[asterisk-users] Top posting - there is no rule.
What's with the occasional Un-Top-posting, there is no rule that says you can't, http://www.asterisk.org/community/rules My preference is top posting, as you see the answer at a quick glance, instead of reaching for the scroll bar (or whatever key stokes are required) to get to the bottom, to find that the answer isn't there yet. Note: Flaming is not an acceptable behaviour :) Alec Davis PS. Sorry to the asterisk-dev list that have seen this already, posted in wrong forum. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Top posting - there is no rule.
On Sun, Apr 3, 2011 at 1:54 AM, Alec Davis siva...@paradise.net.nz wrote: What's with the occasional Un-Top-posting, there is no rule that says you can't, http://www.asterisk.org/community/rules Really? Why bring this up again? The last 60-some odd thread in January wasn't long enough for you? Or perhaps you're just out looking to troll? From the page you linked: 1. Responses should be placed under the original quoted text. -- Thanks, --Warren Selby, dCAP http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Top posting - there is no rule.
Really? Why bring this up again? The last 60-some odd thread in January wasn't long enough for you? Or perhaps you're just out looking to troll?From the page you linked: 5.Responses should be placed under the original quoted text. Sorry my mistake, didn't read to bottom of the rules. Nearly top posted again, it was hard not to... Until I found what causes Outlook to mess up formatting replies. If reply indent option is enabled, and if message is received in HTML format, need to disable HTML format (Send Plain Text) Alec Davis -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BRI detection
On Fri, Apr 01, 2011 at 06:06:45PM +0530, mahesh katta wrote: Hi, I need to configure BRI 4span card in dubai in vicidialnow for dialer perpose. in that i have small confusion which is NT an TE mode . that was i am setting perfectly but dubai telco what they are use for this i dont know which parameters are use for that . please help me. TE (or CPE) is phone. If you you just connect your adapter to a telco, this is what you need. What card is it? What BRI channel driver? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [DIGIUM FAX] HANGUP problem
Hi I noticed that all my fax negociated in *V17 *hangup Here my conf fax show version FAX For Asterisk Components: Applications: 1.8.4-rc2 Digium FAX Driver: 1.8.4_1.3.0 (optimized for i686_32) Here an example of failed fax: -- Channel 'SIP/VOIP-OUT-0043' FAX session '1' started -- FAX handle 0: [ 000.53 ], STAT_EVT_STRT_TX st: IDLE rt: IDLENSTX -- FAX handle 0: [ 000.000171 ], STAT_EVT_TX_HW_RDY st: WT_TX_HW_RDY rt: TRDYNHTY -- FAX handle 0: [ 000.000231 ], P30EVN_SEND_STARTED Channel 'SIP/VOIP-OUT-0043' fax session '1', [ 000.253769 ], stack sent 1 frames (20 ms) of silence. Channel 'SIP/VOIP-OUT-0043' fax session '1', [ 000.767866 ], stack sent 26 frames (520 ms) of energy. Channel 'SIP/VOIP-OUT-0043' fax session '1', [ 000.993930 ], channel sent 38 frames (760 ms) of silence. Channel 'SIP/VOIP-OUT-0043' fax session '1', [ 003.568856 ], stack sent 140 frames (2800 ms) of silence. Channel 'SIP/VOIP-OUT-0043' fax session '1', [ 004.088848 ], stack sent 26 frames (520 ms) of energy. Channel 'SIP/VOIP-OUT-0043' fax session '1', [ 004.734224 ], channel sent 187 frames (3740 ms) of energy. Channel 'SIP/VOIP-OUT-0043' fax session '1', [ 004.814201 ], channel sent 4 frames (80 ms) of silence. -- FAX handle 0: [ 006.630557 ], STAT_INFO_CSI Channel 'SIP/VOIP-OUT-0043' fax session '1', [ 006.894347 ], channel sent 104 frames (2080 ms) of energy. -- FAX handle 0: [ 006.949423 ], STAT_INFO_DIS -- FAX handle 0: [ 006.949526 ], STAT_EVT_DIS st: WT_DIS rt: NDIS -- FAX handle 0: [ 006.949623 ], STAT_EVT_REMOTE_RX st: WT_DIS rt: WDISNRRX -- FAX handle 0: [ 006.949742 ], STAT_NEG_V17_14400 -- FAX handle 0: [ 006.949841 ], STAT_NEG_ECM -- FAX handle 0: [ 006.949932 ], STAT_NEG_MMR -- FAX handle 0: [ 006.950030 ], STAT_NEG_RES_204x196 -- FAX handle 0: [ 006.950122 ], STAT_NEG_A4 -- FAX handle 0: [ 006.950221 ], STAT_INFO_TSI -- FAX handle 0: [ 006.950377 ], STAT_INFO_DCS Channel 'SIP/VOIP-OUT-0043' fax session '1', [ 007.113237 ], stack sent 151 frames (3020 ms) of silence. -- FAX handle 0: [ 008.043552 ], STAT_EVT_TMR_INT_EXP st: WT_DIS rt: NTIX -- FAX handle 0: [ 008.970957 ], STAT_EVT_TX_V21_DONE st: WT_HW_CLS rt: UNEXPECT -- FAX handle 0: [ 010.541022 ], STAT_EVT_HW_CLOSE st: WT_HW_CLS rt: WCLSNCLS -- FAX handle 0: [ 010.541314 ], STAT_SES_COMPLETE -- FAX handle 0: [ 010.541484 ], P30EVN_COMPLETE [Apr 3 12:28:06] ERROR[24201]: res_fax.c:1339 generic_fax_exec: channel 'SIP/VOIP-OUT-0043' FAX session '1' failure, reason: 'fax session timed-out' (TIMEOUT) Can you help pls? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The SIP channel driver - I'm giving up.
On Fri, Apr 01, 2011 at 09:06:53AM +, Tony Mountifield wrote: In article b4ed7e5e-56d7-4ab3-8f9c-0d120a741...@edvina.net, Olle E. Johansson o...@edvina.net wrote: Friends, After having spent many years working with the Asterisk SIP channel driver and the SIPv2 protocol, I have finally realized that this is a dead end. It's getting nowhere and it's way too complicated to set up, run and support in working code. After realizing this, I started a new standardization project together with my friends in Canada, Simon and Marc, to develop a working solution based on the combination of IPv6 and SIP. We have gotten great feedback and now the IETF, the ITU and the IPv6 forum jointly launches the new standard, SIP-six. Hmmm, fancy announcing this on 1 April Sure. A reminder: Asterisk 2.0: http://lists.digium.com/pipermail/asterisk-users/2005-April/091612.html http://lists.digium.com/pipermail/asterisk-users/2005-April/thread.html#91612 SIP 4.0: http://tools.ietf.org/html/draft-kaplan-sip-four-oh-00 -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] hello
I have bought a pair of Apple phones on---www.ofenno.com---they have pretty good quality. Here I would like to recommend it to you. Their company is holding a promotion activity now, so you can buy anything you want on it with free delivery charges. There must be anything you like, I hope you would not miss this chance. All The Best-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6 = No sound/voice when i redirect the call
Hi
i use this into my extension :
exten = _00339,1,Set(foo=${SIP_HEADER(To)})
exten = _00339,2,Set(cut1=${CUT(foo,:,2)})
exten = _00339,3,Set(CLI=${CUT(cut1,,1)})
exten = _00339,4,Set(toexten=${CUT(CLI,@,1)})
exten = _00339,5,Noop(ORIGINAL NUMBER : [ ${toexten} ])
exten = _00339,6,AGI(Ddi-Network.agi,${toexten})
exten = _00339,7,Set(CALLERPRES()=prohib_not_screened)
exten = _00339,8,Dial(SIP/MyOperator/${NUMAPPEL},180,rt)
exten = _00339,9,Hangup
and i have in sip.conf:
[MyOperator]
type=peer
host=host-of-my-operator
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=yes
insecure=port,invite
dtmfmode=rfc2833
disallow=all
allow=g729
allow=alaw
allow=g723
defaultuser=0033xx
secret=x
When i call directly from [MyOperator], no probleme i have sound/Voice
but when a customer call to the 00339xxx.., the call are correct, asterisk
call to my standard SIP/MyOperator/${NUMAPPEL} but no sound/voice
(i receive the call without problems, only sound off)
anyone have a idea of this problems ?
bye
Olivier
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Re: [asterisk-users] Asterisk 1.6 = No sound/voice when i redirect the call
In that situation, I've had to do a pickup macro that kind of primes
the audio.
Dial(SIP/MyOperator/${NUMAPPEL},180,rtU(start-audio))
context start-audio {
s = {
Playback(silence/1);
}
}
The above might help... What it does is plays an audio track on the
callee's channel (SIP/MyOperator-) before bridging the audio.
On 04/03/11 12:01, Olivier CALVANO wrote:
Hi
i use this into my extension :
exten = _00339,1,Set(foo=${SIP_HEADER(To)})
exten = _00339,2,Set(cut1=${CUT(foo,:,2)})
exten = _00339,3,Set(CLI=${CUT(cut1,,1)})
exten = _00339,4,Set(toexten=${CUT(CLI,@,1)})
exten = _00339,5,Noop(ORIGINAL NUMBER : [ ${toexten} ])
exten = _00339,6,AGI(Ddi-Network.agi,${toexten})
exten = _00339,7,Set(CALLERPRES()=prohib_not_screened)
exten = _00339,8,Dial(SIP/MyOperator/${NUMAPPEL},180,rt)
exten = _00339,9,Hangup
and i have in sip.conf:
[MyOperator]
type=peer
host=host-of-my-operator
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=yes
insecure=port,invite
dtmfmode=rfc2833
disallow=all
allow=g729
allow=alaw
allow=g723
defaultuser=0033xx
secret=x
When i call directly from [MyOperator], no probleme i have sound/Voice
but when a customer call to the 00339xxx.., the call are correct, asterisk
call to my standard SIP/MyOperator/${NUMAPPEL} but no sound/voice
(i receive the call without problems, only sound off)
anyone have a idea of this problems ?
bye
Olivier
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Re: [asterisk-users] Asterisk 1.6 = No sound/voice when i redirect the call
Hi Mark
Thanks for your answer, but i am new in asterisk ;=) the context
start-audio ...
i put it into the extension.conf ?
because i have a error:
[Apr 3 20:12:08] WARNING[28078]: config.c:1154 process_text_line: No
'=' (equal sign) in line 134 of /etc/asterisk/extension-operator.conf
[Apr 3 20:12:08] WARNING[28078]: config.c:1154 process_text_line: No
'=' (equal sign) in line 136 of /etc/asterisk/extension-operator.conf
[Apr 3 20:12:17] WARNING[28078]: pbx_config.c:1499 pbx_load_config:
==!!== Unknown directive: s at line 135 -- IGNORING!!!
thanks for your help
olivier
2011/4/3 Mark Murawski markm-li...@intellasoft.net:
In that situation, I've had to do a pickup macro that kind of primes the
audio.
Dial(SIP/MyOperator/${NUMAPPEL},180,rtU(start-audio))
context start-audio {
s = {
Playback(silence/1);
}
}
The above might help... What it does is plays an audio track on the callee's
channel (SIP/MyOperator-) before bridging the audio.
On 04/03/11 12:01, Olivier CALVANO wrote:
Hi
i use this into my extension :
exten = _00339,1,Set(foo=${SIP_HEADER(To)})
exten = _00339,2,Set(cut1=${CUT(foo,:,2)})
exten = _00339,3,Set(CLI=${CUT(cut1,,1)})
exten = _00339,4,Set(toexten=${CUT(CLI,@,1)})
exten = _00339,5,Noop(ORIGINAL NUMBER : [ ${toexten} ])
exten = _00339,6,AGI(Ddi-Network.agi,${toexten})
exten = _00339,7,Set(CALLERPRES()=prohib_not_screened)
exten = _00339,8,Dial(SIP/MyOperator/${NUMAPPEL},180,rt)
exten = _00339,9,Hangup
and i have in sip.conf:
[MyOperator]
type=peer
host=host-of-my-operator
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=yes
insecure=port,invite
dtmfmode=rfc2833
disallow=all
allow=g729
allow=alaw
allow=g723
defaultuser=0033xx
secret=x
When i call directly from [MyOperator], no probleme i have sound/Voice
but when a customer call to the 00339xxx.., the call are correct,
asterisk
call to my standard SIP/MyOperator/${NUMAPPEL} but no sound/voice
(i receive the call without problems, only sound off)
anyone have a idea of this problems ?
bye
Olivier
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[asterisk-users] From 1.4 to 1.8: stdexten issue
Hello all,
I'm in the middle of upgrading my asterisk setup to 1.8 (1.8.2.3) and
I'm completely confused by the gosub/stdexten thing.
I used to call the stdexten macro but I haven't been able to figure out
how to use Gosub.
I'm using the sample extensions.conf and added something like this:
=
[home]
include = stdexten
exten = 1,1,Gosub(${EXTEN},stdexten(SIP/phone1))
=
But if I call 1, all I get is:
[Apr 3 18:20:51] NOTICE[9031]: pbx.c:4119 pbx_extension_helper: No such label
'stdexten' in extension '1' in context 'home'
[Apr 3 18:20:51] WARNING[9031]: pbx.c:10174 pbx_parseable_goto: Priority
'stdexten' must be a number 0, or valid label
[Apr 3 18:20:51] ERROR[9031]: app_stack.c:411 gosub_exec: Gosub address is
invalid: '1,stdexten(SIP/phone1)'
I've googled and pretty much tried all forms of the syntax but I've yet
to make it work. For instance I tried not including stdexten and
calling Gosub with (stdexten,${EXTEN},1(SIP/phone1)) but it didn't work
either...
Can anyone sched some light here? I think I got lost trying to figure
this out...
What am I missing here?
Best,
--
Mathieu Chouquet-Stringer math...@csetco.com
The sun itself sees not till heaven clears.
-- William Shakespeare --
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Re: [asterisk-users] Asterisk 1.6 = No sound/voice when i redirect the call
I gave you the syntax in ael format, if you want to use extensions.conf
you'll have to use the syntax that's applicable, which is:
[start-audio]
exten = s,1,Playback(silence/1)
On 04/03/11 14:14, Olivier CALVANO wrote:
Hi Mark
Thanks for your answer, but i am new in asterisk ;=) the context
start-audio ...
i put it into the extension.conf ?
because i have a error:
[Apr 3 20:12:08] WARNING[28078]: config.c:1154 process_text_line: No
'=' (equal sign) in line 134 of /etc/asterisk/extension-operator.conf
[Apr 3 20:12:08] WARNING[28078]: config.c:1154 process_text_line: No
'=' (equal sign) in line 136 of /etc/asterisk/extension-operator.conf
[Apr 3 20:12:17] WARNING[28078]: pbx_config.c:1499 pbx_load_config:
==!!== Unknown directive: s at line 135 -- IGNORING!!!
thanks for your help
olivier
2011/4/3 Mark Murawskimarkm-li...@intellasoft.net:
In that situation, I've had to do a pickup macro that kind of primes the
audio.
Dial(SIP/MyOperator/${NUMAPPEL},180,rtU(start-audio))
context start-audio {
s = {
Playback(silence/1);
}
}
The above might help... What it does is plays an audio track on the callee's
channel (SIP/MyOperator-) before bridging the audio.
On 04/03/11 12:01, Olivier CALVANO wrote:
Hi
i use this into my extension :
exten =_00339,1,Set(foo=${SIP_HEADER(To)})
exten =_00339,2,Set(cut1=${CUT(foo,:,2)})
exten =_00339,3,Set(CLI=${CUT(cut1,,1)})
exten =_00339,4,Set(toexten=${CUT(CLI,@,1)})
exten =_00339,5,Noop(ORIGINAL NUMBER : [ ${toexten} ])
exten =_00339,6,AGI(Ddi-Network.agi,${toexten})
exten =_00339,7,Set(CALLERPRES()=prohib_not_screened)
exten =_00339,8,Dial(SIP/MyOperator/${NUMAPPEL},180,rt)
exten =_00339,9,Hangup
and i have in sip.conf:
[MyOperator]
type=peer
host=host-of-my-operator
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=yes
insecure=port,invite
dtmfmode=rfc2833
disallow=all
allow=g729
allow=alaw
allow=g723
defaultuser=0033xx
secret=x
When i call directly from [MyOperator], no probleme i have sound/Voice
but when a customer call to the 00339xxx.., the call are correct,
asterisk
call to my standard SIP/MyOperator/${NUMAPPEL} but no sound/voice
(i receive the call without problems, only sound off)
anyone have a idea of this problems ?
bye
Olivier
--
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Re: [asterisk-users] From 1.4 to 1.8: stdexten issue
Mathieu Chouquet-Stringer math...@csetco.com writes:
I've googled and pretty much tried all forms of the syntax but I've yet
to make it work. For instance I tried not including stdexten and
calling Gosub with (stdexten,${EXTEN},1(SIP/phone1)) but it didn't work
either...
stdexten in the default extensions.conf seems to only handle extensions
with at least 2 digits...
/Benny
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Re: [asterisk-users] hello
-BEGIN PGP SIGNED MESSAGE- Hash: RIPEMD160 WTF does that have to do with Asterisk? On 04/03/2011 05:56 AM, ALAEDDINE abbech wrote: I have bought a pair of Apple phones on---www.ofenno.com---they have pretty good quality. Here I would like to recommend it to you. Their company is holding a promotion activity now, so you can buy anything you want on it with free delivery charges. There must be anything you like, I hope you would not miss this chance. All The Best -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - -- Bradley D. Thornton Manager Network Services NorthTech Computer TEL: +1.760.666.2703 (US) TEL: +44.702.405.1909 (UK) http://NorthTech.US -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.10 (GNU/Linux) Comment: Find this cert at x-hkp://pool.sks-keyservers.net iQEcBAEBAwAGBQJNmOWiAAoJEE1wgkIhr9j3BSQH/jFesWnErzA/BLOMEjkCYs5F CTJ9BcoI1gPyxrhfDzQKwtf26HTMwEZNPwURSq9QFP1ceV5mmRp3yk69YqlUsjve dzWAZBcvGlCRuWV09IryToqtuZuCeUq/qIJ4gITIgzFwsYsT0nO307ZrT3BLyfn0 emky3UrgGKM9agTuufVaehonr/VLdln68V/bC0S66VOhhcQ1Dv7Zof4VOLv4+txu IQ8v4djH9yhoTavgbDzJH3SiCreSumSN14cTiLboZOZpTX3RPJqcBSSLXd+rQtfy vSEf0J72ZbzLh8ACmaZqt1lutzrJCh5Gkhvp9Twj9zkX4/yMBLiUYQBTjOWtd9Q= =G1Bj -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hello
Bradley D. Thornton wrote: WTF does that have to do with Asterisk? It's called spam. And either he doesn't know what non-commercial discussion means or he signed up just to send this. Doubtful we'll see him again. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
