiknqrstv] [-C directory] [-D variable] [-d flags]
[-E variable] [-f makefile] [-I directory] [-j max_jobs]
[-m directory] [-V variable] [variable=value] [target ...]
make: illegal option -- -
usage: make [-BPSXeiknqrstv] [-C directory] [-D variable] [-d flags]
[-E variable] [-f makefile] [-I directory] [-j max_jobs]
[-m directory] [-V variable] [variable=value] [target ...]
gmake[1]: Entering directory `/usr/home/tsgan/asterisk-oh323-0.7.1/wrapper'
./check_ver /usr/home/tsgan/pwlib pwlib
./check_ver /usr/home/tsgan/openh323 openh323
g++ -Wall -x c++ -Os -DWRAPTRACING -DWRAPTRACING_LEVEL=5
-DPWLIBVERSION=\1.6.6\ -DOPENH323VERSION=\1.13.5\
-I/usr/home/tsgan/pwlib/include/ptlib/unix
-I/usr/home/tsgan/pwlib/include -I/usr/home/tsgan/openh323/include
-I/usr/home/tsgan/openh323/include/openh323 -I../asterisk-driver -c
wrapper_misc.cxx -o wrapper_misc.o
In file included from /usr/home/tsgan/pwlib/include/ptlib.h:169,
from wrapper_misc.hxx:35,
from wrapper_misc.cxx:34:
/usr/home/tsgan/pwlib/include/ptlib/unix/ptlib/pdirect.h:78: error:
expected unqualified-id before protected
/usr/home/tsgan/pwlib/include/ptlib/unix/ptlib/pdirect.h:78: error:
expected `,' or `;' before protected
/usr/home/tsgan/pwlib/include/ptlib/unix/ptlib/pdirect.h:80: error:
expected constructor, destructor, or type conversion before '*' token
/usr/home/tsgan/pwlib/include/ptlib/unix/ptlib/pdirect.h:80: error:
expected `,' or `;' before '*' token
In file included from /usr/home/tsgan/pwlib/include/ptlib.h:181,
from wrapper_misc.hxx:35,
from wrapper_misc.cxx:34:
/usr/home/tsgan/pwlib/include/ptlib/unix/ptlib/config.h:53: error:
expected unqualified-id before public
/usr/home/tsgan/pwlib/include/ptlib/unix/ptlib/config.h:53: error:
expected `,' or `;' before public
/usr/home/tsgan/pwlib/include/ptlib/unix/ptlib/config.h:55: error:
expected class-name before '(' token
/usr/home/tsgan/pwlib/include/ptlib/unix/ptlib/config.h:55: error:
expected `,' or `;' before '(' token
/usr/home/tsgan/pwlib/include/ptlib/unix/ptlib/config.h:57: error:
expected unqualified-id before protected
/usr/home/tsgan/pwlib/include/ptlib/unix/ptlib/config.h:57: error:
expected `,' or `;' before protected
In file included from /usr/home/tsgan/pwlib/include/ptlib.h:187,
from wrapper_misc.hxx:35,
from wrapper_misc.cxx:34:
/usr/home/tsgan/pwlib/include/ptlib/args.h:121: error: expected
class-name before '{' token
/usr/home/tsgan/pwlib/include/ptlib/args.h:122: error: `PObject' has
not been declared
/usr/home/tsgan/pwlib/include/ptlib/args.h:122: error: ISO C++ forbids
declaration of `PCLASSINFO' with no type
/usr/home/tsgan/pwlib/include/ptlib/args.h:122: error: ISO C++ forbids
declaration of `parameter' with no type
/usr/home/tsgan/pwlib/include/ptlib/args.h:147: error: expected `,' or
`...' before '' token
/usr/home/tsgan/pwlib/include/ptlib/args.h:153: error: ISO C++ forbids
declaration of `PString' with no type
/usr/home/tsgan/pwlib/include/ptlib/args.h:156: error: expected `,' or
`...' before '' token
/usr/home/tsgan/pwlib/include/ptlib/args.h:162: error: ISO C++ forbids
declaration of `PString' with no type
/usr/home/tsgan/pwlib/include/ptlib/args.h:162: error:
`PArgList::PArgList(int)' and `PArgList::PArgList(int)' cannot be
overloaded
/usr/home/tsgan/pwlib/include/ptlib/args.h:177: error: expected `,' or
`...' before '' token
/usr/home/tsgan/pwlib/include/ptlib/args.h:182: error: ISO C++ forbids
declaration of `PString' with no type
/usr/home/tsgan/pwlib/include/ptlib/args.h:189: error: variable or
field `PrintOn' declared void
/usr/home/tsgan/pwlib/include/ptlib/args.h:189: error: `PrintOn'
declared as a `virtual' field
/usr/home/tsgan/pwlib/include/ptlib/args.h:189: error: expected `;'
before '(' token
/usr/home/tsgan/pwlib/include/ptlib/args.h:196: error: variable or
field `ReadFrom' declared void
/usr/home/tsgan/pwlib/include/ptlib/args.h:196: error: `ReadFrom'
declared as a `virtual' field
/usr/home/tsgan/pwlib/include/ptlib/args.h:196: error: expected `;'
before '(' token
/usr/home/tsgan/pwlib/include/ptlib/args.h:206: error: expected `,' or
`...' before '' token
/usr/home/tsgan/pwlib/include/ptlib/args.h:207: error: ISO C++ forbids
declaration of `PString' with no type
/usr/home/tsgan/pwlib/include/ptlib/args.h:215: error: expected `,' or
`...' before '' token
/usr/home/tsgan/pwlib/include/ptlib/args.h:216: error: ISO C++ forbids
declaration of `PStringArray' with no type
/usr/home/tsgan/pwlib/include/ptlib/args.h:216: error: `void
PArgList::SetArgs(int)' and `void PArgList::SetArgs(int)' cannot be
overloaded
/usr/home/tsgan/pwlib/include/ptlib/args.h:249: error: expected `,' or
[Asterisk-Users] 2 minutes pause before ring on H323 channel
I have build asterisk from latest CVS HEAD-05/09/05 with H323 support as
described in README file.
Open H.323 version v1.17.1 and PWLib v1.9.0 on Mandrake Linux 10.2 kernel-2.6.11
I tested it with following phones:
-- XLite (SIP softphone)
-- QMix SIP IP phone (PA168F)
-- SJPhone (H323 softphone)
-- QMix H323 IP phone (PA168F)
-- FireFly (IAX2 softphone)
Everything works fine except a problem with h323 extension dialing. Behavior is
the same for both
SJPhone (soft phone) and QMix (PA168F). When I dial such extension I have to
wait 2 minutes
exactly (120 seconds) before extension rings. After long way of trial and
errors with .conf files
I managed to minimize this time to 1 minute exactly (60 seconds)
exten = 20,1,Dial(H323/h323phone) ; this leads to 120 seconds pause before
ring
exten = 21,1,Dial(H323/[EMAIL PROTECTED]) ; this leads to 60 seconds pause
before ring
After quick debugging session I found that this time goes to the call to
H323EndPoint::MakeCallLocked(fullAddress, token, opts) in
MyH323EndPoint::MakeCall function.
MakeCallLocked is part of OpenH323 and this is too deep for me... I'm not sure
is this is a
problem of OpenH323 or of channel driver but my speculation is that the time
goes in some kind of
timeout wait... who knows...
Does anyone else experienced common problems? Any help to resolve the problem
will be appreciated
And here are my .conf files ... very basic yet ;-)
extensions.conf ===
[general]
static=yes
writeprotect=no
[globals]
SIP_XLITE = SIP/xlite
SIP_PHONE = SIP/sipphone
H323_SJPHONE = H323/[EMAIL PROTECTED]
H323_PHONE = H323/[EMAIL PROTECTED]
IAX_FIREFLY = IAX2/firefly
;
; Inbound
;
[inbound]
exten = s, 1, Answer
exten = s, 2, Playback(ss-noservice)
exten = s, 3, Hangup
;
; Internal Extensions
;
[local]
exten = 10,1,Dial(${SIP_XLITE})
exten = 11,1,Dial(${SIP_PHONE})
exten = 20,1,Dial,${H323_SJPHONE}
exten = 21,1,Dial,${H323_PHONE}
exten = 30,1,Dial(${IAX_FIREFLY})
exten = 0, 1, Answer
exten = 0, 2, Playback(ss-noservice)
exten = 0, 3, Hangup
=== h323.conf ===
[general]
port = 1720
bindaddr = 192.168.0.100
disallow = all
allow = ulaw
gatekeeper = DISABLE
context = inbound
[h323phone]
type = friend
host = 192.168.0.101
context = local
callerid = h323phone 21
[sjphone]
type = friend
host = 192.168.0.1
context = local
callerid = sjphone 20
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[Asterisk-Users] [Asterisk-Dev] Re: oh323 compile problem in FreeBSD
Following is the errors when I tried to compile oh323 in FreeBSD 5.3.
Asterisk is updated from cvs.
asterisk# gmake
for x in wrapper asterisk-driver; do gmake -C $x build || exit 1 ; done
make: illegal option -- -
usage: make [-BPSXeiknqrstv] [-C directory] [-D variable] [-d flags]
[-E variable] [-f makefile] [-I directory] [-j max_jobs]
[-m directory] [-V variable] [variable=value] [target ...]
make: illegal option -- -
usage: make [-BPSXeiknqrstv] [-C directory] [-D variable] [-d flags]
[-E variable] [-f makefile] [-I directory] [-j max_jobs]
[-m directory] [-V variable] [variable=value] [target ...]
gmake[1]: Entering directory `/usr/home/tsgan/asterisk-oh323-0.7.1/wrapper'
./check_ver /usr/home/tsgan/pwlib pwlib
./check_ver /usr/home/tsgan/openh323 openh323
g++ -Wall -x c++ -Os -DWRAPTRACING -DWRAPTRACING_LEVEL=5
-DPWLIBVERSION=\1.6.6\ -DOPENH323VERSION=\1.13.5\
-I/usr/home/tsgan/pwlib/include/ptlib/unix
-I/usr/home/tsgan/pwlib/include -I/usr/home/tsgan/openh323/include
-I/usr/home/tsgan/openh323/include/openh323 -I../asterisk-driver -c
wrapper_misc.cxx -o wrapper_misc.o
In file included from /usr/home/tsgan/pwlib/include/ptlib.h:169,
from wrapper_misc.hxx:35,
from wrapper_misc.cxx:34:
/usr/home/tsgan/pwlib/include/ptlib/unix/ptlib/pdirect.h:78: error:
expected unqualified-id before protected
/usr/home/tsgan/pwlib/include/ptlib/unix/ptlib/pdirect.h:78: error:
expected `,' or `;' before protected
/usr/home/tsgan/pwlib/include/ptlib/unix/ptlib/pdirect.h:80: error:
expected constructor, destructor, or type conversion before '*' token
/usr/home/tsgan/pwlib/include/ptlib/unix/ptlib/pdirect.h:80: error:
expected `,' or `;' before '*' token
In file included from /usr/home/tsgan/pwlib/include/ptlib.h:181,
from wrapper_misc.hxx:35,
from wrapper_misc.cxx:34:
/usr/home/tsgan/pwlib/include/ptlib/unix/ptlib/config.h:53: error:
expected unqualified-id before public
/usr/home/tsgan/pwlib/include/ptlib/unix/ptlib/config.h:53: error:
expected `,' or `;' before public
/usr/home/tsgan/pwlib/include/ptlib/unix/ptlib/config.h:55: error:
expected class-name before '(' token
/usr/home/tsgan/pwlib/include/ptlib/unix/ptlib/config.h:55: error:
expected `,' or `;' before '(' token
/usr/home/tsgan/pwlib/include/ptlib/unix/ptlib/config.h:57: error:
expected unqualified-id before protected
/usr/home/tsgan/pwlib/include/ptlib/unix/ptlib/config.h:57: error:
expected `,' or `;' before protected
In file included from /usr/home/tsgan/pwlib/include/ptlib.h:187,
from wrapper_misc.hxx:35,
from wrapper_misc.cxx:34:
/usr/home/tsgan/pwlib/include/ptlib/args.h:121: error: expected
class-name before '{' token
/usr/home/tsgan/pwlib/include/ptlib/args.h:122: error: `PObject' has
not been declared
/usr/home/tsgan/pwlib/include/ptlib/args.h:122: error: ISO C++ forbids
declaration of `PCLASSINFO' with no type
/usr/home/tsgan/pwlib/include/ptlib/args.h:122: error: ISO C++ forbids
declaration of `parameter' with no type
/usr/home/tsgan/pwlib/include/ptlib/args.h:147: error: expected `,' or
`...' before '' token
/usr/home/tsgan/pwlib/include/ptlib/args.h:153: error: ISO C++ forbids
declaration of `PString' with no type
/usr/home/tsgan/pwlib/include/ptlib/args.h:156: error: expected `,' or
`...' before '' token
/usr/home/tsgan/pwlib/include/ptlib/args.h:162: error: ISO C++ forbids
declaration of `PString' with no type
/usr/home/tsgan/pwlib/include/ptlib/args.h:162: error:
`PArgList::PArgList(int)' and `PArgList::PArgList(int)' cannot be
overloaded
/usr/home/tsgan/pwlib/include/ptlib/args.h:177: error: expected `,' or
`...' before '' token
/usr/home/tsgan/pwlib/include/ptlib/args.h:182: error: ISO C++ forbids
declaration of `PString' with no type
/usr/home/tsgan/pwlib/include/ptlib/args.h:189: error: variable or
field `PrintOn' declared void
/usr/home/tsgan/pwlib/include/ptlib/args.h:189: error: `PrintOn'
declared as a `virtual' field
/usr/home/tsgan/pwlib/include/ptlib/args.h:189: error: expected `;'
before '(' token
/usr/home/tsgan/pwlib/include/ptlib/args.h:196: error: variable or
field `ReadFrom' declared void
/usr/home/tsgan/pwlib/include/ptlib/args.h:196: error: `ReadFrom'
declared as a `virtual' field
/usr/home/tsgan/pwlib/include/ptlib/args.h:196: error: expected `;'
before '(' token
/usr/home/tsgan/pwlib/include/ptlib/args.h:206: error: expected `,' or
`...' before '' token
/usr/home/tsgan/pwlib/include/ptlib/args.h:207: error: ISO C++ forbids
declaration of `PString' with no type
/usr/home/tsgan/pwlib/include/ptlib/args.h:215: error: expected `,' or
`...' before '' token
/usr/home/tsgan/pwlib/include/ptlib/args.h:216: error: ISO C++ forbids
declaration of `PStringArray' with no type
/usr/home/tsgan/pwlib/include/ptlib/args.h:216: error: `void
PArgList::SetArgs(int)' and `void PArgList::SetArgs(int)' cannot be
overloaded
/usr/home/tsgan/pwlib/include/ptlib/args.h:249: error: expected `,' or
[Asterisk-Users] Problem with calls on hold
Hello - I recently offloaded some of the SIP traffic on to a separate Asterisk box and interconnected our main Asterisk system with the new system via IAX. The SIP clients are running 7960's. When a call is put on hold, often times when the call is pulled off hold, there seems to be no RTP in at least one direction. There seems to only be voice in one direction. There is usually at least a delay and distortion before voice works in both directions again. Also, when this happens, sometimes I can hear calls that seem to bleed over from other active IAX - SIP sessions. Music on hold is enabled and the other end can hear the music; however, when the call is pulled off, the other end either still hears music or does not hear anything for a short while. Basically the call comes in via a ZAP channel over a PRI into our main system, is fed over IAX to our second system and then is connected to the SIP channel (client). I have tried both enabling and disabling IAX trunking and jitterbuffers. I've also added a zap card (fxo) and enabled it to allow for a timing source. The new system is running the latest CVS of Asterisk and libraries as of a couple days ago, while the other one is running a CVS version as of Jun of last year. I'm using RSA for auth between the servers (IAX). Any help would be appreciated. Thanks. Jeroen ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] beginner in Asterisk configuration
bonjour, tout d'abord merci de bien vouloir m'initier à la configuration de serveur Asterisk. J'expose mon soucis, je tourne sous windows XP, donc g pri asterisk version windows == Astwind 0.1.1 Je ne pe pas changer d'ordinateur. Mon objectif : Lancer mon serveur sans qu'il y aille d'erreur, sans aucun client ! j'ai juste le serveur Quoi que je change dans la configuration ( extensions.conf, iax.conf et interfaces pour le reseau) sa me marque apres execution de safe_asterisk -vcf : Asterisk ended with exit status 1 Asterisk died with code 1 Si vous pouviez me fournir un extensions.conf qui tourne je suis preneur :) voila ma extensions.conf actuel : [general] static=yes writeprotect=no [globals] [local] ignorepat = 9 include = iaxtel700 [iaxtel700] exten = 700,1,Echo() merci beacoup lucas _ MSN Messenger : personnalisez votre messagerie instantanée ! http://g.msn.fr/FR1001/866 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with IAX trunking
Hi all,
I'm trying to get IAX2 trunking between two * boxes and am having
extreme difficulty :) What happens is when the sending * server (the one
initiating the call) receives the ACCEPT back from the receiving server
it immediately replies with INVAL. I've checked the code and it seems to
be not matching the accept packet with the relevant item in the iaxs
array due to the following condition in the match function evaluating as
false:
if ((cur-peercallno == callno) ||
((dcallno == cur-callno) !cur-peercallno))
dcallno and cur-callno are identical and =16384 indicating a trunk
call but cur-peercallno is non-zero hence it fails.
I'm using Asterisk 1.0.7 on both servers. From the code this looks like
a bug but the fact that Google searches show no-one having the same
problem suggests a config problem ;) I'll attach the configs and
ethereal traces of this happening.
Now it gets more interesting :) If I change type=friend to type=peer in
both config files the call goes through fine but upon closer inspection
it seems to trunk in the outgoing direction only. The streams coming
back when two calls are made seem to be in their own packets with low
call numbers (16384).
If anyone would like further information then just ask :) If it turns
out to be a bug then I'd be happy to attempt a fix.
Many thanks,
Peter Spikings
This message has been comprehensively scanned for viruses,
please visit http://virus.e2e-filter.com/ for details.
Frame 1 (110 bytes on wire, 110 bytes captured)
Arrival Time: May 13, 2005 09:54:04.252924000
Time delta from previous packet: 0.0 seconds
Time since reference or first frame: 0.0 seconds
Frame Number: 1
Packet Length: 110 bytes
Capture Length: 110 bytes
Protocols in frame: eth:ip:udp:iax2
Ethernet II, Src: 00:50:da:b2:3f:aa, Dst: 00:90:fb:08:2a:db
Destination: 00:90:fb:08:2a:db (Portwell_08:2a:db)
Source: 00:50:da:b2:3f:aa (3com_b2:3f:aa)
Type: IP (0x0800)
Internet Protocol, Src Addr: 10.10.8.253 (10.10.8.253), Dst Addr: 10.10.8.252
(10.10.8.252)
Version: 4
Header length: 20 bytes
Differentiated Services Field: 0x10 (DSCP 0x04: Unknown DSCP; ECN: 0x00)
0001 00.. = Differentiated Services Codepoint: Unknown (0x04)
..0. = ECN-Capable Transport (ECT): 0
...0 = ECN-CE: 0
Total Length: 96
Identification: 0x0014 (20)
Flags: 0x04 (Don't Fragment)
0... = Reserved bit: Not set
.1.. = Don't fragment: Set
..0. = More fragments: Not set
Fragment offset: 0
Time to live: 64
Protocol: UDP (0x11)
Header checksum: 0x145d (correct)
Source: 10.10.8.253 (10.10.8.253)
Destination: 10.10.8.252 (10.10.8.252)
User Datagram Protocol, Src Port: 4569 (4569), Dst Port: 4569 (4569)
Source port: 4569 (4569)
Destination port: 4569 (4569)
Length: 76
Checksum: 0x7a39 (correct)
Inter-Asterisk eXchange v2
Packet type: Full packet (1)
.100 = Source call: 16384
.000 = Destination call: 0
0... = Retransmission: False
Timestamp: 12
Outbound seq.no.: 0
Inbound seq.no.: 0
Type: IAX (6)
IAX type: NEW (1)
Information Element: Protocol version (0x0B)
IE id: Protocol version (0x0B)
Length: 2
Protocol version: 0x0002
Information Element: Number/extension being called (0x01)
IE id: Number/extension being called (0x01)
Length: 3
Number/extension being called: 601
Information Element: Calling number (0x02)
IE id: Calling number (0x02)
Length: 3
Calling number: 299
Information Element: Name of caller (0x04)
IE id: Name of caller (0x04)
Length: 14
Name of caller: Peter Spikings
Information Element: Desired language (0x0A)
IE id: Desired language (0x0A)
Length: 2
Desired language: en
Information Element: Desired codec format (0x09)
IE id: Desired codec format (0x09)
Length: 4
Desired codec format: GSM compression (0x0002)
Information Element: Actual codec capability (0x08)
IE id: Actual codec capability (0x08)
Length: 4
Actual codec capability: 0xf802
...0 = G.723.1
compression: Not supported
..1. = GSM compression:
Supported
.0.. = Raw mu-law data
(G.711): Not supported
0... = Raw A-law data
(G.711): Not supported
...0
RE: [Asterisk-Users] beginner in Asterisk configuration
Maybe in English you get more answer Ismael. Tutu Lord [EMAIL PROTECTED] Enviado por: [EMAIL PROTECTED] 05/13/2005 11:14 AM Por favor, responda a Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Para asterisk-users@lists.digium.com cc Asunto RE: [Asterisk-Users] beginner in Asterisk configuration bonjour, tout d'abord merci de bien vouloir m'initier à la configuration de serveur Asterisk. J'expose mon soucis, je tourne sous windows XP, donc g pri asterisk version windows == Astwind 0.1.1 Je ne pe pas changer d'ordinateur. Mon objectif : Lancer mon serveur sans qu'il y aille d'erreur, sans aucun client ! j'ai juste le serveur Quoi que je change dans la configuration ( extensions.conf, iax.conf et interfaces pour le reseau) sa me marque apres execution de safe_asterisk -vcf : Asterisk ended with exit status 1 Asterisk died with code 1 Si vous pouviez me fournir un extensions.conf qui tourne je suis preneur :) voila ma extensions.conf actuel : [general] static=yes writeprotect=no [globals] [local] ignorepat = 9 include = iaxtel700 [iaxtel700] exten = 700,1,Echo() merci beacoup lucas _ MSN Messenger : personnalisez votre messagerie instantanée ! http://g.msn.fr/FR1001/866 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: chan_capi, chan_misdn and chan_modem
On Fri, May 13, 2005 at 09:55:58AM +0200, Armin Schindler wrote: On Fri, 13 May 2005, Paul Hales wrote: I battled with chan_capi during the week, and it was not fun. Since I'm working on chan_capi, I would like to know what problems exist. Can you please be more specific on what problems you have encountered? It's good to see a capi expert working on that module! Could you elaborate on your plans for chan_capi? Cheers, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] beginner in Asterisk configuration
I had to use google translate to answer your question, so I'm going to reply with my answer in the same way, the extra time it took me to decipher the question will probably have to be reinvested: Le safe_asterisk est juste un manuscrit pour remettre en marche le serveur si il se termine inopinément. Le safe_asterisk juste de course sans arguments et vous de ligne de commande devrait faire bien. Pour se relier à la console courez: Astérisque - rv -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tutu Lord Sent: 13 May 2005 10:15 To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] beginner in Asterisk configuration bonjour, tout d'abord merci de bien vouloir m'initier à la configuration de serveur Asterisk. J'expose mon soucis, je tourne sous windows XP, donc g pri asterisk version windows == Astwind 0.1.1 Je ne pe pas changer d'ordinateur. Mon objectif : Lancer mon serveur sans qu'il y aille d'erreur, sans aucun client ! j'ai juste le serveur Quoi que je change dans la configuration ( extensions.conf, iax.conf et interfaces pour le reseau) sa me marque apres execution de safe_asterisk -vcf : Asterisk ended with exit status 1 Asterisk died with code 1 Si vous pouviez me fournir un extensions.conf qui tourne je suis preneur :) voila ma extensions.conf actuel : [general] static=yes writeprotect=no [globals] [local] ignorepat = 9 include = iaxtel700 [iaxtel700] exten = 700,1,Echo() merci beacoup lucas _ MSN Messenger : personnalisez votre messagerie instantanée ! http://g.msn.fr/FR1001/866 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 1-800 free calls
On Thu, 12 May 2005, Juanjo Portela wrote: I was using iaxtel to make calls to 1-800 phones for free, but unfortunatelly it is no working ... freenum.org or e164.org ENUMs. SIP/[EMAIL PROTECTED] seems to be the main provider providing ENUM free phone coverage for +1. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SIP and FastStart
I am using Asterisk-oh323 v0.7.1 with GNUGK. Please advise what must be done to make FastStart work with SIP phones. Thanks. On 5/12/05, VoIP Newbie [EMAIL PROTECTED] wrote: Hi all, When I enabled faststart in oh323.conf, calls from H323 endpoint to SIP phones could not complete. The originating phone kept ringing when calls were answered by SIP phones. fastStart=yes h245Tunnelling =yes h245inSetup=yes Please can you advise. Many Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: chan_capi, chan_misdn and chan_modem
On Fri, 13 May 2005, Louis-David Mitterrand wrote: On Fri, May 13, 2005 at 09:55:58AM +0200, Armin Schindler wrote: On Fri, 13 May 2005, Paul Hales wrote: I battled with chan_capi during the week, and it was not fun. Since I'm working on chan_capi, I would like to know what problems exist. Can you please be more specific on what problems you have encountered? It's good to see a capi expert working on that module! Could you elaborate on your plans for chan_capi? Currently I have three topics: - fix the problem with kernel 2.6 (looks like a CAPI problem) - cleanup in chan_capi.c (I noticed some errors) - add native bridging using CAPI Line-Interconnect Then I hope to receive some reports on what is buggy/not working, wishlist and hopefully also some reports on what works well. I also was thinking about an application for receiving fax over CAPI, but I'm not yet familiar with the current asterisk fax support, so I need to learn more here. Maybe some else can inlight me here... Armin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to decrease Asterisk load
Should I believe that at this time there is no DSP capable cards working with Asterisk? - Original Message - i From: izo [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, May 13, 2005 6:11 AM Subject: Re: [Asterisk-Users] How to decrease Asterisk load On 5/12/05, Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote: Digium cards do not have a built in DSP. Neither do the Sangoma as far as I know. I don't know about VoiceTronix. As yet ! As for digium cards latest cvs commits suggest that there is some ongoing development on hardware based echo cancelation and dtmf detection. So its just a matter of time. rgrds m. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: chan_capi, chan_misdn and chan_modem
hello armin, Armin Schindler schrieb: - cleanup in chan_capi.c (I noticed some errors) - add native bridging using CAPI Line-Interconnect this would be very nice I also was thinking about an application for receiving fax over CAPI, but I'm not yet familiar with the current asterisk fax support, so I need to learn more here. Maybe some else can inlight me here... chan_capi currently supports receiving and sending of faxes utilizing the onboard DSPs of the eicon cards. please look for the neccessary patches at: http://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2 regards frank sautter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk newbie
I've just installed Astrisk with AMP. All work well but one thing is not clear. I wanna add users to allow calls between SIP phones. I've added extension but seems not to be enought. How i can add SIP users and allow calls between they ? Thanks ! Oz -- O-Zone ! No (C) 2005 www.zerozone.it ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SpanDSP TXFax and multipage faxes problems
Hi ! Does anyone managed to send multipage faxes (in single TIFF file) with app_txfax from spandsp package (i'm using 0.0.2pre18, libtiff 3.7.1)? If so, I'm interested in format of TIFF file that has been sent sent succesfully (tiffinfo fax-filename). I'm having problems with app_txfax, sending first page successfuly 99 % of the time, but never managed to send second or third page (treid sending to Panasonic fax and to app_rxfax on other Asterisk through the PSTN). Receiving FAX machine hangs up after first page it got (no errors reported) and app_txfax is holding channel for ever (until soft hangup Zap/XX). Here is my tiffinfo on 3 sample files I have tried to fax (on all three only first page being received): LOW-RES fax file (204xx98): lowres:/var/spool/asterisk/fax-out# tiffinfo FAX68.TIF TIFF Directory at offset 0x9160 Image Width: 1728 Image Length: 1173 Resolution: 204, 98 pixels/inch Bits/Sample: 1 Compression Scheme: CCITT Group 3 Photometric Interpretation: min-is-white FillOrder: lsb-to-msb Samples/Pixel: 1 Rows/Strip: 1173 Planar Configuration: single image plane Page Number: 0-3 Group 3 Options: EOL padding (4 = 0x4) TIFF Directory at offset 0x11eb6 Image Width: 1728 Image Length: 1173 Resolution: 204, 98 pixels/inch Bits/Sample: 1 Compression Scheme: CCITT Group 3 Photometric Interpretation: min-is-white FillOrder: lsb-to-msb Samples/Pixel: 1 Rows/Strip: 1173 Planar Configuration: single image plane Page Number: 1-3 Group 3 Options: EOL padding (4 = 0x4) TIFF Directory at offset 0x16e3e Image Width: 1728 Image Length: 1173 Resolution: 204, 98 pixels/inch Bits/Sample: 1 Compression Scheme: CCITT Group 3 Photometric Interpretation: min-is-white FillOrder: lsb-to-msb Samples/Pixel: 1 Rows/Strip: 1173 Planar Configuration: single image plane Page Number: 2-3 Group 3 Options: EOL padding (4 = 0x4) same file HI-RES (204x196) : pbx:/var/spool/asterisk/fax-out# tiffinfo FAX67.TIF TIFF Directory at offset 0x128e8 Subfile Type: multi-page document (2 = 0x2) Image Width: 1728 Image Length: 2347 Resolution: 204, 196 pixels/inch Bits/Sample: 1 Compression Scheme: CCITT Group 3 Photometric Interpretation: min-is-white FillOrder: lsb-to-msb Samples/Pixel: 1 Rows/Strip: 2347 Planar Configuration: single image plane Page Number: 0-3 Group 3 Options: EOL padding (4 = 0x4) TIFF Directory at offset 0x24a6c Subfile Type: multi-page document (2 = 0x2) Image Width: 1728 Image Length: 2347 Resolution: 204, 196 pixels/inch Bits/Sample: 1 Compression Scheme: CCITT Group 3 Photometric Interpretation: min-is-white FillOrder: lsb-to-msb Samples/Pixel: 1 Rows/Strip: 2347 Planar Configuration: single image plane Page Number: 1-3 Group 3 Options: EOL padding (4 = 0x4) TIFF Directory at offset 0x2ed84 Subfile Type: multi-page document (2 = 0x2) Image Width: 1728 Image Length: 2347 Resolution: 204, 196 pixels/inch Bits/Sample: 1 Compression Scheme: CCITT Group 3 Photometric Interpretation: min-is-white FillOrder: lsb-to-msb Samples/Pixel: 1 Rows/Strip: 2347 Planar Configuration: single image plane Page Number: 2-3 Group 3 Options: EOL padding (4 = 0x4) Here is tiffinfo of sample fax TIFF file that I have found looking for the TIFF/FAX specification - this one doesn't go either ! pbx:/var/spool/asterisk/fax-out# tiffinfo SAMPLEF.TIF TIFFReadDirectory: Warning, SAMPLEF.TIF: unknown field with tag 32860 (0x805c) encountered. TIFFReadDirectory: Warning, SAMPLEF.TIF: unknown field with tag 32861 (0x805d) encountered. TIFF Directory at offset 0x62cf Subfile Type: multi-page document (2 = 0x2) Image Width: 1728 Image Length: 852 Resolution: 204, 98 pixels/inch Bits/Sample: 1 Compression Scheme: CCITT Group 3 Photometric Interpretation: min-is-white FillOrder: lsb-to-msb Date Time: 1990:01:05 14:14:38 Document Name: C:\EFAX\EXAMPLE.TXT Image Description: FAX Orientation: row 0 top, col 0 lhs Samples/Pixel: 1 Rows/Strip: 852 Planar Configuration: single image plane Page Number: 0-3 Software: EFax Version 2.2d74 Tag 32860: Tiff Ver 2.0 Tag 32861: 555- Group 3 Options: EOL padding (4 = 0x4) Fax Data: clean (0 = 0x0) Bad Fax Lines: 0 Consecutive Bad Fax Lines: 0 TIFFReadDirectory: Warning, SAMPLEF.TIF: unknown field with tag 32860 (0x805c) encountered. TIFFReadDirectory: Warning, SAMPLEF.TIF: unknown field with tag 32861 (0x805d) encountered. TIFF Directory at offset 0x7fd9 Subfile Type: multi-page document (2 = 0x2) Image Width: 1728 Image Length: 292 Resolution: 204, 98 pixels/inch Bits/Sample: 1 Compression Scheme: CCITT Group 3 Photometric Interpretation: min-is-white FillOrder: lsb-to-msb Date Time: 1990:01:05 14:14:39 Document Name: C:\EFAX\EXAMPLE.TXT Image Description: FAX Orientation: row 0 top, col 0 lhs Samples/Pixel: 1 Rows/Strip: 292 Planar Configuration: single image plane Page Number: 1-3 Software: EFax Version 2.2d74 Tag 32860: Tiff Ver 2.0 Tag 32861: 555- Group 3 Options: EOL padding (4 = 0x4) Fax
[Asterisk-Users] Unchanged sound through Asterisk
Hi! To me, it seems like Asterisk are involved in alternating the sound/voice running through it. One thing is that it mutes DTMF digits. I also got an Adit 600 channel bank connected via MGCP, which _might_ have something to do with it, but I can't find any settings in it, regarding DTMF mutes. How can I make sure Asterisk is _not_ changing/transforming the sound (as muting DTMF etc.) in any way? Thanks! -- Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IPVolution release info....
Thanks for this precision !! Certainly, a good news for Asterisk users community. - Original Message - From: Wiley Siler To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, May 12, 2005 10:16 PM Subject: [Asterisk-Users] IPVolution release info From atacomm From: Jessee J Holmes [mailto:[EMAIL PROTECTED] Sent: Thursday, May 12, 2005 2:24 PMTo: Wiley SilerSubject: Re: Got a date yet? No specific release date as of yet; but, we're hoping to have a physical date soon. So far planned release is either in June or July. Right now they developers are cleaning up the echo cancellation code on the chip andfinalizing things in that aspect. So far everything has been looking good and positive. Jessee Holmes Atacomm / Ataractic Corporation www.atacomm.com V: 1-877-700-VOIP [EMAIL PROTECTED] Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ Atacomm can also provide you with competitive rates from your local carriers. Remember: E-mail is not a secure medium. Please do not send payment information via e-mail. On May 12, 2005, at 1:27 PM, Wiley Siler wrote: ipVolution TDM60 ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ASTCC Compilation Error
Hi, When trying to compile ASTCC i am getting the following error: [EMAIL PROTECTED]:/usr/src/astcc# make install mkdir -p /var/www mkdir -p /var/www/html/_astcc mkdir -p /var/www/cgi-bin/astcc-admin chmod 755 ./astcc.agi chmod 755 ./astcc-admin.cgi echo | ./astcc.agi /dev/null Can't locate Asterisk/AGI.pm in @INC (@INC contains: /usr/lib/perl5/5.8.6/i486-linux /usr/lib/perl5/5.8.6 /usr/lib/perl5/site_perl/5.8.6/i486-linux /usr/lib/perl5/site_perl/5.8.6 /usr/lib/perl5/site_perl .) at ./astcc.agi line 47. BEGIN failed--compilation aborted at ./astcc.agi line 47. make: *** [install] Error 2 Anyone can help please? Thanks, Robson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN Clock Source
I apologise in advance if this is a silly question, as legacy telephone technologies are really not my forte. Is there an E1 card that can provide clock source? (E.g. Make my asterisk server look like a telco to my legacy PBX system?). What I am trying to achieve is the following: --ISDN---| Asterisk |---ISDN| Legacy PBX |-- All E1's and T1's have an embedded clock in the transmit side of the circuit. There is no way to turn it off or on; its part of the spec. You choose whether you want to use the provided clocking when you configure the hardware on the end of the E1/T1. For zaptel cards, see the examples in /etc/zaptel.conf. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sound card Line-In as MOH source
Assuming that I am broadcasting 'legal' content, not having an external live source to play will unsell the concept to many businesses that have already purchased an external MOH source and want to integrate it. Also, sometimes it is legal to broadcast radio (ie, you have paid the fees/etc) and you really don't want to be playing the same music over and over and over and over again Note that in some larger markets, a radio station _may_ have enough interest to allow the moh rebroadcast at either no cost, or may even subsidize your efforts to increase their own audience. Its not always a cost issue. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 2 minutes pause before ring on H323 channel
Peter Valkov wrote: I have build asterisk from latest CVS HEAD-05/09/05 with H323 support as described in README file. Open H.323 version v1.17.1 and PWLib v1.9.0 on Mandrake Linux 10.2 kernel-2.6.11 I tested it with following phones: -- XLite (SIP softphone) -- QMix SIP IP phone (PA168F) -- SJPhone (H323 softphone) -- QMix H323 IP phone (PA168F) -- FireFly (IAX2 softphone) Everything works fine except a problem with h323 extension dialing. Behavior is the same for both SJPhone (soft phone) and QMix (PA168F). When I dial such extension I have to wait 2 minutes exactly (120 seconds) before extension rings. After long way of trial and errors with .conf files I managed to minimize this time to 1 minute exactly (60 seconds) exten = 20,1,Dial(H323/h323phone) ; this leads to 120 seconds pause before ring exten = 21,1,Dial(H323/[EMAIL PROTECTED]) ; this leads to 60 seconds pause before ring Peter, hi; I haven't looked at the openh323 code, and I might not get time to... but in my limited experience, 60 second delays are almost always DNS timeouts. jd -- John Daragon [EMAIL PROTECTED] argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voipjet anyone?
--- Kevin Bockman [EMAIL PROTECTED] wrote:
May 12 22:27:05 VERBOSE[2442]: -- Executing
[1;36;40mDial[0;37;40m([1;35;40mSIP/101-ad89[0;37;40m,
[1;35;40mIAX2/voipjet/4803442640[0;37;40m) in new stack
May 12 22:27:05 VERBOSE[2442]: -- Called voipjet/4803442640
May 12 22:27:05 WARNING[2442]: Call rejected by 66.246.220.19: No such
context/extension
The number needs to be 10 digits. Add a 1 in front.
Tried that, doesnt seem to matter.
Hmm.. did you do an extensions reload? If so, then it should give you a
different error. Show us.
My iax.conf:
[voipjet]
type=peer
;host=216.118.117.46 primary east
host=69.25.60.30
username=username
secret=pass md5
auth=md5
context=default
disallow=all
allow=ilbc
jitterbuffer=yes
dropcount=1
extensions.conf:
exten = _1NXXNXX,1,Dial,IAX2/voipjet/${EXTEN},,tTwW
shouldn't this be:
exten = _1NXXNXX,1,Dial,IAX2/'user'@voipjet/${EXTEN},,tTwW ?
Julius Igugu
SouthWork Co. Ltd.
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Re: [Asterisk-Users] 2 minutes pause before ring on H323 channel
John Daragon wrote: Peter Valkov wrote: I have build asterisk from latest CVS HEAD-05/09/05 with H323 support as described in README file. Open H.323 version v1.17.1 and PWLib v1.9.0 on Mandrake Linux 10.2 kernel-2.6.11 I tested it with following phones: -- XLite (SIP softphone) -- QMix SIP IP phone (PA168F) -- SJPhone (H323 softphone) -- QMix H323 IP phone (PA168F) -- FireFly (IAX2 softphone) Everything works fine except a problem with h323 extension dialing. Behavior is the same for both SJPhone (soft phone) and QMix (PA168F). When I dial such extension I have to wait 2 minutes exactly (120 seconds) before extension rings. After long way of trial and errors with .conf files I managed to minimize this time to 1 minute exactly (60 seconds) exten = 20,1,Dial(H323/h323phone) ; this leads to 120 seconds pause before ring exten = 21,1,Dial(H323/[EMAIL PROTECTED]) ; this leads to 60 seconds pause before ring Peter, hi; I haven't looked at the openh323 code, and I might not get time to... but in my limited experience, 60 second delays are almost always DNS timeouts. Yep - down in openh323/src/transports.cxx there's a method H323TransportAddress::GetIpAndPorts() which is called (eventually) by MakeCallLocked(). This in turn calls GetPortByService() and GetHostByAddress(). My guess is that the 60 second wait is caused by a request to a DNS server that is never honoured. Of course, I've been wrong before... jd -- John Daragon [EMAIL PROTECTED] argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help needed on setting up realtime
I installed Asterisk CVS-NHEAD-05/13/05-01:59:30 and placed few call in and through successfully. I was trying to set up the Realtime - picking the sip.conf and extensions.conf from mysql. I was going through some wiki pages, but what i don't understand is - which configuration change makes asterisk stop looking at extensions.conf and sip.conf for sip peers and pick the same from database. Please suggest. Thank you. Sharath ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voipjet anyone?
I am not an expert yet ;)...but VoipJet is very picky.. in your exten =
when I tried with the default it didn't work, when tried as they ask in the
FAQ's it workedyou must keep the exact format with the account
number...
[Voipjet]
exten = _1NXXNXX,1,SetCallerID(4153574000); Set your CallerID as a ten
;it number like this. See our FAQ
exten = _1NXXNXX,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN} ; VoipJet.com
NANPA
exten = _011.,1,SetCallerID(4153574000); Set your CallerID as a ten digit
number like this. See our FAQ.
exten = _011.,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN} ; VoipJet.com WORLD
;Do not change IAX2/1234 in the above two lines!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of JD
Sent: Friday, May 13, 2005 1:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] voipjet anyone?
Is it me... or is it voipjet?
This week I've been trying various providers, just can't seem to get
voipjet to work.
I signed up with voipjet but so far can't get it to work inbound or out
bound.
I always get 'all circuits busy'.
May 12 22:27:05 VERBOSE[2442]: -- Executing
[1;36;40mDial[0;37;40m([1;35;40mSIP/101-ad89[0;37;40m,
[1;35;40mIAX2/voipjet/4803442640[0;37;40m) in new stack
May 12 22:27:05 VERBOSE[2442]: -- Called voipjet/4803442640
May 12 22:27:05 WARNING[2442]: Call rejected by 66.246.220.19: No such
context/extension
May 12 22:27:05 DEBUG[2442]: Immediately destroying 6, having received
reject
Outbound settings:
notransfer=yes
auth=md5
context=from-pstn
host= 66.246.220.19
secret= md5hashstring
type=friend ; also tried peer and user
username=1234
Im using [EMAIL PROTECTED], but that shouldnt matter; people have this
working or is it me?
JD
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[Asterisk-Users] In/out calls from/to same sip provider
Hi. I'm new to asterisk and, one way or the other, I manage to get it working for me. But I'm having a hard time getting calls going to and coming from the same provider, since the definition of the peer in sip.conf seems to be different AND not compatible for incoming and outgoing call. Outgoing calls need a secret and username definition in the peer context of sip.conf, while incoming ones will have nothing to do with those fields. So I can have incoming or outgoing calls regarding one provider, but not both. I've also tried the sample sintax exten =_42X.,1,Dial(SIP/user:[EMAIL PROTECTED]:[EMAIL PROTECTED],30,rT) that comes with the distribution (debian-sarge), but only to get asterisk unable to create sip channel because host [EMAIL PROTECTED] doesn't exist. The address is that of the provider. voip.org and asteriskdocs.org seems to lead me nowhere. I must be missing something obvious, but can't figure out what it is. Anybody? Thanks. -- Pizco Dominguez -- -- GPGKEY: gpg --keyserver pgp.rediris.es --recv-key 8DE37A4D FINGERPRINT:85CB 4323 F322 5837 EDB5 2033 6FB2 C326 8DE3 7A4D -- -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voipjet anyone?
Personally I thought that VOIPJET has the best service and documentation including simple up to date CDRs also. They do not offer incoming, at least not to me If you follow the instructions on their site it will work, if you are useing AAH then maybe you should look into editing the files by had. Checking the server ping time will help. Everyone with a nonroutable IP address will be surprised which ones are faster. On 5/13/05, JD [EMAIL PROTECTED] wrote: Is it me... or is it voipjet? This week I've been trying various providers, just can't seem to get voipjet to work. I signed up with voipjet but so far can't get it to work inbound or out bound. I always get 'all circuits busy'. May 12 22:27:05 VERBOSE[2442]: -- Executing [1;36;40mDial[0;37;40m([1;35;40mSIP/101-ad89[0;37;40m, [1;35;40mIAX2/voipjet/4803442640[0;37;40m) in new stack May 12 22:27:05 VERBOSE[2442]: -- Called voipjet/4803442640 May 12 22:27:05 WARNING[2442]: Call rejected by 66.246.220.19: No such context/extension May 12 22:27:05 DEBUG[2442]: Immediately destroying 6, having received reject Outbound settings: notransfer=yes auth=md5 context=from-pstn host= 66.246.220.19 secret= md5hashstring type=friend ; also tried peer and user username=1234 Im using [EMAIL PROTECTED], but that shouldnt matter; people have this working or is it me? JD ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- sig Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! /sig ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voipjet anyone?
VoipJet are not too bad, little pricey though.. theres better around.. a matter of looking :-) Regards, Sahil Gupta VoiceValley On Fri, 13 May 2005, Andrew Latham wrote: Personally I thought that VOIPJET has the best service and documentation including simple up to date CDRs also. They do not offer incoming, at least not to me If you follow the instructions on their site it will work, if you are useing AAH then maybe you should look into editing the files by had. Checking the server ping time will help. Everyone with a nonroutable IP address will be surprised which ones are faster. On 5/13/05, JD [EMAIL PROTECTED] wrote: Is it me... or is it voipjet? This week I've been trying various providers, just can't seem to get voipjet to work. I signed up with voipjet but so far can't get it to work inbound or out bound. I always get 'all circuits busy'. May 12 22:27:05 VERBOSE[2442]: -- Executing [1;36;40mDial[0;37;40m([1;35;40mSIP/101-ad89[0;37;40m, [1;35;40mIAX2/voipjet/4803442640[0;37;40m) in new stack May 12 22:27:05 VERBOSE[2442]: -- Called voipjet/4803442640 May 12 22:27:05 WARNING[2442]: Call rejected by 66.246.220.19: No such context/extension May 12 22:27:05 DEBUG[2442]: Immediately destroying 6, having received reject Outbound settings: notransfer=yes auth=md5 context=from-pstn host= 66.246.220.19 secret= md5hashstring type=friend ; also tried peer and user username=1234 Im using [EMAIL PROTECTED], but that shouldnt matter; people have this working or is it me? JD ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- sig Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! /sig ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sound card Line-In as MOH source
Hi, this is a rather ugly solution I devised. Create a script called 'ast-playlinein' (or whatever) in /usr/sbin, as follows: #!/bin/bash /usr/bin/arecord -q -c 1 -r 8000 --buffer-size=2048 -f S16_LE -D hw:0,0 -t raw In musiconhold.conf: [classes] default = custom:/var/lib/asterisk/mohmp3,/usr/sbin/ast-playlinein And that's it. Basically, the script takes input from line-in and sends it to standard output in the format Asterisk expects. You also have to select line-in as recording source using alsamixer utility. If you are using a sound system other than ALSA, odds are that it has some tool similar to arecord. Bear in mind that /var/lib/asterisk/mohmp3 directory still has to contain at least one file with .mp3 extension (can be whatever you like). Hope this helps. -- Niksa ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTCC Compilation Error
You need the asterisk perl module. Check here: http://asterisk.gnuinter.net/ Darren Wiebe [EMAIL PROTECTED] Robson Ribeiro wrote: Hi, When trying to compile ASTCC i am getting the following error: [EMAIL PROTECTED]:/usr/src/astcc# make install mkdir -p /var/www mkdir -p /var/www/html/_astcc mkdir -p /var/www/cgi-bin/astcc-admin chmod 755 ./astcc.agi chmod 755 ./astcc-admin.cgi echo | ./astcc.agi /dev/null Can't locate Asterisk/AGI.pm in @INC (@INC contains: /usr/lib/perl5/5.8.6/i486-linux /usr/lib/perl5/5.8.6 /usr/lib/perl5/site_perl/5.8.6/i486-linux /usr/lib/perl5/site_perl/5.8.6 /usr/lib/perl5/site_perl .) at ./astcc.agi line 47. BEGIN failed--compilation aborted at ./astcc.agi line 47. make: *** [install] Error 2 Anyone can help please? Thanks, Robson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7940G
I need help configuring my Cisco 7940G's for my office. I have [EMAIL PROTECTED] running on the server. Right now all my phone is saying is "Defaulting CM to TFTP Server". I have 5 Cisco 7940G's, a Cisco ATA186, and a Zyxel 2000W Wi-Fi Phone. Right now, my VOIP lines are coming in on the ATA. I don't want to set the ATA up as extensions until this server is running with all extensions and the lines are working properly in asterisk. You can contact me at [EMAIL PROTECTED] or my cell at 517-242-1800. Thanks, Adam Collard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk extensions from Mysql
Hello I was just stuck around as to how I configure my Asterisk to access extensions from Mysql. I have made all the necessary changes in the extconfig.conf, the extensions.conf, res_mysql.conf, res_config_odbc.conf,res_odbc.conf as they have mentioned on the site www.voip-info.org. But still I am getting the error as May 13 07:52:49 ERROR [6420]: res_config_mysql.c:588 mysql_reconnect: MySQL RealTime: Failed to connect database server asterisk on. Check debug for more info. Does anybody know how to go about configuring Asterisk to access extensions from the mysql database? Please kindly do reply Regards, Bharat M. Sarvan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7940G
There is a good section in the handbook on setting up Cisco phones. you well need to get 7.4 SIP firmware from cisco. it sounds like you are running MGCP now. http://asteriskathome.sourceforge.net/handbook/index.html --- Adam Collard [EMAIL PROTECTED] wrote: I need help configuring my Cisco 7940G's for my office. I have [EMAIL PROTECTED] running on the server. Right now all my phone is saying is Defaulting CM to TFTP Server. I have 5 Cisco 7940G's, a Cisco ATA186, and a Zyxel 2000W Wi-Fi Phone. Right now, my VOIP lines are coming in on the ATA. I don't want to set the ATA up as extensions until this server is running with all extensions and the lines are working properly in asterisk. You can contact me at [EMAIL PROTECTED] or my cell at 517-242-1800. Thanks, Adam Collard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Discover Yahoo! Stay in touch with email, IM, photo sharing and more. Check it out! http://discover.yahoo.com/stayintouch.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Autodial and autoanswer
Hello! Is it possible to make the console autoanswer incoming calls to some extensions? Something like this: ; Dial Console with user pickup exten = 123,1,Dial(Console/dsp) ; Dial Console with autoanswer exten = 321,1,Autoanswer(Console/dsp) I want to be able to place calls through the manager-interface to the console, but still want to be able to recieve calls without autoanswering them... Sincerely, Markus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: chan_capi, chan_misdn and chan_modem
Then I hope to receive some reports on what is buggy/not working, wishlist and hopefully also some reports on what works well. There are at least two anoying bugs: 1. The Busy-Applicatzion does not work, there seems to be no was to singnal Busy to the caller is no SIP-Phone is ready to answer the call. 2. Dial-Application does not really detect the reason for Failings. As an Example you should have a look at the LCR script available at Telefonsparbuch.de: The script trys to do some Fallback but it does not work with chan_capi. Elmar ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voipjet anyone?
I agree. I've been using voipjet since before their formal launch..I have account number 63. They've been amongst the most reliable in my experience. If they offered DIDs in my area I'd have those as well. Michael On Fri, 13 May 2005 07:42:32 -0500, Andrew Latham wrote: Personally I thought that VOIPJET has the best service and documentation including simple up to date CDRs also. They do not offer incoming, at least not to me If you follow the instructions on their site it will work, if you are useing AAH then maybe you should look into editing the files by had. Checking the server ping time will help. Everyone with a nonroutable IP address will be surprised which ones are faster. On 5/13/05, JD [EMAIL PROTECTED] wrote: Is it me... or is it voipjet? This week I've been trying various providers, just can't seem to get voipjet to work. I signed up with voipjet but so far can't get it to work inbound or out bound. I always get 'all circuits busy'. May 12 22:27:05 VERBOSE[2442]: -- Executing [1;36;40mDial[0;37;40m([1;35;40mSIP/101-ad89[0;37;40m, [1;35;40mIAX2/voipjet/4803442640[0;37;40m) in new stack May 12 22:27:05 VERBOSE[2442]: -- Called voipjet/4803442640 May 12 22:27:05 WARNING[2442]: Call rejected by 66.246.220.19: No such context/extension May 12 22:27:05 DEBUG[2442]: Immediately destroying 6, having received reject Outbound settings: notransfer=yes auth=md5 context=from-pstn host= 66.246.220.19 secret= md5hashstring type=friend ; also tried peer and user username=1234 Im using [EMAIL PROTECTED], but that shouldnt matter; people have this working or is it me? JD ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- sig Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! /sig ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Extension never ring, goes straight to VM
I installed a new * server and copied the sip.conf and extensions.conf from my existing setup to the new box. I created my outbound trunk with a different broadvoice account and am able to dial out without issue. I am able to dial all extensions but I go straight to voicemail without any ringing, this happens to both local and outside numbers calling my extensions. I have the correct firewall ports open and even went so far as to put the * box in the dmz for my router. All of these clients are behind the same firewall (no firewall in between them). One of the clients (xlite softphone) is on my laptop, the laptop can ping all of the phones and the pbx. Looking in the logs I get the following lines: ,9419263581,7001,ext-local,VDC Support 9419263581,SIP/9418660063-183c,,VoiceMail,[EMAIL PROTECTED],2005-05-11 16:45:48,2005-05-11 16:45:48,2005-05-11 16:46:03,15,15,ANSWERED,DOCUMENTATION ,7009,7001,from-internal,Tims Laptop 7009,SIP/7009-5067,,ResetCDR,w,2005-05-11 16:51:51,2005-05-11 16:51:52,2005-05-11 16:52:03,12,11,ANSWERED,DOCUMENTATION The one going to an external line (first entry) really was answered as I picked up the phone The second line however was not answered by me as it went straight to voicemail. These phones and conf files work fine on my original system but when moved to new system (changed ip for sip stuff is really all that needed changing) they don't work the same. Here is my sip.conf: http://www.vulgardisplayofpower.com/files/sip.conf Here is my extensions.conf http://www.vulgardisplayofpower.com/files/extensions.conf anyone see where I went wrong? I don't see anything in the logfiles to indicate a problem even running the debug flag I'm new to * and am not yet a master, can some of you experience users see anything wrong with my config? thanks a bunch! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Current status of voicemail monitoring?
Hi all. I'm curious as to the current status and development of a way to monitor incoming voicemail in Asterisk. IE: The screen calls with the answering machine feature -- the ability to listen to and break into a currently-recording voicemail if you want to. This feature would be very helpful for our application. I've seen various things in the archives about this but none where someone has said, Yeah, we have this, it works. Does anyone have a current implementation of this and/or a reasonable alternative, and how did you do it? Thanks much for the help. Nathan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk with ShoreTel 210 (MGCP)
can you post your mgcp.conf file. From the debug output it looks like * can not find the gateway in the mgcp.conf (* goes on to tell you it can not match the endpoint, because it first has to find the gateway device...) - Original Message - From: Ben Dugdale [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, May 12, 2005 6:51 PM Subject: Re: [Asterisk-Users] Asterisk with ShoreTel 210 (MGCP) Duane Cox wrote: Yes * can work with MGCP phones directly. You have a configuration issue. Glad to hear it! a typical mgcp.conf might be: [general] port= 2427 bindaddr= 0.0.0.0 [10.21.4.2] accountcode = 1123 amaflags= billing context = main host= 10.21.4.2 wcardep = aaln/* callerid= YOUR NAME 1231231234 callwaiting = no callreturn = yes cancallforward = yes canreinvite = no threewaycalling = no transfer= no dtmfmode= none line = aaln/1 Where does a person find a list of the mgcp.conf options and meanings? ( I've tried 'man mgcp' 'man mgcp.conf' and looked for info in the doc directory of the * source (I did make documentation at install) )? turn on MGCP debug mgcp debug and see what messages are going to and fro. I'm now using Asterisk CVS-HEAD-05/12/05-16:10:03 Here is what I see at the console: MGCP Debugging Enabled *CLI MGCP read: RSIP 11630 [EMAIL PROTECTED] MGCP 1.0 RM: restart X-ShoreModel: S1 from 192.168.90.209:2427 Verb: 'RSIP', Identifier: '11630', Endpoint: '[EMAIL PROTECTED]', Version: 'MGCP 1.0' 3 headers, 0 lines May 12 16:31:56 NOTICE[28300]: chan_mgcp.c:1644 find_subchannel_and_lock: Gateway '192.168.90.209' (and thus its endpoint 'SHOR_001049007E83') does not exist MGCP read: RSIP 11630 [EMAIL PROTECTED] MGCP 1.0 RM: restart X-ShoreModel: S1 Here is what I see with ngrep port 2727 interface: eth0 (192.168.90.0/255.255.255.0) filter: ip and ( port 2727 ) # U 192.168.90.209:2427 - 192.168.90.6:2727 RSIP 11625 [EMAIL PROTECTED] MGCP 1.0.RM: restart.X-ShoreModel: S1. I've changed mgcp.conf to pretty much exactly look like the example: [192.168.90.209] accountcode = 1000 ; record this in cdr as account identification for billing amaflags= billing ; record this in cdr as flagged for 'billing', 'documentation', or 'omit' context = local host= 192.168.90.209 wcardep = aaln/*; enables wildcard endpoint and sets it to 'aaln/*' another common format is '*' callerid= Duane Cox 123 ; now lets setup line 1 using per endpoint configuration... callwaiting = no callreturn = yes cancallforward = yes canreinvite = no transfer= no dtmfmode= inband line = aaln/1 ; now lets save this config to line1 aka aaln/1 I would suspect that your end device is setup looking for host/domain name convention and not IP. If so, change * to match or change your device to IP and not dns. I'm not sure I know what you mean by this. I tried setting it up in DNS to see if the name it reported changed to a FQDN but it stayed the same. I wonder if the phone is trying to register as '[EMAIL PROTECTED]' but * interperets that as it being behind another gateway because of a special character in the string (@)? What does the normal MGCP negotiation look like? GoodLuck, Duane Cox Thanks, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi, chan_misdn and chan_modem
Hi, time to clear some things up. :) The new version of chan_capi (0.4.0) is still work in progress (no, I have not dropped chan_capi in favour of BRIstuff). I harmonized the dialstring syntax with chan_zap, so you can just use CAPI/g1/... instead of those strange constructions with the outgoing msn. It also contains fixes (contributed by Jan Stocke) to make it work on BSD. Also it will then work properly with p2p BRIs in Austria. Chan_capi 0.4.0 will work with Asterisk stable and cvs head. It does not distinguish between certain card types (CAPI means Common isdn API), maybe you (or the Wiki) are mistaking this with BRIstuff. In the beginning BRIstuff was only intended as a driver package for our BRI hardware. But more and more enhancements (to asterisk and libpri) were added and i merged and maintain patches and applications from other people that were contributed under the GPL (and thus could not be integrated into the asterisk cvs tree). We provide a version for Asterisk stable and for cvs head. If you compare chan_capi, bristuff and chan_misdn then chan_capi and chan_misdn would fall into the same category as they are just channel drivers which do not touch the asterisk core at all. BRIstuff changes some things in Asterisk to better support European users and contains modificatiosn that we made for clients. Regarding stability chan_capi and BRIstuff (for Asterisk stable) will fall into the same category. They are used in hundreds of production installations around the globe. I cannot comment on the stability of chan_misdn because i have never used it (i have read the source though), but i made the experience the authors of chan_misdn (Beronet) supply patches for bristuff to their customers that enable them to use BRIstuff with their hardware (instead of chan_misdn). They do not distinguish between cards (so also our Junghanns.NET cards work with chan_misdn) because chan_misdn does not talk directly to the card. This is done by the mISDN kernel modules. The driver for the HFC-4S/8S based cards (used in the Junghanns.NET amd Beronet cards) was not written by Beronet but by the author of PBX4Linux, Andreas Eversberg. So, for your hfc-pci based isdn card you can use the zaphfc module from BRIstuff and use it with chan_zap OR you can use it with the mISDN driver and chan_misdn OR you can use it with the mISDN driver plus the capi layer of mISDN and chan_capi. I also have a W6692 card laying on my desk (contributed by Michael Sandee) and will write a zaptel driver for that card, but this is rather a longterm project. ;) best regards Klaus -- Klaus-Peter Junghanns Am Freitag, den 13.05.2005, 08:46 +0200 schrieb Jan Louw: Could someone please comment on the current state of chan_capi, chan_misdn and chan_modem channel drivers in terms of functionality (echo cancelation, fax, latency etc) and stability. Specifically, which channel driver would be best for a passive PCI HFC or W6692 ISDN card. The chan_misdn wiki claims that chan_capi distinguishes between junghanns and non-junghans cards, and that chan_misdn is better suited for general misdn compatibility. A second point I'd like some clarification on is the purpose of Junghann's BRIStuff patch. Is this patch only necessary for chan_capi or also for chan_misdn? Does this patch add functionality to asterisk or is it only intended to smooth chan_capi integration into asterisk? Thanks in advance! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voipjet anyone?
I'm #11 but I have notice of late a few problems but nothing major given the price differences assuming you don't have the volume to commit to another carrier directly for the destinations you are after. -- William ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iax trunking not works!
hello, iax trunking not working we actually testing dial 500(Digium) two or three calls simultaneously but bandwidth graph shows 95 to 100kbps not match the results shows on wiki iax bandwidth pages i enable trunk=yes in iax.conf is there any tweaking or optimization because i desperately need some solution for this Thanks In Advance. Adnan Ahmed. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voipjet anyone?
I use AAH with VoipJet and it works perfectly. Setup was a breeze with absolutely no hand coding of configs required. VoipJet is without a doubt the best outbound provider I have come across. No problems at all yet. knock on wood And the call quality has been awesome. Anyone having trouble getting their AAH setup with voipjet may contact me off list for assistance if they want. Cheers, W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Latham Sent: Friday, May 13, 2005 5:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] voipjet anyone? Personally I thought that VOIPJET has the best service and documentation including simple up to date CDRs also. They do not offer incoming, at least not to me If you follow the instructions on their site it will work, if you are useing AAH then maybe you should look into editing the files by had. Checking the server ping time will help. Everyone with a nonroutable IP address will be surprised which ones are faster. On 5/13/05, JD [EMAIL PROTECTED] wrote: Is it me... or is it voipjet? This week I've been trying various providers, just can't seem to get voipjet to work. I signed up with voipjet but so far can't get it to work inbound or out bound. I always get 'all circuits busy'. May 12 22:27:05 VERBOSE[2442]: -- Executing [1;36;40mDial[0;37;40m([1;35;40mSIP/101-ad89[0;37;40m, [1;35;40mIAX2/voipjet/4803442640[0;37;40m) in new stack May 12 22:27:05 VERBOSE[2442]: -- Called voipjet/4803442640 May 12 22:27:05 WARNING[2442]: Call rejected by 66.246.220.19: No such context/extension May 12 22:27:05 DEBUG[2442]: Immediately destroying 6, having received reject Outbound settings: notransfer=yes auth=md5 context=from-pstn host= 66.246.220.19 secret= md5hashstring type=friend ; also tried peer and user username=1234 Im using [EMAIL PROTECTED], but that shouldnt matter; people have this working or is it me? JD ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- sig Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! /sig ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] iax trunking not works!
What codec are you using? -Original Message- From: Adnan Ahmed [mailto:[EMAIL PROTECTED] Sent: Friday, May 13, 2005 9:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] iax trunking not works! hello, iax trunking not working we actually testing dial 500(Digium) two or three calls simultaneously but bandwidth graph shows 95 to 100kbps not match the results shows on wiki iax bandwidth pages i enable trunk=yes in iax.conf is there any tweaking or optimization because i desperately need some solution for this Thanks In Advance. Adnan Ahmed. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voipjet anyone?
1.3 cents minute dialing? That is one of the lowest prices out there. Maybe for you in Australia but in North America, it is a very nice deal. W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sahil Gupta Sent: Friday, May 13, 2005 5:49 AM To: Andrew Latham; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] voipjet anyone? VoipJet are not too bad, little pricey though.. theres better around.. a matter of looking :-) Regards, Sahil Gupta VoiceValley On Fri, 13 May 2005, Andrew Latham wrote: Personally I thought that VOIPJET has the best service and documentation including simple up to date CDRs also. They do not offer incoming, at least not to me If you follow the instructions on their site it will work, if you are useing AAH then maybe you should look into editing the files by had. Checking the server ping time will help. Everyone with a nonroutable IP address will be surprised which ones are faster. On 5/13/05, JD [EMAIL PROTECTED] wrote: Is it me... or is it voipjet? This week I've been trying various providers, just can't seem to get voipjet to work. I signed up with voipjet but so far can't get it to work inbound or out bound. I always get 'all circuits busy'. May 12 22:27:05 VERBOSE[2442]: -- Executing [1;36;40mDial[0;37;40m([1;35;40mSIP/101-ad89[0;37;40m, [1;35;40mIAX2/voipjet/4803442640[0;37;40m) in new stack May 12 22:27:05 VERBOSE[2442]: -- Called voipjet/4803442640 May 12 22:27:05 WARNING[2442]: Call rejected by 66.246.220.19: No such context/extension May 12 22:27:05 DEBUG[2442]: Immediately destroying 6, having received reject Outbound settings: notransfer=yes auth=md5 context=from-pstn host= 66.246.220.19 secret= md5hashstring type=friend ; also tried peer and user username=1234 Im using [EMAIL PROTECTED], but that shouldnt matter; people have this working or is it me? JD ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- sig Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! /sig ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to decrease Asterisk load
Mamadou Lamine KA wrote: Should I believe that at this time there is no DSP capable cards working with Asterisk? That is correct as far as I know. The entire DESIGN of Asterisk is to do the DSP work in software. Rumor has it that Digium is coming out with a DSP version of their cards (or a DSP addon board), but it will be expensive, like all DSP based boards are. Since they are not actually shipping it yet, this is all speculation. - Original Message - i From: izo [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, May 13, 2005 6:11 AM Subject: Re: [Asterisk-Users] How to decrease Asterisk load On 5/12/05, Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote: Digium cards do not have a built in DSP. Neither do the Sangoma as far as I know. I don't know about VoiceTronix. As yet ! As for digium cards latest cvs commits suggest that there is some ongoing development on hardware based echo cancelation and dtmf detection. So its just a matter of time. rgrds m. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to decrease Asterisk load
It is not that they are not working with Asterisk... It is that there are none available. Go check out the link that was sent to you before... Here it is again http://www.zapatatelephony.org/ As you can see, Zapata (which drives Asterisk) was originally designed to be a chipless DSP solution. All the functions of a DSP without the need for specialized equipment. Plain old x86 doing the functions of DSP. So that is why there are no cards with DSP for Asterisk. It was designed to not need them. The fact that new cards are getting DSPs now is just due to the fact that * servers are being loaded so much now that offloading echo cancellation and such to a DSP should allow for higher capacity systems across the board. Of ccourse, being vaporware, we have not even seen if it will be that much of a benefit. Cheers, W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mamadou Lamine KA Sent: Friday, May 13, 2005 3:40 AM To: izo; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to decrease Asterisk load Should I believe that at this time there is no DSP capable cards working with Asterisk? - Original Message - i From: izo [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, May 13, 2005 6:11 AM Subject: Re: [Asterisk-Users] How to decrease Asterisk load On 5/12/05, Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote: Digium cards do not have a built in DSP. Neither do the Sangoma as far as I know. I don't know about VoiceTronix. As yet ! As for digium cards latest cvs commits suggest that there is some ongoing development on hardware based echo cancelation and dtmf detection. So its just a matter of time. rgrds m. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voipjet anyone?
Almost positive iLBC is not allowed Use uLaw...
This is directly form the install instructions...
Step 3A (recommended): Set your codec to G.711 ulaw for optimal sound
clarity and minimal transmission delay. In iax.conf (found in
/etc/asterisk) locate the codec section and include the following only.
disallow=all ; Prevent all codecs...
allow = ulaw ; ...except G.711 ulaw
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin
Bockman
Sent: Thursday, May 12, 2005 10:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] voipjet anyone?
May 12 22:27:05 VERBOSE[2442]: -- Executing
[1;36;40mDial[0;37;40m([1;35;40mSIP/101-ad89[0;37;40m,
[1;35;40mIAX2/voipjet/4803442640[0;37;40m) in new stack May 12
22:27:05 VERBOSE[2442]: -- Called voipjet/4803442640 May 12 22:27:05
WARNING[2442]: Call rejected by 66.246.220.19: No such
context/extension
The number needs to be 10 digits. Add a 1 in front.
Tried that, doesnt seem to matter.
Hmm.. did you do an extensions reload? If so, then it should give you a
different error. Show us.
My iax.conf:
[voipjet]
type=peer
;host=216.118.117.46 primary east
host=69.25.60.30
username=username
secret=pass md5
auth=md5
context=default
disallow=all
allow=ilbc
jitterbuffer=yes
dropcount=1
extensions.conf:
exten = _1NXXNXX,1,Dial,IAX2/voipjet/${EXTEN},,tTwW
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RE: [Asterisk-Users] French SIP or IAX phones
Videotel !!! : French software, Video hard phone, Excellent browser... see it at : http://www.call.fr Works fine with Asterisk. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] la part de Martin Roy Envoye : vendredi 13 mai 2005 00:52 A : asterisk-users@lists.digium.com Objet : [Asterisk-Users] French SIP or IAX phones Is there any SIP or IAX phones that can be configure in french instead of english. I tested Cisco 7960 phones but I can't change the language it's only available in english with the SIP firmware. I have a customer that's located in France and he wants french phones if possible. So I'm wondering if there's any one out there that found a phone that can be change to french. Thanks Martin Roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel and zttest
I am having trouble with zttest on a Tyan board, dual AMD Opteron's on
FC3. Tried different kernels, no IRQ sharing, everything looks in
order. My zaptel modules load fine, but if I run zttest, it just
hangs. Below is the strace output and you can see where it stops.
Anyone have any ideas?
[EMAIL PROTECTED] zaptel]# strace ./zttest
execve(./zttest, [./zttest], [/* 24 vars */]) = 0
uname({sys=Linux, node=asterisktest.astroshapes.com, ...}) = 0
brk(0) = 0x502000
mmap(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0)
= 0x2aac
access(/etc/ld.so.preload, R_OK) = -1 ENOENT (No such file or
directory)
open(/etc/ld.so.cache, O_RDONLY) = 3
fstat(3, {st_mode=S_IFREG|0644, st_size=150975, ...}) = 0
mmap(NULL, 150975, PROT_READ, MAP_PRIVATE, 3, 0) = 0x2aac1000
close(3)= 0
open(/lib64/tls/libc.so.6, O_RDONLY) = 3# ./zttest
Opened pseudo zap interface, measuring accuracy...
--- Results after 0 passes ---
Best: 0.00 -- Worst: 100.00 -- Average: 100.00
read(3, \177ELF\2\1\1\0\0\0\0\0\0\0\0\0\3\0\0\1\0\0\0p\305\1\0...,
640) = 640
lseek(3, 624, SEEK_SET) = 624
read(3, \4\0\0\0\20\0\0\0\1\0\0\0GNU\0\0\0\0\0\2\0\0\0\4\0\0\0..., 32)
= 32
fstat(3, {st_mode=S_IFREG|0755, st_size=1605832, ...}) = 0
mmap(NULL, 2297832, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_DENYWRITE, 3,
0) = 0x2abc1000
mprotect(0x2ace9000, 1085416, PROT_NONE) = 0
mmap(0x2adc1000, 184320, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x10) = 0x2adc1000
mmap(0x2adee000, 16360, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0x2adee000
close(3)= 0
mmap(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0)
= 0x2adf2000
mmap(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0)
= 0x2adf3000
mprotect(0x2ade8000, 16384, PROT_READ) = 0
mprotect(0x2abbf000, 4096, PROT_READ) = 0
arch_prctl(0x1002, 0x2adf2b00) = 0
munmap(0x2aac1000, 150975) = 0
open(/dev/zap/pseudo, O_RDWR) = 3
fstat(1, {st_mode=S_IFCHR|0620, st_rdev=makedev(136, 1), ...}) = 0
mmap(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0)
= 0x2aac1000
write(1, Opened pseudo zap interface, mea..., 51Opened pseudo zap
interface, measuring accuracy...
) = 51
rt_sigaction(SIGHUP, {0x400910, [HUP], SA_RESTART|0x400}, {SIG_DFL},
8) = 0
rt_sigaction(SIGINT, {0x400910, [INT], SA_RESTART|0x400}, {SIG_DFL},
8) = 0
read(3,
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[Asterisk-Users] ISDN passive card (HiSAX driver) / Fax reciever
Hello all. I have an ISDN passive card (HC HFC 2BDS0) using HiSAX driver (since this is a passive card, I can't use the CAPI driver... ). All I want is recieve faxes and store them on diferent folders depending the destination number. So far I was able to do a similar thing with voice (vbox), but what I really pretend is recieving faxes. My system is a SuSE 9.2 and has instaled asterisk a asterisk-spandsp. So my question is simple... is it possible? Has someone do anything similar? Where can I find documentation? Best regards, -vcf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Why always getting max retries error during idle?
My home asterisk seems to work- I can call from one internal phone to another. However, just leaving my system idle always generates an error message relating to a NOTIFY. See the log below. Any ideas? Thanks, Mike --MESSAGE FILE- to 172.31.254.106:5065May 13 11:01:28 VERBOSE[716]: Retransmitting #5 (no NAT):NOTIFY sip:[EMAIL PROTECTED]:5065 SIP/2.0Via: SIP/2.0/UDP 172.31.254.2:5060;branch=z9hG4bK6e7b6440From: "asterisk" sip:[EMAIL PROTECTED];tag=as116e0fc1To: sip:[EMAIL PROTECTED]:5065Contact: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 102 NOTIFYUser-Agent: Asterisk PBXEvent: message-summaryContent-Type: application/simple-message-summaryContent-Length: 42 Messages-Waiting: noVoice-Message: 0/0 to 172.31.254.106:5065May 13 11:01:29 WARNING[716]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request)May 13 11:01:32 VERBOSE[716]: ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Tyan Transport GX28 with TDM400
I want to know if I buy a Tyan Transport GX28 (B2881) will it work with a TDM400 card? As the expansion slots are only (2) 64-bit 133/100MHz PCI-X. I never tried PCI 2.2 compliant card in a PCI-X slot so I don't know if it can even fit in the slot and if it does will it be seen? Thanks Martin Roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dropped Calls between Sip and Zaptel
Hi, I am having trouble with dropped calls in Asterisk. I've done a bunch of searching but all I could find was setting busydetect and callprogress to yes in zapata.conf to help combat the problem, but I did this to no avail. The following is the output from /var/log/asterisk/full at the time the call was dropped on me. May 13 08:37:13 DEBUG[5379]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found May 13 08:37:16 DEBUG[8480]: Didn't get a frame from channel: SIP/cronus-116-78ed May 13 08:37:16 DEBUG[8480]: Bridge stops bridging channels Zap/1-1 and SIP/cronus-116-78ed May 13 08:37:16 DEBUG[8480]: update_user_counter(cronus-116) - decrement outUse counter May 13 08:37:16 DEBUG[8480]: Exiting with DIALSTATUS=ANSWER. May 13 08:37:16 VERBOSE[8480]: == Spawn extension (macro-netvoice-stdexten, s, 302) exited non-zero on 'Zap/1-1' in macro 'netvoice-stdexten May 13 08:37:16 VERBOSE[8480]: == Spawn extension (main-menu, 116, 1) exited non-zero on 'Zap/1-1' May 13 08:37:16 DEBUG[8480]: Hangup: channel: 1 index = 0, normal = 21, callwait = -1, thirdcall = -1 May 13 08:37:16 DEBUG[8480]: disabled echo cancellation on channel 1 May 13 08:37:16 DEBUG[8480]: Set option TDD MODE, value: OFF(0) on Zap/1-1 May 13 08:37:16 DEBUG[8480]: Updated conferencing on 1, with 0 conference users May 13 08:37:16 VERBOSE[8480]: -- Hungup 'Zap/1-1' May 13 08:37:17 DEBUG[5379]: Auto destroying call '[EMAIL PROTECTED]' Does anyone know any other solutions to this problem? Faulty hardware perhaps? By the way I have a TDM400P with 4 FXO ports on it and have asterisk running on Debian 3.0. Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voipjet anyone?
Look a little closer...
WARNING[2442]: Call rejected by 66.246.220.19: No such context/extension
that certainly does not imply an incorrect codec!
Almost positive iLBC is not allowed Use uLaw...
This is directly form the install instructions...
Step 3A (recommended): Set your codec to G.711 ulaw for optimal sound
clarity and minimal transmission delay. In iax.conf (found in
/etc/asterisk) locate the codec section and include the following only.
disallow=all ; Prevent all codecs...
allow = ulaw ; ...except G.711 ulaw
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin
Bockman
Sent: Thursday, May 12, 2005 10:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] voipjet anyone?
May 12 22:27:05 VERBOSE[2442]: -- Executing
[1;36;40mDial[0;37;40m([1;35;40mSIP/101-ad89[0;37;40m,
[1;35;40mIAX2/voipjet/4803442640[0;37;40m) in new stack May 12
22:27:05 VERBOSE[2442]: -- Called voipjet/4803442640 May 12 22:27:05
WARNING[2442]: Call rejected by 66.246.220.19: No such
context/extension
The number needs to be 10 digits. Add a 1 in front.
Tried that, doesnt seem to matter.
Hmm.. did you do an extensions reload? If so, then it should give you a
different error. Show us.
My iax.conf:
[voipjet]
type=peer
;host=216.118.117.46 primary east
host=69.25.60.30
username=username
secret=pass md5
auth=md5
context=default
disallow=all
allow=ilbc
jitterbuffer=yes
dropcount=1
extensions.conf:
exten = _1NXXNXX,1,Dial,IAX2/voipjet/${EXTEN},,tTwW
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---End of Original Message-
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[Asterisk-Users] My experience with our VS-1 Asterisk server
I own and operate a number of franchised Sylvan Learning Centers where I recently upgraded to an all VOIP phone system (Asterisk) with one VS-1 and about 25 extensions scattered around the country. I had originally setup a Dell 420 SC but the Dell had incurable buss issues with single span and quad span T1 cards. I wasted a LOT of time trying to get the Dell to work and finally gave up on it and bought a VS-1 from thevoipconnection.com which has worked great right out of the box for about a month now . We connect to our T1 using Digium's T110TE which works great. I would be happy to answer questions by email or phone about my experience with our VS-1. Anthony Gagliardo 501-765-0229 Message: 21 Date: Thu, 12 May 2005 22:37:08 -0400 From: The VoIP Connection [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] * Server To: 'snacktime' [EMAIL PROTECTED], 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Chris, Obviously we can't publish a list of our customers on this or any other news group, but if you would like some references we would be happy to provide them. I know some of them are on the list, maybe they will be kind enough to share their opinions. The VS-1 has been performing flawlessly in production at numerous locations for over a year now, and it should continue to do so for many more. The reason we don't specify an Asterisk version on our web site and data sheet is that the marketing hype for the VS-1 pre-dates Version 1.0. The latest VS-1 comes with two versions of Asterisk installed: One is a stable version (currently 1.0.7), the other is a development version which is built from CVS head and the occasional assorted patches. The management interface allows the administrator to easily select which of the two versions of Asterisk (stable or development)they wish to run. Either version can be updated or modified by the administrator should they choose to. Since the door has been opened, I'll offer up a little more hype: Unlike some other turn-key products, the VS-1 is not an attempt to dumb down or obfuscate Asterisk. It is not impaired or restricted in any way. Our web management interface is a layer on top of the standard configuration file interface and it does not interfere with or overwrite direct edits to the files. The system comes with most of the popular functionality pre-configured: voicemail, festival, moh, meetme, FOP, etc. and also includes a number of other tools and utilities that ease setup, phone provisioning and remote administration. Like a lot of good technology, the VS-1 is 1% inspiration and 99% perspiration. A significant amount of engineering has gone into this unassuming little black box. It is designed for stability and reliability as opposed to ultra high performance, but with limited transcoding it can easily handle a full T1 of PRI. It's also pre-configured for several popular VoIP service providers and it gets along just fine with a TDM400P card. We stand behind it with a one year replacement warrantee and 30 day money back guarantee. Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: snacktime [mailto:[EMAIL PROTECTED] Sent: Thursday, May 12, 2005 4:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] * Server On 5/12/05, Montague, Clarence [EMAIL PROTECTED] wrote: Any reviews/comments out there on this server? Looks solid.. But would like to know if anyone has purchased one of these before. Any other companies out there offer pre-built * servers that someone would like to comment on? http://www.thevoipconnection.com/store/catalog/product_16214_VS1trade_ Asterisk_Voice_Server.html Personally I would want to see the full specifications and get some more information about it's track record in production use. If this unit was vouched for by some recognized names that have used it in production, and if they stated what version of asterisk was used instead of just saying it's their own 'certified' version, I might be inclined to say it looks like a good deal for a small business or office environment. As is it seems short on details and long on marketing hype. Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: chan_capi, chan_misdn and chan_modem
On Fri, 13 May 2005, Frank Sautter wrote: hello armin, I also was thinking about an application for receiving fax over CAPI, but I'm not yet familiar with the current asterisk fax support, so I need to learn more here. Maybe some else can inlight me here... chan_capi currently supports receiving and sending of faxes utilizing the onboard DSPs of the eicon cards. please look for the neccessary patches at: http://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2 That is exactly what I was thinking about. I did not have a close look into the patch yet, but this archive seems to be incomplete. Only changed files are part of the patch, but the real app_capiFax.c is missing. The patch was obviously created without the '-N' option of diff. Can you please check that? Anyway, I think this should be part of the chan_capi package. I will wait for klaus-peter's release of 0.4.0, before starting with new things. So the CAPI on kernel 2.6 problem is on top now... Armin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voipjet anyone?
On Fri, 13 May 2005 07:59:09 -0700 Wiley Siler [EMAIL PROTECTED] wrote: Almost positive iLBC is not allowed Use uLaw... They do allow for iLBC. From their FAQ page: Codecs. Carriers with primarily business customers should use the G.711 codec when sending VoIP traffic to VoipJet. This ensures that all calls are of the highest sound quality and free from compression degradation. You've paid for VoipJet Tier 1 telco termination and G.711 lets you fully enjoy it. And don't forget some other codecs can add significantly to a call's end-to-end latency, too. If you need to save bandwidth (admittedly very expensive in some parts of the world) then the iLBC codec bundled with Asterisk makes an excellent choice. It's free to use and takes one-fouth the bandwidth of G.711. Another important reason to try iLBC is if there is jitter and packet loss on your network's connection to us, because G.711 really needs ideal conditions to work well. Finally, GSM potentially uses even less bandwidth and CPU processing power than iLBC. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] About Voip Technology : RTP over TCP
I am interested in implementing RTP over TCP Why? If you want to permit operation through a firewall that blocks UDP, there are packages that provide VPN tunnels over TCP or even HTTP. You could then run any VoIP system over that VPN. As you said, delay performance would sometimes be awful. Skype will automatically fall back to TCP if a UDP connection attempt fails. Most of the commercial instant messaging packages that support voice or video can work over TCP. If your purpose is to improve performance on networks with high packet loss rates, IMHO you would get better results from a UDP-based system that permits forward error correction, by transmitting each voice frame in two or more packets. If you can't afford the increased bandwidth, a system of retransmission such as used by popular streaming protocols would still be better than TCP. One more point is What is feasibility of implementing RTP over TCP in case of NAT (Network Address Translation) is there ? Any of the above systems can work through NAT. If both endpoints are behind NATs, and you can't set up port forwarding on either, then of course you must connect via an intermediate server. Skype and the IM services do that automatically. If your desire for TCP is not related to firewalls or packet loss, I'd be interested in hearing about your application. --Stewart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: SpanDSP TXFax and multipage faxes problems
I'd really like more info on how to correctly format a tiff for tx_fax too. The only Tiff's we've been able to send using tx_fax with consistant success are the ones that rx_fax creates when it receives an incoming fax. Here's the Tiff info from one such if its any help. TIFF Directory at offset 0x17b9a Image Width: 1728 Image Length: 2150 Resolution: 77, 77 pixels/cm Bits/Sample: 1 Compression Scheme: CCITT Group 3 Photometric Interpretation: min-is-white FillOrder: lsb-to-msb Date Time: 2005/05/04 13:07:54 Host Computer: asterisk1.local Software: spandsp Orientation: row 0 top, col 0 lhs Samples/Pixel: 1 Rows/Strip: (infinite) Planar Configuration: single image plane Page Number: 0-2 Group 3 Options: 2-d encoding+EOL padding (5 = 0x5) Fax Data: clean (0 = 0x0) Fax Receive Time: 83 secs TIFF Directory at offset 0x22ce4 Image Width: 1728 Image Length: 2148 Resolution: 77, 77 pixels/cm Bits/Sample: 1 Compression Scheme: CCITT Group 3 Photometric Interpretation: min-is-white FillOrder: lsb-to-msb Date Time: 2005/05/04 13:08:35 Host Computer: asterisk1.local Software: spandsp Orientation: row 0 top, col 0 lhs Samples/Pixel: 1 Rows/Strip: (infinite) Planar Configuration: single image plane Page Number: 1-2 Group 3 Options: 2-d encoding+EOL padding (5 = 0x5) Fax Data: receiver regenerated (1 = 0x1) Bad Fax Lines: 7 Consecutive Bad Fax Lines: 4 Fax Receive Time: 39 secs Unfortantly when i create tiff's using other programs (for example ghostscript) with identical settings, they send, but the page sizes get all messed up and parts of each page get cut off. On 5/13/05, Nenad Radosavljevic [EMAIL PROTECTED] wrote: Hi ! Does anyone managed to send multipage faxes (in single TIFF file) with app_txfax from spandsp package (i'm using 0.0.2pre18, libtiff 3.7.1)? If so, I'm interested in format of TIFF file that has been sent sent succesfully (tiffinfo fax-filename). I'm having problems with app_txfax, sending first page successfuly 99 % of the time, but never managed to send second or third page (treid sending to Panasonic fax and to app_rxfax on other Asterisk through the PSTN). Receiving FAX machine hangs up after first page it got (no errors reported) and app_txfax is holding channel for ever (until soft hangup Zap/XX). Here is my tiffinfo on 3 sample files I have tried to fax (on all three only first page being received): LOW-RES fax file (204xx98): lowres:/var/spool/asterisk/fax-out# tiffinfo FAX68.TIF TIFF Directory at offset 0x9160 Image Width: 1728 Image Length: 1173 Resolution: 204, 98 pixels/inch Bits/Sample: 1 Compression Scheme: CCITT Group 3 Photometric Interpretation: min-is-white FillOrder: lsb-to-msb Samples/Pixel: 1 Rows/Strip: 1173 Planar Configuration: single image plane Page Number: 0-3 Group 3 Options: EOL padding (4 = 0x4) TIFF Directory at offset 0x11eb6 Image Width: 1728 Image Length: 1173 Resolution: 204, 98 pixels/inch Bits/Sample: 1 Compression Scheme: CCITT Group 3 Photometric Interpretation: min-is-white FillOrder: lsb-to-msb Samples/Pixel: 1 Rows/Strip: 1173 Planar Configuration: single image plane Page Number: 1-3 Group 3 Options: EOL padding (4 = 0x4) TIFF Directory at offset 0x16e3e Image Width: 1728 Image Length: 1173 Resolution: 204, 98 pixels/inch Bits/Sample: 1 Compression Scheme: CCITT Group 3 Photometric Interpretation: min-is-white FillOrder: lsb-to-msb Samples/Pixel: 1 Rows/Strip: 1173 Planar Configuration: single image plane Page Number: 2-3 Group 3 Options: EOL padding (4 = 0x4) same file HI-RES (204x196) : pbx:/var/spool/asterisk/fax-out# tiffinfo FAX67.TIF TIFF Directory at offset 0x128e8 Subfile Type: multi-page document (2 = 0x2) Image Width: 1728 Image Length: 2347 Resolution: 204, 196 pixels/inch Bits/Sample: 1 Compression Scheme: CCITT Group 3 Photometric Interpretation: min-is-white FillOrder: lsb-to-msb Samples/Pixel: 1 Rows/Strip: 2347 Planar Configuration: single image plane Page Number: 0-3 Group 3 Options: EOL padding (4 = 0x4) TIFF Directory at offset 0x24a6c Subfile Type: multi-page document (2 = 0x2) Image Width: 1728 Image Length: 2347 Resolution: 204, 196 pixels/inch Bits/Sample: 1 Compression Scheme: CCITT Group 3 Photometric Interpretation: min-is-white FillOrder: lsb-to-msb Samples/Pixel: 1 Rows/Strip: 2347 Planar Configuration: single image plane Page Number: 1-3 Group 3 Options: EOL padding (4 = 0x4) TIFF Directory at offset 0x2ed84 Subfile Type: multi-page document (2 = 0x2) Image Width: 1728 Image Length: 2347 Resolution: 204, 196 pixels/inch Bits/Sample: 1 Compression Scheme: CCITT Group 3 Photometric Interpretation: min-is-white FillOrder: lsb-to-msb Samples/Pixel: 1 Rows/Strip: 2347 Planar Configuration: single image plane Page
Re: [Asterisk-Users] Re: chan_capi, chan_misdn and chan_modem
On Fri, May 13, 2005 at 12:33:10PM +0200, Armin Schindler wrote: On Fri, 13 May 2005, Louis-David Mitterrand wrote: On Fri, May 13, 2005 at 09:55:58AM +0200, Armin Schindler wrote: On Fri, 13 May 2005, Paul Hales wrote: I battled with chan_capi during the week, and it was not fun. Since I'm working on chan_capi, I would like to know what problems exist. Can you please be more specific on what problems you have encountered? It's good to see a capi expert working on that module! Could you elaborate on your plans for chan_capi? Currently I have three topics: - fix the problem with kernel 2.6 (looks like a CAPI problem) You mean the loud cracking, poping and line crosstalk on SMP-enabled kernels I notified you about? - cleanup in chan_capi.c (I noticed some errors) - add native bridging using CAPI Line-Interconnect Nice. Then I hope to receive some reports on what is buggy/not working, wishlist and hopefully also some reports on what works well. ECT (explicit call transfer) seems broken (last time I tried it). I also was thinking about an application for receiving fax over CAPI, but I'm not yet familiar with the current asterisk fax support, so I need to learn more here. Maybe some else can inlight me here... Faxing in asterisk is Steve Underwood's ([EMAIL PROTECTED]) specialty. -- Slight disorientation after prolonged system uptime is normal for new Linux users. Please do not adjust your browser. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Why always getting max retries error during idle?
Hi Mike, Probably the same problem i had i while back. The ATA-box dont support message waiting indicatons from asterisk and therefore dont respond to the message, asterisk restries 5 times before giving up with a warning in the log. Iresolved it by removing the mailbox= in sip.conf for that ATA-box. //Magnus From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael StahlSent: Friday, May 13, 2005 5:09 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Why always getting "max retries" error during idle? My home asterisk seems to work- I can call from one internal phone to another. However, just leaving my system idle always generates an error message relating to a NOTIFY. See the log below. Any ideas? Thanks, Mike --MESSAGE FILE- to 172.31.254.106:5065May 13 11:01:28 VERBOSE[716]: Retransmitting #5 (no NAT):NOTIFY sip:[EMAIL PROTECTED]:5065 SIP/2.0Via: SIP/2.0/UDP 172.31.254.2:5060;branch=z9hG4bK6e7b6440From: "asterisk" sip:[EMAIL PROTECTED];tag=as116e0fc1To: sip:[EMAIL PROTECTED]:5065Contact: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 102 NOTIFYUser-Agent: Asterisk PBXEvent: message-summaryContent-Type: application/simple-message-summaryContent-Length: 42 Messages-Waiting: noVoice-Message: 0/0 to 172.31.254.106:5065May 13 11:01:29 WARNING[716]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request)May 13 11:01:32 VERBOSE[716]: ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk extensions from Mysql
I guess you don't know how to read. Failed to connect database server asterisk on. Check debug for more info. Holy cow! You failed to connect to your database! Imagine that. I wonder why it isn't working. Hmm. Could it be that? Did you check the debug for more info? Probably not seeing as you are emailing the list. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voipjet anyone?
Robert Webb wrote: On Fri, 13 May 2005 07:59:09 -0700 Wiley Siler [EMAIL PROTECTED] wrote: Almost positive iLBC is not allowed Use uLaw... They do allow for iLBC. From their FAQ page: Codecs. Carriers with primarily business customers should use the G.711 codec when sending VoIP traffic to VoipJet. This ensures that all calls are of the highest sound quality and free from compression degradation. You've paid for VoipJet Tier 1 telco termination and G.711 lets you fully enjoy it. And don't forget some other codecs can add significantly to a call's end-to-end latency, too. If you need to save bandwidth (admittedly very expensive in some parts of the world) then the iLBC codec bundled with Asterisk makes an excellent choice. It's free to use and takes one-fouth the bandwidth of G.711. Another important reason to try iLBC is if there is jitter and packet loss on your network's connection to us, because G.711 really needs ideal conditions to work well. Finally, GSM potentially uses even less bandwidth and CPU processing power than iLBC. They do not support SpeeX or G726. -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voipjet anyone?
Good catch. Did not see the FAQ. Robert, are you the one having problems getting this running in AAH? W -Original Message- From: Robert Webb [mailto:[EMAIL PROTECTED] Sent: Thursday, May 12, 2005 11:22 PM To: asterisk-users@lists.digium.com; Wiley Siler Subject: Re: [Asterisk-Users] voipjet anyone? On Fri, 13 May 2005 07:59:09 -0700 Wiley Siler [EMAIL PROTECTED] wrote: Almost positive iLBC is not allowed Use uLaw... They do allow for iLBC. From their FAQ page: Codecs. Carriers with primarily business customers should use the G.711 codec when sending VoIP traffic to VoipJet. This ensures that all calls are of the highest sound quality and free from compression degradation. You've paid for VoipJet Tier 1 telco termination and G.711 lets you fully enjoy it. And don't forget some other codecs can add significantly to a call's end-to-end latency, too. If you need to save bandwidth (admittedly very expensive in some parts of the world) then the iLBC codec bundled with Asterisk makes an excellent choice. It's free to use and takes one-fouth the bandwidth of G.711. Another important reason to try iLBC is if there is jitter and packet loss on your network's connection to us, because G.711 really needs ideal conditions to work well. Finally, GSM potentially uses even less bandwidth and CPU processing power than iLBC. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] About Voip Technology : RTP over TCP
On 5/13/05, Atul Thosar [EMAIL PROTECTED] wrote: hello All I am reading information about VoIP technology For that i am concentrating on SIP (Session Initiation Protocol) and RTP (Real Time Transport Protocol). I am interested in implementing RTP over TCP I found that there are some disadvantages of TCP, some are 1) TCP doesn't support multicasting. TCP stacks don't always support the standard either.. 2) Through TCP is reliable, it heavily depends on retransmission of lost or corrupted packets. That increases Delay. Our aim is real time delivery of packets not reliability. quite the opisite. reliably cheaper calls... 3) TCP congestion control mechanism decreases the congestion window when packet losses are detected. yes and your point... Apart from these flaws i am interested in implementing RTP over TCP read about beowulfs and the use of interconnects over Ethernet.. Is it possible ? yes, been done What is feasibility of implementing RTP over TCP ? see above One more point is What is feasibility of implementing RTP over TCP in case of NAT (Network Address Translation) is there ? what port? DNAT? SNAT? loaded question. Need your valuable suggestion. Thank You ! Atul Thosar ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- sig Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! /sig ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voipjet anyone?
You are completely correct.
I see by the called number that the user is in Phoenix? I am too.
Call me at 4804230118 ext. 1003 if you want some off list assistance
with this.
I have mine running just fine with AAH 0.09.
Thanks,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
Adamson
Sent: Friday, May 13, 2005 9:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] voipjet anyone?
Look a little closer...
WARNING[2442]: Call rejected by 66.246.220.19: No such context/extension
that certainly does not imply an incorrect codec!
Almost positive iLBC is not allowed Use uLaw...
This is directly form the install instructions...
Step 3A (recommended): Set your codec to G.711 ulaw for optimal sound
clarity and minimal transmission delay. In iax.conf (found in
/etc/asterisk) locate the codec section and include the following
only.
disallow=all ; Prevent all codecs...
allow = ulaw ; ...except G.711 ulaw
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin
Bockman
Sent: Thursday, May 12, 2005 10:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] voipjet anyone?
May 12 22:27:05 VERBOSE[2442]: -- Executing
[1;36;40mDial[0;37;40m([1;35;40mSIP/101-ad89[0;37;40m,
[1;35;40mIAX2/voipjet/4803442640[0;37;40m) in new stack May 12
22:27:05 VERBOSE[2442]: -- Called voipjet/4803442640 May 12
22:27:05
WARNING[2442]: Call rejected by 66.246.220.19: No such
context/extension
The number needs to be 10 digits. Add a 1 in front.
Tried that, doesnt seem to matter.
Hmm.. did you do an extensions reload? If so, then it should give you
a different error. Show us.
My iax.conf:
[voipjet]
type=peer
;host=216.118.117.46 primary east
host=69.25.60.30
username=username
secret=pass md5
auth=md5
context=default
disallow=all
allow=ilbc
jitterbuffer=yes
dropcount=1
extensions.conf:
exten = _1NXXNXX,1,Dial,IAX2/voipjet/${EXTEN},,tTwW
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RE: [Asterisk-Users] voipjet anyone?
It would be nice if you post how you set this up to either the wiki or right here. Just a few lines would do nicely. There seems to be allot of people who use voipjet and aah and both are good products. Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Friday, May 13, 2005 10:48 AM To: Andrew Latham; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] voipjet anyone? I use AAH with VoipJet and it works perfectly. Setup was a breeze with absolutely no hand coding of configs required. VoipJet is without a doubt the best outbound provider I have come across. No problems at all yet. knock on wood And the call quality has been awesome. Anyone having trouble getting their AAH setup with voipjet may contact me off list for assistance if they want. Cheers, W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Latham Sent: Friday, May 13, 2005 5:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] voipjet anyone? Personally I thought that VOIPJET has the best service and documentation including simple up to date CDRs also. They do not offer incoming, at least not to me If you follow the instructions on their site it will work, if you are useing AAH then maybe you should look into editing the files by had. Checking the server ping time will help. Everyone with a nonroutable IP address will be surprised which ones are faster. On 5/13/05, JD [EMAIL PROTECTED] wrote: Is it me... or is it voipjet? This week I've been trying various providers, just can't seem to get voipjet to work. I signed up with voipjet but so far can't get it to work inbound or out bound. I always get 'all circuits busy'. May 12 22:27:05 VERBOSE[2442]: -- Executing [1;36;40mDial[0;37;40m([1;35;40mSIP/101-ad89[0;37;40m, [1;35;40mIAX2/voipjet/4803442640[0;37;40m) in new stack May 12 22:27:05 VERBOSE[2442]: -- Called voipjet/4803442640 May 12 22:27:05 WARNING[2442]: Call rejected by 66.246.220.19: No such context/extension May 12 22:27:05 DEBUG[2442]: Immediately destroying 6, having received reject Outbound settings: notransfer=yes auth=md5 context=from-pstn host= 66.246.220.19 secret= md5hashstring type=friend ; also tried peer and user username=1234 Im using [EMAIL PROTECTED], but that shouldnt matter; people have this working or is it me? JD ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- sig Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! /sig ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] broadvoice replacement
Does anyone know of a BYOD provider that terminates calls to NCFA numbers (UK 'national rate'). I enjoyed broadvoices unlimited to those numbers, but this is getting silly now, it doesnt work and no answer if after switching to a new provider it will ever work. Can anyone suggest an alternative provider that serves NCFA? -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Current status of voicemail monitoring?
On Friday 13 May 2005 10:13 am, Nathan Pralle wrote: Hi all. I'm curious as to the current status and development of a way to monitor incoming voicemail in Asterisk. IE: The screen calls with the answering machine feature -- the ability to listen to and break into a currently-recording voicemail if you want to. This feature would be very helpful for our application. I've seen various things in the archives about this but none where someone has said, Yeah, we have this, it works. Does anyone have a current implementation of this and/or a reasonable alternative, and how did you do it? Hackish option -- See call come in using FOP (flash operator), watch it go to voicemail, use ZapBarge to listen to voicemail being recorded. If you want to talk to the caller, use FOP and drag the incomming Zap line onto your phone again, then answer your phone. See FOP docs on transfering calls. Hackish, but it works. I've used it myself several times. This, of course, assumes you are using Zap channels for incomming calls. If not, then you'd need to find another way to listen to incomming calls - perhaps ChanSpy, tho i've not been able to get that to work - crashes my * box with CVS HEAD. Anyway, let me know if thats helpful or not, and we can play around. Oh, another thought: One other way to do this (and this will be comples): 1. call comes in, extension is dialed. 2. Your ext macro (assuming you use a macro for extensions) first calls a script to add your phone to a dynamic meetme conf. 3. The macro then dials this call to the dynamic meet me conf. 4. The script refered to in '2' should me a manger script. The script could use standard manager API stuff like 'action: originate' to dial to your phone from the meetme conf. If your phone doesnt answer, the voicemail app would then play the audio and record from meetme instead of directly from the incomming line/ 5. If you wanted to listen to the caller recording VM at that point instead of answer the call, just dial into the dynamic meetme conf with your phone muted. This assumes the meetme conf was created with the 'q' option to disable announce of joins. 6. If you want to talk, then just unmute your phone. Only question is how to turn off VM. Does standard app_voicemail.o respond to DTMF to stop recording? Would DTMF work in the meetme to stop recording? Anyway, those are just the top-of-the-head thoughts. -josiah -- Josiah Bryan IT Coordinator Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] delay before call file execution
hello i want to insert delay into callfile execution. UA6000(callbackNumber) this will create call file UA---asterisk(callfile) how to insert delay into this callfile execution. thanks Kamran __ Do you Yahoo!? Make Yahoo! your home page http://www.yahoo.com/r/hs ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: chan_capi, chan_misdn and chan_modem
On Fri, 13 May 2005, Louis-David Mitterrand wrote: On Fri, May 13, 2005 at 12:33:10PM +0200, Armin Schindler wrote: On Fri, 13 May 2005, Louis-David Mitterrand wrote: On Fri, May 13, 2005 at 09:55:58AM +0200, Armin Schindler wrote: On Fri, 13 May 2005, Paul Hales wrote: I battled with chan_capi during the week, and it was not fun. Since I'm working on chan_capi, I would like to know what problems exist. Can you please be more specific on what problems you have encountered? It's good to see a capi expert working on that module! Could you elaborate on your plans for chan_capi? Currently I have three topics: - fix the problem with kernel 2.6 (looks like a CAPI problem) You mean the loud cracking, poping and line crosstalk on SMP-enabled kernels I notified you about? I meant http://bugme.osdl.org/show_bug.cgi?id=4356 Yours is still on my list, but I wasn't able yet to reproduce it. Armin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] In/out calls from/to same sip provider
But I'm having a hard time getting calls going to and coming from the same provider, since the definition of the peer in sip.conf seems to be different AND not compatible for incoming and outgoing call. Sometimes what is needed can be provider-dependent. Every provider I've seen gives an example for both incoming and outgoing calls. Often, SIP provider have 2 different entries for incoming and outgoing. Why not start there? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaptel.conf multiple devices
I have the same problem, any ideas people? Gustavo Alvarez Sander crombeen at rommelweb.nl Sun May 1 12:17:31 CDT 2005 * Previous message: [Asterisk-Users] Asterisk 1.0.6 stable IAX2 Firefly supervised call transfer? * Next message: [Asterisk-Users] TDM400P Power Connector * Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] Hi there my zaptel hardware is giving errors while loading but they seem to load just fine. the lights wil work and my wctdm card is also workin and the isdn works to But when I stop asterisk I have to reload al cards again is this normal? This is my zaptel.conf is there no way to group these because my te110p is giving an error that it can't find channel 35 but 35 belongs to my wctdm. Maybe my zaptel.conf is not that good, I can't find any documentation on multiple cards in one system Thanks ZT_SPANCONFIG failed on span 2: No such device or address (6) make: *** [loadlinux26] Error 1 ZT_CHANCONFIG failed on channel 35: No such device or address (6) FATAL: Error running install command for wcte11xp [root at MYPBX src]# loadzone=nl defaultzone=nl span=1,1,3,ccs,ami bchan=1-2 dchan=3 span=1,1,0,ccs,hdb3 bchan=4-18,20-34 # set this to 1-15,17-31 for E1 dchan=19 # set this to 16 for E1 fxoks=35-36 -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050501/0e599ba7/attachment.htm * Previous message: [Asterisk-Users] Asterisk 1.0.6 stable IAX2 Firefly supervised call transfer? * Next message: [Asterisk-Users] TDM400P Power Connector * Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] More information about the Asterisk-Users mailing list ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] broadvoice replacement
On Fri, 2005-05-13 at 09:02 -0700, trixter http://www.0xdecafbad.com wrote: Does anyone know of a BYOD provider that terminates calls to NCFA numbers (UK 'national rate'). I enjoyed broadvoices unlimited to those numbers, but this is getting silly now, it doesnt work and no answer if after switching to a new provider it will ever work. Can anyone suggest an alternative provider that serves NCFA? Hi Trixter, I use Telappliant (www.voiptalk.org). They can do UK National and Geographic numbers, and they support IAX and SIP. HTH Chris -- Chris -- E Mail: [EMAIL PROTECTED] SIP: [EMAIL PROTECTED] IAXTEL: 17003366726 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: chan_capi, chan_misdn and chan_modem
Armin Schindler wrote: please look for the neccessary patches at: http://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2 That is exactly what I was thinking about. I did not have a close look into the patch yet, but this archive seems to be incomplete. Only changed files are part of the patch, but the real app_capiFax.c is missing. The patch was obviously created without the '-N' option of diff. Can you please check that? you are right. i just updated the patch. it should work now. Anyway, I think this should be part of the chan_capi package. yes, i was very happy to hear from klaus-peter he has restarted to improve chan_capi (i thought he lost interest in chan_capi and concentrated only on his bri cards). i hope klaus-peter will include the fax support into chan_capi-0.4.0! So the CAPI on kernel 2.6 problem is on top now... fine. freundliche grüße frank sautter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi, chan_misdn and chan_modem
hallo klaus-peter, Klaus-Peter Junghanns wrote: The new version of chan_capi (0.4.0) is still work in progress (no, I have not dropped chan_capi in favour of BRIstuff). that was my assumption, as there was no progress so many months. i'm very happy, that you are back on developing chan_capi! I harmonized the dialstring syntax with chan_zap, so you can just use CAPI/g1/... instead of those strange constructions with the outgoing msn. It also contains fixes (contributed by Jan Stocke) to make it work on BSD. Chan_capi 0.4.0 will work with Asterisk stable and cvs head. sounds as if this are interesting features. maybe you could take a look on the patches of carl sempla and cedrik hans (faxing with eicon cards) and mine (transfer capability, limitation of MSNs, cvs-head) both available using: http://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2 grüße frank sautter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voipjet anyone?
Wel, that is the real issue. There is no secret method. You literally just add it to the trunks then set it in your outbound routing being careful to make sure you have dial patterns set that match correctly. So Here is the trunk definition - Make sure your trunk name is voipjet with no quotes Enter this in your peer settings auth=md5 context=default host=69.25.60.30 notransfer=yes secret=your secret hash here type=peer username=your username here Nothing required for incoming settings or registration. Nothing required for dial rules or prefix. Change the host if you are not on west coast... - Now go to the Outbound Routes page. - Create a route for your VoipJet Account (ToVoipJet or something like that is a good name) Set a dial pattern. I use this for LD and international... 011. 1NXXNXX Add your trunk to the Trunk Sequence (you have to select it and apply it) and then submit - Now, all that being said, the most common problem (other than something on their side) is going to be if the user malforms their settings. Don't set your trunk name to 1234 (your username) for example. Set it to voipjet. Make sure the context is correctly defined exactly like above. It is default. Let me know if that does not work for you. It works perfect for me W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ariel Batista Sent: Friday, May 13, 2005 8:50 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] voipjet anyone? It would be nice if you post how you set this up to either the wiki or right here. Just a few lines would do nicely. There seems to be allot of people who use voipjet and aah and both are good products. Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Friday, May 13, 2005 10:48 AM To: Andrew Latham; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] voipjet anyone? I use AAH with VoipJet and it works perfectly. Setup was a breeze with absolutely no hand coding of configs required. VoipJet is without a doubt the best outbound provider I have come across. No problems at all yet. knock on wood And the call quality has been awesome. Anyone having trouble getting their AAH setup with voipjet may contact me off list for assistance if they want. Cheers, W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Latham Sent: Friday, May 13, 2005 5:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] voipjet anyone? Personally I thought that VOIPJET has the best service and documentation including simple up to date CDRs also. They do not offer incoming, at least not to me If you follow the instructions on their site it will work, if you are useing AAH then maybe you should look into editing the files by had. Checking the server ping time will help. Everyone with a nonroutable IP address will be surprised which ones are faster. On 5/13/05, JD [EMAIL PROTECTED] wrote: Is it me... or is it voipjet? This week I've been trying various providers, just can't seem to get voipjet to work. I signed up with voipjet but so far can't get it to work inbound or out bound. I always get 'all circuits busy'. May 12 22:27:05 VERBOSE[2442]: -- Executing [1;36;40mDial[0;37;40m([1;35;40mSIP/101-ad89[0;37;40m, [1;35;40mIAX2/voipjet/4803442640[0;37;40m) in new stack May 12 22:27:05 VERBOSE[2442]: -- Called voipjet/4803442640 May 12 22:27:05 WARNING[2442]: Call rejected by 66.246.220.19: No such context/extension May 12 22:27:05 DEBUG[2442]: Immediately destroying 6, having received reject Outbound settings: notransfer=yes auth=md5 context=from-pstn host= 66.246.220.19 secret= md5hashstring type=friend ; also tried peer and user username=1234 Im using [EMAIL PROTECTED], but that shouldnt matter; people have this working or is it me? JD ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- sig Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! /sig ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com
[Asterisk-Users] Chanspy crash
Hackish, but it works. I've used it myself several times. This, of course, assumes you are using Zap channels for incomming calls. If not, then you'd need to find another way to listen to incomming calls - perhaps ChanSpy, tho i've not been able to get that to work - crashes my * box with CVS HEAD. -josiah It does this to me too. We should send in a bug report. I think there have been other people reporting it also. It would be reeel nice-like [Beverly Hillbillies] to have this working. Kevin Bockman ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] About Voip Technology : RTP over TCP
TCP is too slow for Real time Apps. If you have packet errors TCP will try to resend the packet. This will create latency issues. This is why UDP is used for Voip. 1 or 2 missing packets is not going to be missed. If you look at your Stats. you'll see a few of them. Stewart Nelson wrote: I am interested in implementing RTP over TCP Why? If you want to permit operation through a firewall that blocks UDP, there are packages that provide VPN tunnels over TCP or even HTTP. You could then run any VoIP system over that VPN. As you said, delay performance would sometimes be awful. Skype will automatically fall back to TCP if a UDP connection attempt fails. Most of the commercial instant messaging packages that support voice or video can work over TCP. If your purpose is to improve performance on networks with high packet loss rates, IMHO you would get better results from a UDP-based system that permits forward error correction, by transmitting each voice frame in two or more packets. If you can't afford the increased bandwidth, a system of retransmission such as used by popular streaming protocols would still be better than TCP. One more point is What is feasibility of implementing RTP over TCP in case of NAT (Network Address Translation) is there ? Any of the above systems can work through NAT. If both endpoints are behind NATs, and you can't set up port forwarding on either, then of course you must connect via an intermediate server. Skype and the IM services do that automatically. If your desire for TCP is not related to firewalls or packet loss, I'd be interested in hearing about your application. --Stewart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sangoma fdc 3?
Works great for me also. On Mon, 2005-05-09 at 14:33 +0200, Altus Snyman wrote: How well does the sangoma cards work with fedora core 3 Im doing the research on what hardware/os I need to use Please help and advice ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voipjet anyone?
I had the same problem, there were a couple problems, mostly with my
dialplan:
AAH 1.0 Config
Trunk Settings
Trunk Name: voipjet
Outbound Caller ID: Tech Data Pros 9495027819
Maximum Channels: 4
Dial Rules: 1949+NXX ; you need to add local area code
Outgoing Settings:
Trunkname: voipjet
PEER Details:
allow= ulaw
auth=md5
context=to-voipjet
disallow=all
host=69.25.60.30
notransfer=yes
secret= mysecret
type=peer
username=myuserid
Outbound Routing Settings
Route Name: FWDOUT
Dial Patterns: 393|.
Note: for toll free add the following (You will need to dial * before the
toll free number:
*1800NXX
*1888NXX
*1877NXX
*1866NXX
Trunk Sequence: 0 IAX2/fwd
Save all your settings, click on red bar to apply, wait 5-10 seconds
depending on the speed of your machine. Dial 393612 for current time and
393613 for an echo test.
Kerry Garrison
http://geekgazette.com - http://techdatapros.com IT/VoIP Consultants
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler
Sent: Friday, May 13, 2005 8:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] voipjet anyone?
You are completely correct.
I see by the called number that the user is in Phoenix? I am too.
Call me at 4804230118 ext. 1003 if you want some off list assistance with
this.
I have mine running just fine with AAH 0.09.
Thanks,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Friday, May 13, 2005 9:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] voipjet anyone?
Look a little closer...
WARNING[2442]: Call rejected by 66.246.220.19: No such context/extension
that certainly does not imply an incorrect codec!
Almost positive iLBC is not allowed Use uLaw...
This is directly form the install instructions...
Step 3A (recommended): Set your codec to G.711 ulaw for optimal sound
clarity and minimal transmission delay. In iax.conf (found in
/etc/asterisk) locate the codec section and include the following
only.
disallow=all ; Prevent all codecs...
allow = ulaw ; ...except G.711 ulaw
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin
Bockman
Sent: Thursday, May 12, 2005 10:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] voipjet anyone?
May 12 22:27:05 VERBOSE[2442]: -- Executing
[1;36;40mDial[0;37;40m([1;35;40mSIP/101-ad89[0;37;40m,
[1;35;40mIAX2/voipjet/4803442640[0;37;40m) in new stack May 12
22:27:05 VERBOSE[2442]: -- Called voipjet/4803442640 May 12
22:27:05
WARNING[2442]: Call rejected by 66.246.220.19: No such
context/extension
The number needs to be 10 digits. Add a 1 in front.
Tried that, doesnt seem to matter.
Hmm.. did you do an extensions reload? If so, then it should give you
a different error. Show us.
My iax.conf:
[voipjet]
type=peer
;host=216.118.117.46 primary east
host=69.25.60.30
username=username
secret=pass md5
auth=md5
context=default
disallow=all
allow=ilbc
jitterbuffer=yes
dropcount=1
extensions.conf:
exten = _1NXXNXX,1,Dial,IAX2/voipjet/${EXTEN},,tTwW
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